Problem dialling from Switzerland to mobile phones in Italy

Paolodic · February 28, 2018, 10:35pm

Ciao!
I have the following problem:
dialling an Italian mobile Phone from Switzerland doesn’t work. If I dial a fix line in Italy it is OK!
Fix phone numer: 0039 010 XXXXXX 6 digits
Mobile phone number: 0039 349 XXXXXXX 7 digits

Outbound route has only “0.” in the dial pattern+ service and emergency numbers with “1.”

Problem:
Calls to fix numbers are OK
Calls to mobile numbers: fail. The called subscriber receive a single ring, then the call is lost. On the caller side, I have a single ring tone and then silence!

All calls to Switzerland are fine.

My config:
Raspbk 13.0.194.2
Provider: Swisscom (over Swisscom Internet Box and pbx configured as a SIP Phone)

Can anyone help me?

Grazie!
Paolo

cynjut · March 1, 2018, 3:15pm

Not without troubleshooting info and log files. Find a failed call in /var/log/asterisk/full and see what the provider is saying about the call failure.

After that, contact your provider with the information about the call (date, time, your number, the recipient number) and see what their analysis says.

This is probably going to be a provider problem. Since the phone rings once, the call circuit is getting established, but something in the call setup is failing.

Paolodic · March 2, 2018, 10:00am

Hi Dave,
Thank you for the answer.
I have got a look at the Asterisk log, for one successful call to Switzerland and one unsuccessful to Italy. They are the same, with the exception when the call is ringing.
When the call is failing, I hear a single cutted ring tone. On the called party, the phone rings and when the call is answered, there is no voice channel.
Here the extract of the logs (just the differences).

FAIL:
[2018-03-01 19:24:14] VERBOSE[3512][C-000004a2] netsock2.c: Using SIP RTP TOS bits 184
[2018-03-01 19:24:14] VERBOSE[3512][C-000004a2] netsock2.c: Using SIP RTP CoS mark 5
[2018-03-01 19:24:14] VERBOSE[3512][C-000004a2] app_dial.c: Called SIP/swisscom/00393405319359
[2018-03-01 19:24:15] VERBOSE[3512][C-000004a2] app_dial.c: SIP/swisscom-00000f2a is ringing
[2018-03-01 19:24:15] VERBOSE[3512][C-000004a2] app_dial.c: SIP/swisscom-00000f2a is making progress passing it to SIP/302-00000f29
[2018-03-01 19:24:18] VERBOSE[3512][C-000004a2] app_dial.c: SIP/swisscom-00000f2a is ringing
[2018-03-01 19:24:26] VERBOSE[3512][C-000004a2] app_macro.c: Spawn extension (macro-dialout-trunk, s, 23) exited non-zero on ‘SIP/302-00000f29’ in macro ‘dialout-trunk’
[2018-03-01 19:24:26] VERBOSE[3512][C-000004a2] pbx.c: Spawn extension (from-internal, 00393405319359, 6) exited non-zero on ‘SIP/302-00000f29’
[2018-03-01 19:24:26] VERBOSE[3512][C-000004a2] pbx.c: Executing [h@from-internal:1] Macro(“SIP/302-00000f29”, “hangupcall”) in new stack
[2018-03-01 19:24:26] VERBOSE[3512][C-000004a2] pbx.c: Executing [s@macro-hangupcall:1] GotoIf(“SIP/302-00000f29”, “1?theend”) in new stack
[2018-03-01 19:24:26] VERBOSE[3512][C-000004a2] pbx_builtins.c: Goto (macro-hangupcall,s,3)

SUCCESS:
[2018-03-01 21:32:40] VERBOSE[5340][C-000004a6] netsock2.c: Using SIP RTP TOS bits 184
[2018-03-01 21:32:40] VERBOSE[5340][C-000004a6] netsock2.c: Using SIP RTP CoS mark 5
[2018-03-01 21:32:40] VERBOSE[5340][C-000004a6] app_dial.c: Called SIP/swisscom/0919946769
[2018-03-01 21:32:40] VERBOSE[5340][C-000004a6] app_dial.c: SIP/swisscom-00000f38 is ringing
[2018-03-01 21:32:54] VERBOSE[5340][C-000004a6] app_dial.c: SIP/swisscom-00000f38 answered SIP/302-00000f37
[2018-03-01 21:32:54] VERBOSE[5348][C-000004a6] bridge_channel.c: Channel SIP/swisscom-00000f38 joined ‘simple_bridge’ basic-bridge <4f37d75d-01dc-4eb2-a59c-ce5ac7f06061>
[2018-03-01 21:32:54] VERBOSE[5340][C-000004a6] bridge_channel.c: Channel SIP/302-00000f37 joined ‘simple_bridge’ basic-bridge <4f37d75d-01dc-4eb2-a59c-ce5ac7f06061>
[2018-03-01 21:38:32] VERBOSE[5340][C-000004a6] bridge_channel.c: Channel SIP/302-00000f37 left ‘simple_bridge’ basic-bridge <4f37d75d-01dc-4eb2-a59c-ce5ac7f06061>
[2018-03-01 21:38:32] VERBOSE[5348][C-000004a6] bridge_channel.c: Channel SIP/swisscom-00000f38 left ‘simple_bridge’ basic-bridge <4f37d75d-01dc-4eb2-a59c-ce5ac7f06061>
[2018-03-01 21:38:32] VERBOSE[5340][C-000004a6] app_macro.c: Spawn extension (macro-dialout-trunk, s, 23) exited non-zero on ‘SIP/302-00000f37’ in macro ‘dialout-trunk’
[2018-03-01 21:38:32] VERBOSE[5340][C-000004a6] pbx.c: Spawn extension (from-internal, 0919946769, 6) exited non-zero on ‘SIP/302-00000f37’
[2018-03-01 21:38:32] VERBOSE[5340][C-000004a6] pbx.c: Executing [h@from-internal:1] Macro(“SIP/302-00000f37”, “hangupcall”) in new stack
[2018-03-01 21:38:32] VERBOSE[5340][C-000004a6] pbx.c: Executing [s@macro-hangupcall:1] GotoIf(“SIP/302-00000f37”, “1?theend”) in new stack
[2018-03-01 21:38:32] VERBOSE[5340][C-000004a6] pbx_builtins.c: Goto (macro-hangupcall,s,3)

Another point:
The Swisscom router has also an analog telephone port. If I connect an analog telephone to that port (bypassing Freepbx) I have no problems.

Thanks!
Paolo

Stewart1 · March 2, 2018, 5:57pm

Your issue seems similar to Can not make an outbound (mobile carrier drops call after one ring). It may even be the same; do you know whether VoLTE is involved? Do you know whether the called phone is still with Vodaphone? Have you called other IT mobiles (Tim, Wind, H3G) successfully?

Capture a SIP trace by issuing
sip set debug on
at the Asterisk command line, which will add a SIP trace to the normal Asterisk log file. Make a failing call and post the trace. (Edit it first to mask names, phone numbers, account numbers, public IP addresses and any other information that you consider personal. Replace the redacted information with xxxxxx so we can we what has been changed.)

How does your Swisscom router connect to the internet? If there is a separate modem or ONT, it should be possible to capture the SIP traffic when you call from the Analog port, to see what is different.

Also, I’m curious what Swisscom’s rate is for this call. (If you can’t easily solve the problem, another trunk provider may be a workaround, and be less expensive as well.)

Paolodic · March 7, 2018, 8:38pm

Ho Stewart1,
Sorry for my late answer. The email went to a wrong gmail cathegory.
I discovered the issue with Vodafone but the same happens with TIM.
The Swisscom router has 2 analog subscribers directly on the router itself, so it is very difficult to trace the calls.
I will try to activate the trace of the SIP calls directly on the PBX, as you suggested, and the post the results.

Cheers
Paolo

Stewart1 · March 8, 2018, 9:30am

Yes, but is the (fiber, xDSL or cable) modem built into the router? If it is a separate device, it should be possible by using a managed switch or an old dumb hub, to capture traffic between router and modem.

Or, there may be some syslog or other debugging features in the router that will give us a SIP trace.

Otherwise, we await your trace from the PBX.

Paolodic · March 8, 2018, 12:57pm

It’s just a DSL router (1 box), no way to trace the analog calls and even no way to get some log files from it…!
As soon as i can, i will make the trace as suggested.
tnx

Paolodic · March 9, 2018, 8:19pm

Finally I have the los (part 1):

<------------->
[2018-03-08 22:04:07] VERBOSE[1926] chan_sip.c: — (9 headers 0 lines) —
[2018-03-08 22:04:07] VERBOSE[1926] chan_sip.c: Really destroying SIP dialog ‘[email protected]:5060’ Method: OPTIONS
[2018-03-08 22:04:13] VERBOSE[1926] chan_sip.c:
<— SIP read from UDP:192.168.1.202:5060 —>
INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.202:5060;branch=z9hG4bK8b6e1343629f2387b731d8ed9aa65f26;rport
From: “Gigaset” sip:[email protected];tag=2203342263
To: sip:[email protected];user=phone
Call-ID: 2928297543@192_168_1_202
CSeq: 2 INVITE
Contact: sip:[email protected]:5060
Max-Forwards: 70
User-Agent: E630A GO/42.245.00.000.000
Supported: replaces
Allow-Events: message-summary, refer, ua-profile, talk, check-sync
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY, REFER, UPDATE
Content-Type: application/sdp
Content-Length: 381

v=0
o=302 5004 8 IN IP4 192.168.1.202
s=Mapping
c=IN IP4 192.168.1.202
t=0 0
m=audio 5004 RTP/AVP 9 8 0 96 97 2 18 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:96 G726-32/8000
a=rtpmap:97 AAL2-G726-32/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
<------------->
[2018-03-08 22:04:13] VERBOSE[1926] chan_sip.c: — (14 headers 17 lines) —
[2018-03-08 22:04:13] VERBOSE[1926] chan_sip.c: Sending to 192.168.1.202:5060 (NAT)
[2018-03-08 22:04:13] VERBOSE[1926][C-0000002f] chan_sip.c: Sending to 192.168.1.202:5060 (NAT)
[2018-03-08 22:04:13] VERBOSE[1926][C-0000002f] chan_sip.c: Using INVITE request as basis request - 2928297543@192_168_1_202
[2018-03-08 22:04:13] VERBOSE[1926][C-0000002f] chan_sip.c: Found peer ‘302’ for ‘302’ from 192.168.1.202:5060
[2018-03-08 22:04:13] VERBOSE[1926][C-0000002f] chan_sip.c:
<— Reliably Transmitting (no NAT) to 192.168.1.202:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.202:5060;branch=z9hG4bK8b6e1343629f2387b731d8ed9aa65f26;received=192.168.1.202;rport=5060
From: “Gigaset” sip:[email protected];tag=2203342263
To: sip:[email protected];user=phone;tag=as426c39e8
Call-ID: 2928297543@192_168_1_202
CSeq: 2 INVITE
Server: FPBX-13.0.194.2(13.13.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="3c798139"
Content-Length: 0

<------------>
[2018-03-08 22:04:13] VERBOSE[1926][C-0000002f] chan_sip.c: Scheduling destruction of SIP dialog ‘2928297543@192_168_1_202’ in 6400 ms (Method: INVITE)
[2018-03-08 22:04:13] VERBOSE[1926] chan_sip.c:
<— SIP read from UDP:192.168.1.202:5060 —>
ACK sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.202:5060;branch=z9hG4bK8b6e1343629f2387b731d8ed9aa65f26;rport
From: “Gigaset” sip:[email protected];tag=2203342263
To: sip:[email protected];user=phone;tag=as426c39e8
Call-ID: 2928297543@192_168_1_202
CSeq: 2 ACK
Contact: sip:[email protected]:5060
Max-Forwards: 70
User-Agent: E630A GO/42.245.00.000.000
Content-Length: 0

<------------->
[2018-03-08 22:04:13] VERBOSE[1926] chan_sip.c: — (10 headers 0 lines) —
[2018-03-08 22:04:13] VERBOSE[1926] chan_sip.c:
<— SIP read from UDP:192.168.1.202:5060 —>
INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.202:5060;branch=z9hG4bK666f0652b473a73d839127b14c1e21e;rport
From: “Gigaset” sip:[email protected];tag=2203342263
To: sip:[email protected];user=phone
Call-ID: 2928297543@192_168_1_202
CSeq: 3 INVITE
Contact: sip:[email protected]:5060
Authorization: Digest username=“302”, realm=“asterisk”, algorithm=MD5, uri="sip:[email protected];user=phone", nonce=“3c798139”, response="753a05b96cb7c802df1642869e0af7d3"
Max-Forwards: 70
User-Agent: E630A GO/42.245.00.000.000
Supported: replaces
Allow-Events: message-summary, refer, ua-profile, talk, check-sync
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY, REFER, UPDATE
Content-Type: application/sdp
Content-Length: 381

v=0
o=302 5004 8 IN IP4 192.168.1.202
s=Mapping
c=IN IP4 192.168.1.202
t=0 0
m=audio 5004 RTP/AVP 9 8 0 96 97 2 18 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:96 G726-32/8000
a=rtpmap:97 AAL2-G726-32/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
<------------->
[2018-03-08 22:04:13] VERBOSE[1926] chan_sip.c: — (15 headers 17 lines) —
[2018-03-08 22:04:13] VERBOSE[1926][C-0000002f] chan_sip.c: Sending to 192.168.1.202:5060 (no NAT)
[2018-03-08 22:04:13] VERBOSE[1926][C-0000002f] chan_sip.c: Using INVITE request as basis request - 2928297543@192_168_1_202
[2018-03-08 22:04:13] VERBOSE[1926][C-0000002f] chan_sip.c: Found peer ‘302’ for ‘302’ from 192.168.1.202:5060
[2018-03-08 22:04:13] VERBOSE[1926][C-0000002f] netsock2.c: Using SIP RTP TOS bits 184
[2018-03-08 22:04:13] VERBOSE[1926][C-0000002f] netsock2.c: Using SIP RTP CoS mark 5
[2018-03-08 22:04:13] VERBOSE[1926][C-0000002f] chan_sip.c: Found RTP audio format 9
[2018-03-08 22:04:13] VERBOSE[1926][C-0000002f] chan_sip.c: Found RTP audio format 8
[2018-03-08 22:04:13] VERBOSE[1926][C-0000002f] chan_sip.c: Found RTP audio format 0
[2018-03-08 22:04:13] VERBOSE[1926][C-0000002f] chan_sip.c: Found RTP audio format 96
[2018-03-08 22:04:13] VERBOSE[1926][C-0000002f] chan_sip.c: Found RTP audio format 97
[2018-03-08 22:04:13] VERBOSE[1926][C-0000002f] chan_sip.c: Found RTP audio format 2
[2018-03-08 22:04:13] VERBOSE[1926][C-0000002f] chan_sip.c: Found RTP audio format 18
[2018-03-08 22:04:13] VERBOSE[1926][C-0000002f] chan_sip.c: Found RTP audio format 101
[2018-03-08 22:04:13] VERBOSE[1926][C-0000002f] chan_sip.c: Found audio description format G722 for ID 9
[2018-03-08 22:04:13] VERBOSE[1926][C-0000002f] chan_sip.c: Found audio description format PCMA for ID 8
[2018-03-08 22:04:13] VERBOSE[1926][C-0000002f] chan_sip.c: Found audio description format PCMU for ID 0
[2018-03-08 22:04:13] VERBOSE[1926][C-0000002f] chan_sip.c: Found audio description format G726-32 for ID 96
[2018-03-08 22:04:13] VERBOSE[1926][C-0000002f] chan_sip.c: Found audio description format AAL2-G726-32 for ID 97
[2018-03-08 22:04:13] VERBOSE[1926][C-0000002f] chan_sip.c: Found audio description format G726-32 for ID 2
[2018-03-08 22:04:13] VERBOSE[1926][C-0000002f] chan_sip.c: Found audio description format G729 for ID 18
[2018-03-08 22:04:13] VERBOSE[1926][C-0000002f] chan_sip.c: Found audio description format telephone-event for ID 101
[2018-03-08 22:04:13] VERBOSE[1926][C-0000002f] chan_sip.c: Capabilities: us - (ulaw|alaw|gsm|g726), peer - audio=(ulaw|g726|alaw|g722|g729|g726aal2)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g726)
[2018-03-08 22:04:13] VERBOSE[1926][C-0000002f] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[2018-03-08 22:04:13] VERBOSE[1926][C-0000002f] chan_sip.c: Peer audio RTP is at port 192.168.1.202:5004
[2018-03-08 22:04:13] VERBOSE[1926][C-0000002f] chan_sip.c: Looking for 003934XXXXXXXX in from-internal (domain 192.168.1.1)
[2018-03-08 22:04:13] VERBOSE[1926][C-0000002f] sip/route.c: sip_route_dump: route/path hop: sip:[email protected]:5060
[2018-03-08 22:04:13] VERBOSE[1926][C-0000002f] chan_sip.c:
<— Transmitting (no NAT) to 192.168.1.202:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.202:5060;branch=z9hG4bK666f0652b473a73d839127b14c1e21e;received=192.168.1.202;rport=5060
From: “Gigaset” sip:[email protected];tag=2203342263
To: sip:[email protected];user=phone
Call-ID: 2928297543@192_168_1_202
CSeq: 3 INVITE
Server: FPBX-13.0.194.2(13.13.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Length: 0

<------------>
[2018-03-08 22:04:13] VERBOSE[2544][C-0000002f] pbx.c: Executing [003934XXXXXXXX@from-internal:1] Macro(“SIP/302-0000009c”, “user-callerid,LIMIT,EXTERNAL,”) in new stack
[2018-03-08 22:04:13] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@macro-user-callerid:1] Set(“SIP/302-0000009c”, “TOUCH_MONITOR=1520543053.156”) in new stack
[2018-03-08 22:04:13] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@macro-user-callerid:2] Set(“SIP/302-0000009c”, “AMPUSER=302”) in new stack
[2018-03-08 22:04:13] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@macro-user-callerid:3] GotoIf(“SIP/302-0000009c”, “0?report”) in new stack
[2018-03-08 22:04:13] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@macro-user-callerid:4] ExecIf(“SIP/302-0000009c”, “1?Set(REALCALLERIDNUM=302)”) in new stack
[2018-03-08 22:04:13] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@macro-user-callerid:5] Set(“SIP/302-0000009c”, “AMPUSER=302”) in new stack
[2018-03-08 22:04:13] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@macro-user-callerid:6] GotoIf(“SIP/302-0000009c”, “0?limit”) in new stack
[2018-03-08 22:04:13] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@macro-user-callerid:7] Set(“SIP/302-0000009c”, “AMPUSERCIDNAME=PT (302)”) in new stack
[2018-03-08 22:04:13] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@macro-user-callerid:8] GotoIf(“SIP/302-0000009c”, “0?report”) in new stack
[2018-03-08 22:04:13] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@macro-user-callerid:9] Set(“SIP/302-0000009c”, “AMPUSERCID=302”) in new stack
[2018-03-08 22:04:13] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@macro-user-callerid:10] Set(“SIP/302-0000009c”, “__DIAL_OPTIONS=Ttr”) in new stack
[2018-03-08 22:04:13] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@macro-user-callerid:11] Set(“SIP/302-0000009c”, “CALLERID(all)=“PT (302)” <302>”) in new stack
[2018-03-08 22:04:13] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@macro-user-callerid:12] GotoIf(“SIP/302-0000009c”, “0?limit”) in new stack
[2018-03-08 22:04:13] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@macro-user-callerid:13] ExecIf(“SIP/302-0000009c”, “1?Set(GROUP(concurrency_limit)=302)”) in new stack
[2018-03-08 22:04:13] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@macro-user-callerid:14] NoOp(“SIP/302-0000009c”, “Macro Depth is 1”) in new stack
[2018-03-08 22:04:13] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@macro-user-callerid:15] GotoIf(“SIP/302-0000009c”, “1?report2:macroerror”) in new stack
[2018-03-08 22:04:13] VERBOSE[2544][C-0000002f] pbx_builtins.c: Goto (macro-user-callerid,s,16)
[2018-03-08 22:04:13] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@macro-user-callerid:16] GotoIf(“SIP/302-0000009c”, “1?continue”) in new stack
[2018-03-08 22:04:13] VERBOSE[2544][C-0000002f] pbx_builtins.c: Goto (macro-user-callerid,s,34)
[2018-03-08 22:04:13] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@macro-user-callerid:34] Set(“SIP/302-0000009c”, “CALLERID(number)=302”) in new stack
[2018-03-08 22:04:13] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@macro-user-callerid:35] Set(“SIP/302-0000009c”, “CALLERID(name)=PT (302)”) in new stack
[2018-03-08 22:04:13] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@macro-user-callerid:36] GotoIf(“SIP/302-0000009c”, “0?cnum”) in new stack
[2018-03-08 22:04:13] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@macro-user-callerid:37] Set(“SIP/302-0000009c”, “CDR(cnam)=PT (302)”) in new stack
[2018-03-08 22:04:13] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@macro-user-callerid:38] Set(“SIP/302-0000009c”, “CDR(cnum)=302”) in new stack
[2018-03-08 22:04:13] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@macro-user-callerid:39] Set(“SIP/302-0000009c”, “CHANNEL(language)=it”) in new stack
[2018-03-08 22:04:13] VERBOSE[2544][C-0000002f] pbx.c: Executing [003934XXXXXXXX@from-internal:2] Gosub(“SIP/302-0000009c”, “sub-record-check,s,1(out,003934XXXXXXXX,never)”) in new stack
[2018-03-08 22:04:13] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@sub-record-check:1] GotoIf(“SIP/302-0000009c”, “0?initialized”) in new stack
[2018-03-08 22:04:13] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@sub-record-check:2] Set(“SIP/302-0000009c”, “__REC_STATUS=INITIALIZED”) in new stack
[2018-03-08 22:04:13] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@sub-record-check:3] Set(“SIP/302-0000009c”, “NOW=1520543053”) in new stack
[2018-03-08 22:04:13] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@sub-record-check:4] Set(“SIP/302-0000009c”, “__DAY=08”) in new stack
[2018-03-08 22:04:13] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@sub-record-check:5] Set(“SIP/302-0000009c”, “__MONTH=03”) in new stack
[2018-03-08 22:04:13] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@sub-record-check:6] Set(“SIP/302-0000009c”, “__YEAR=2018”) in new stack
[2018-03-08 22:04:13] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@sub-record-check:7] Set(“SIP/302-0000009c”, “__TIMESTR=20180308-220413”) in new stack
[2018-03-08 22:04:13] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@sub-record-check:8] Set(“SIP/302-0000009c”, “__FROMEXTEN=302”) in new stack
[2018-03-08 22:04:13] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@sub-record-check:9] Set(“SIP/302-0000009c”, “__MON_FMT=wav”) in new stack
[2018-03-08 22:04:13] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@sub-record-check:10] NoOp(“SIP/302-0000009c”, “Recordings initialized”) in new stack
[2018-03-08 22:04:13] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@sub-record-check:11] ExecIf(“SIP/302-0000009c”, “0?Set(ARG3=dontcare)”) in new stack
[2018-03-08 22:04:13] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@sub-record-check:12] Set(“SIP/302-0000009c”, “REC_POLICY_MODE_SAVE=”) in new stack
[2018-03-08 22:04:13] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@sub-record-check:13] ExecIf(“SIP/302-0000009c”, “0?Set(REC_STATUS=NO)”) in new stack
[2018-03-08 22:04:13] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@sub-record-check:14] GotoIf(“SIP/302-0000009c”, “3?checkaction”) in new stack
[2018-03-08 22:04:13] VERBOSE[2544][C-0000002f] pbx_builtins.c: Goto (sub-record-check,s,17)
[2018-03-08 22:04:13] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@sub-record-check:17] GotoIf(“SIP/302-0000009c”, “1?sub-record-check,out,1”) in new stack
[2018-03-08 22:04:13] VERBOSE[2544][C-0000002f] pbx_builtins.c: Goto (sub-record-check,out,1)
[2018-03-08 22:04:13] VERBOSE[2544][C-0000002f] pbx.c: Executing [out@sub-record-check:1] NoOp(“SIP/302-0000009c”, “Outbound Recording Check from 302 to 003934XXXXXXXX”) in new stack
[2018-03-08 22:04:13] VERBOSE[2544][C-0000002f] pbx.c: Executing [out@sub-record-check:2] Set(“SIP/302-0000009c”, “RECMODE=dontcare”) in new stack
[2018-03-08 22:04:13] VERBOSE[2544][C-0000002f] pbx.c: Executing [out@sub-record-check:3] ExecIf(“SIP/302-0000009c”, “1?Goto(routewins)”) in new stack
[2018-03-08 22:04:13] VERBOSE[2544][C-0000002f] pbx_builtins.c: Goto (sub-record-check,out,7)
[2018-03-08 22:04:13] VERBOSE[2544][C-0000002f] pbx.c: Executing [out@sub-record-check:7] Gosub(“SIP/302-0000009c”, “recordcheck,1(never,out,003934XXXXXXXX)”) in new stack
[2018-03-08 22:04:13] VERBOSE[2544][C-0000002f] pbx.c: Executing [recordcheck@sub-record-check:1] NoOp(“SIP/302-0000009c”, “Starting recording check against never”) in new stack
[2018-03-08 22:04:13] VERBOSE[2544][C-0000002f] pbx.c: Executing [recordcheck@sub-record-check:2] Goto(“SIP/302-0000009c”, “never”) in new stack
[2018-03-08 22:04:13] VERBOSE[2544][C-0000002f] pbx_builtins.c: Goto (sub-record-check,recordcheck,14)
[2018-03-08 22:04:13] VERBOSE[2544][C-0000002f] pbx.c: Executing [recordcheck@sub-record-check:14] Set(“SIP/302-0000009c”, “__REC_POLICY_MODE=NEVER”) in new stack
[2018-03-08 22:04:13] VERBOSE[2544][C-0000002f] pbx.c: Executing [recordcheck@sub-record-check:15] Goto(“SIP/302-0000009c”, “stoprec”) in new stack
[2018-03-08 22:04:13] VERBOSE[2544][C-0000002f] pbx_builtins.c: Goto (sub-record-check,recordcheck,25)
[2018-03-08 22:04:13] VERBOSE[2544][C-0000002f] pbx.c: Executing [recordcheck@sub-record-check:25] NoOp(“SIP/302-0000009c”, “Stopping recording: out, 003934XXXXXXXX”) in new stack
[2018-03-08 22:04:13] VERBOSE[2544][C-0000002f] pbx.c: Executing [recordcheck@sub-record-check:26] Set(“SIP/302-0000009c”, “__REC_STATUS=STOPPED”) in new stack
[2018-03-08 22:04:13] VERBOSE[2544][C-0000002f] pbx.c: Executing [recordcheck@sub-record-check:27] System(“SIP/302-0000009c”, “/var/lib/asterisk/bin/stoprecording.php “SIP/302-0000009c””) in new stack
[2018-03-08 22:04:14] VERBOSE[2544][C-0000002f] pbx.c: Executing [recordcheck@sub-record-check:28] Return(“SIP/302-0000009c”, “”) in new stack
[2018-03-08 22:04:14] VERBOSE[2544][C-0000002f] pbx.c: Executing [out@sub-record-check:8] Return(“SIP/302-0000009c”, “”) in new stack
[2018-03-08 22:04:14] VERBOSE[2544][C-0000002f] pbx.c: Executing [003934XXXXXXXX@from-internal:3] Set(“SIP/302-0000009c”, “MOHCLASS=default”) in new stack
[2018-03-08 22:04:14] VERBOSE[2544][C-0000002f] pbx.c: Executing [003934XXXXXXXX@from-internal:4] ExecIf(“SIP/302-0000009c”, “1?Set(TRUNKCIDOVERRIDE=<220>)”) in new stack
[2018-03-08 22:04:14] VERBOSE[2544][C-0000002f] pbx.c: Executing [003934XXXXXXXX@from-internal:5] Set(“SIP/302-0000009c”, “_NODEST=”) in new stack
[2018-03-08 22:04:14] VERBOSE[2544][C-0000002f] pbx.c: Executing [003934XXXXXXXX@from-internal:6] Macro(“SIP/302-0000009c”, “dialout-trunk,3,003934XXXXXXXX,off”) in new stack
[2018-03-08 22:04:14] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@macro-dialout-trunk:1] Set(“SIP/302-0000009c”, “DIAL_TRUNK=3”) in new stack
[2018-03-08 22:04:14] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@macro-dialout-trunk:2] GosubIf(“SIP/302-0000009c”, “0?sub-pincheck,s,1()”) in new stack
[2018-03-08 22:04:14] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@macro-dialout-trunk:3] GotoIf(“SIP/302-0000009c”, “0?disabletrunk,1”) in new stack
[2018-03-08 22:04:14] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@macro-dialout-trunk:4] Set(“SIP/302-0000009c”, “DIAL_NUMBER=003934XXXXXXXX”) in new stack
[2018-03-08 22:04:14] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@macro-dialout-trunk:5] Set(“SIP/302-0000009c”, “DIAL_TRUNK_OPTIONS=Ttr”) in new stack
[2018-03-08 22:04:14] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@macro-dialout-trunk:6] Set(“SIP/302-0000009c”, “OUTBOUND_GROUP=OUT_3”) in new stack
[2018-03-08 22:04:14] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@macro-dialout-trunk:7] Set(“SIP/302-0000009c”, “DIAL_TRUNK_OPTIONS=Tt”) in new stack
[2018-03-08 22:04:14] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@macro-dialout-trunk:8] GotoIf(“SIP/302-0000009c”, “0?nomax”) in new stack
[2018-03-08 22:04:14] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@macro-dialout-trunk:9] GotoIf(“SIP/302-0000009c”, “0?chanfull”) in new stack
[2018-03-08 22:04:14] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@macro-dialout-trunk:10] GotoIf(“SIP/302-0000009c”, “0?skipoutcid”) in new stack
[2018-03-08 22:04:14] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@macro-dialout-trunk:11] Macro(“SIP/302-0000009c”, “outbound-callerid,3”) in new stack
[2018-03-08 22:04:14] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@macro-outbound-callerid:1] ExecIf(“SIP/302-0000009c”, “0?Set(CALLERPRES(name-pres)=)”) in new stack
[2018-03-08 22:04:14] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@macro-outbound-callerid:2] ExecIf(“SIP/302-0000009c”, “0?Set(CALLERPRES(num-pres)=)”) in new stack
[2018-03-08 22:04:14] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@macro-outbound-callerid:3] ExecIf(“SIP/302-0000009c”, “0?Set(REALCALLERIDNUM=302)”) in new stack
[2018-03-08 22:04:14] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@macro-outbound-callerid:4] GotoIf(“SIP/302-0000009c”, “1?normcid”) in new stack
[2018-03-08 22:04:14] VERBOSE[2544][C-0000002f] pbx_builtins.c: Goto (macro-outbound-callerid,s,7)
[2018-03-08 22:04:14] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@macro-outbound-callerid:7] Set(“SIP/302-0000009c”, “USEROUTCID=”) in new stack
[2018-03-08 22:04:14] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@macro-outbound-callerid:8] Set(“SIP/302-0000009c”, “EMERGENCYCID=”) in new stack
[2018-03-08 22:04:14] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@macro-outbound-callerid:9] Set(“SIP/302-0000009c”, “TRUNKOUTCID=<004191YYYYYYY>”) in new stack
[2018-03-08 22:04:14] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@macro-outbound-callerid:10] GotoIf(“SIP/302-0000009c”, “1?trunkcid”) in new stack
[2018-03-08 22:04:14] VERBOSE[2544][C-0000002f] pbx_builtins.c: Goto (macro-outbound-callerid,s,15)
[2018-03-08 22:04:14] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@macro-outbound-callerid:15] ExecIf(“SIP/302-0000009c”, “1?Set(CALLERID(all)=<004191YYYYYYY>)”) in new stack
[2018-03-08 22:04:14] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@macro-outbound-callerid:16] ExecIf(“SIP/302-0000009c”, “0?Set(CALLERID(all)=)”) in new stack
[2018-03-08 22:04:14] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@macro-outbound-callerid:17] ExecIf(“SIP/302-0000009c”, “1?Set(CALLERID(all)=<220>)”) in new stack
[2018-03-08 22:04:14] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@macro-outbound-callerid:18] ExecIf(“SIP/302-0000009c”, “0?Set(CALLERPRES(name-pres)=prohib_passed_screen)”) in new stack
[2018-03-08 22:04:14] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@macro-outbound-callerid:19] ExecIf(“SIP/302-0000009c”, “0?Set(CALLERPRES(num-pres)=prohib_passed_screen)”) in new stack
[2018-03-08 22:04:14] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@macro-outbound-callerid:20] Set(“SIP/302-0000009c”, “CDR(outbound_cnum)=220”) in new stack
[2018-03-08 22:04:14] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@macro-outbound-callerid:21] Set(“SIP/302-0000009c”, “CDR(outbound_cnam)=”) in new stack
[2018-03-08 22:04:14] WARNING[1880] func_cdr.c: CDR requires a value (CDR(variable)=value)
)[2018-03-08 22:04:14] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@macro-dialout-trunk:12] GosubIf(“SIP/302-0000009c”, “0?sub-flp-3,s,1()”) in new stack
[2018-03-08 22:04:14] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@macro-dialout-trunk:13] Set(“SIP/302-0000009c”, “OUTNUM=003934XXXXXXXX”) in new stack
[2018-03-08 22:04:14] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@macro-dialout-trunk:14] Set(“SIP/302-0000009c”, “custom=SIP/swisscom”) in new stack
[2018-03-08 22:04:14] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@macro-dialout-trunk:15] ExecIf(“SIP/302-0000009c”, “0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)Tt)”) in new stack
[2018-03-08 22:04:14] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@macro-dialout-trunk:16] ExecIf(“SIP/302-0000009c”, “0?Set(DIAL_TRUNK_OPTIONS=TtM(confirm))”) in new stack
[2018-03-08 22:04:14] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@macro-dialout-trunk:17] Macro(“SIP/302-0000009c”, “dialout-trunk-predial-hook,”) in new stack
[2018-03-08 22:04:14] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit(“SIP/302-0000009c”, “”) in new stack
[2018-03-08 22:04:14] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@macro-dialout-trunk:18] GotoIf(“SIP/302-0000009c”, “0?bypass,1”) in new stack
[2018-03-08 22:04:14] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@macro-dialout-trunk:19] ExecIf(“SIP/302-0000009c”, “1?Set(CONNECTEDLINE(num,i)=003934XXXXXXXX)”) in new stack
[2018-03-08 22:04:14] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@macro-dialout-trunk:20] ExecIf(“SIP/302-0000009c”, “1?Set(CONNECTEDLINE(name,i)=CID:220)”) in new stack
[2018-03-08 22:04:14] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@macro-dialout-trunk:21] ExecIf(“SIP/302-0000009c”, “0?Set(CONNECTEDLINE(name,i)=CID:(Hidden)220)”) in new stack
[2018-03-08 22:04:14] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@macro-dialout-trunk:22] GotoIf(“SIP/302-0000009c”, “0?customtrunk”) in new stack
[2018-03-08 22:04:14] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@macro-dialout-trunk:23] Dial(“SIP/302-0000009c”, “SIP/swisscom/003934XXXXXXXX,300,Tt”) in new stack
[2018-03-08 22:04:14] VERBOSE[2544][C-0000002f] netsock2.c: Using SIP RTP TOS bits 184
[2018-03-08 22:04:14] VERBOSE[2544][C-0000002f] netsock2.c: Using SIP RTP CoS mark 5
[2018-03-08 22:04:14] VERBOSE[2544][C-0000002f] chan_sip.c: Audio is at 11212
[2018-03-08 22:04:14] VERBOSE[2544][C-0000002f] chan_sip.c: Adding codec ulaw to SDP
[2018-03-08 22:04:14] VERBOSE[2544][C-0000002f] chan_sip.c: Adding codec alaw to SDP
[2018-03-08 22:04:14] VERBOSE[2544][C-0000002f] chan_sip.c: Adding codec g722 to SDP
[2018-03-08 22:04:14] VERBOSE[2544][C-0000002f] chan_sip.c: Adding codec g726 to SDP
[2018-03-08 22:04:14] VERBOSE[2544][C-0000002f] chan_sip.c: Adding codec gsm to SDP
[2018-03-08 22:04:14] VERBOSE[2544][C-0000002f] chan_sip.c: Adding codec g729 to SDP
[2018-03-08 22:04:14] VERBOSE[2544][C-0000002f] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[2018-03-08 22:04:14] VERBOSE[2544][C-0000002f] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.1:5060:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK107daa34;rport
Max-Forwards: 70
From: sip:[email protected];tag=as68e5ce64
To: sip:[email protected]:5060
Contact: sip:[email protected]:5060
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: FPBX-13.0.194.2(13.13.1)
Date: Thu, 08 Mar 2018 21:04:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 403

v=0
o=root 1631960659 1631960659 IN IP4 192.168.1.201
s=Asterisk PBX 13.13.1
c=IN IP4 192.168.1.201
t=0 0
m=audio 11212 RTP/AVP 0 8 9 111 3 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

[2018-03-08 22:04:14] VERBOSE[2544][C-0000002f] app_dial.c: Called SIP/swisscom/003934XXXXXXXX
[2018-03-08 22:04:14] VERBOSE[1926] chan_sip.c:
<— SIP read from UDP:192.168.1.1:5060 —>
SIP/2.0 100 Trying
From: sip:[email protected];tag=as68e5ce64
To: sip:[email protected]:5060
Call-ID: [email protected]
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.1.201:5060;rport=5060;branch=z9hG4bK107daa34
Supported: replaces,timer
X-Serialnumber: 2.4T1537A0107841
User-Agent: SAH / 2.4 / v08.08.28.00
Contact: sip:[email protected]:5060
Content-Length: 0

<------------->
[2018-03-08 22:04:14] VERBOSE[1926] chan_sip.c: — (11 headers 0 lines) —
[2018-03-08 22:04:14] VERBOSE[1926] chan_sip.c:
<— SIP read from UDP:192.168.1.1:5060 —>
SIP/2.0 407 Proxy Authentication Required
From: sip:[email protected];tag=as68e5ce64
To: sip:[email protected]:5060;tag=e48288-55050bfc-13c4-55013-7f446-31ac0596-7f446
Call-ID: [email protected]
CSeq: 102 INVITE
Proxy-Authenticate: Digest realm=“Home”,domain=“b2b.domain”,nonce=“4ba6b43ae482885bfa0d36”,stale=true,algorithm=MD5,qop="auth"
Via: SIP/2.0/UDP 192.168.1.201:5060;rport=5060;branch=z9hG4bK107daa34
Supported: replaces,timer
X-Serialnumber: 2.4T1537A0107841
User-Agent: SAH / 2.4 / v08.08.28.00
Allow: INVITE, ACK, BYE, NOTIFY, INFO, CANCEL, MESSAGE, OPTIONS
Content-Length: 0

<------------->
[2018-03-08 22:04:14] VERBOSE[1926] chan_sip.c: — (12 headers 0 lines) —
[2018-03-08 22:04:14] VERBOSE[1926][C-0000002f] chan_sip.c: Transmitting (NAT) to 192.168.1.1:5060:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK107daa34;rport
Max-Forwards: 70
From: sip:[email protected];tag=as68e5ce64
To: sip:[email protected]:5060;tag=e48288-55050bfc-13c4-55013-7f446-31ac0596-7f446
Contact: sip:[email protected]:5060
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: FPBX-13.0.194.2(13.13.1)
Content-Length: 0

[2018-03-08 22:04:14] VERBOSE[1926][C-0000002f] chan_sip.c: Audio is at 11212
[2018-03-08 22:04:14] VERBOSE[1926][C-0000002f] chan_sip.c: Adding codec ulaw to SDP
[2018-03-08 22:04:14] VERBOSE[1926][C-0000002f] chan_sip.c: Adding codec alaw to SDP
[2018-03-08 22:04:14] VERBOSE[1926][C-0000002f] chan_sip.c: Adding codec g722 to SDP
[2018-03-08 22:04:14] VERBOSE[1926][C-0000002f] chan_sip.c: Adding codec g726 to SDP
[2018-03-08 22:04:14] VERBOSE[1926][C-0000002f] chan_sip.c: Adding codec gsm to SDP
[2018-03-08 22:04:14] VERBOSE[1926][C-0000002f] chan_sip.c: Adding codec g729 to SDP
[2018-03-08 22:04:14] VERBOSE[1926][C-0000002f] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[2018-03-08 22:04:14] VERBOSE[1926][C-0000002f] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.1:5060:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK35c3db48;rport
Max-Forwards: 70
From: sip:[email protected];tag=as68e5ce64
To: sip:[email protected]:5060
Contact: sip:[email protected]:5060
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: FPBX-13.0.194.2(13.13.1)
Proxy-Authorization: Digest username=“220”, realm=“Home”, algorithm=MD5, uri=“sip:b2b.domain”, nonce=“4ba6b43ae482885bfa0d36”, response=“c4a1f6fc8fe30d8d205d877cc48b161f”, qop=auth, cnonce=“792adb61”, nc=00000001
Date: Thu, 08 Mar 2018 21:04:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 403

v=0
o=root 1631960659 1631960660 IN IP4 192.168.1.201
s=Asterisk PBX 13.13.1
c=IN IP4 192.168.1.201
t=0 0
m=audio 11212 RTP/AVP 0 8 9 111 3 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

[2018-03-08 22:04:14] VERBOSE[1926] chan_sip.c:
<— SIP read from UDP:192.168.1.1:5060 —>
SIP/2.0 100 Trying
From: sip:[email protected];tag=as68e5ce64
To: sip:[email protected]:5060
Call-ID: [email protected]
CSeq: 103 INVITE
Via: SIP/2.0/UDP 192.168.1.201:5060;rport=5060;branch=z9hG4bK35c3db48
Supported: replaces,timer
X-Serialnumber: 2.4T1537A0107841
User-Agent: SAH / 2.4 / v08.08.28.00
Contact: sip:[email protected]:5060
Content-Length: 0

Paolodic · March 9, 2018, 8:20pm

And part 2:

<------------->
[2018-03-08 22:04:14] VERBOSE[1926] chan_sip.c: — (11 headers 0 lines) —
[2018-03-08 22:04:15] VERBOSE[1926] chan_sip.c:
<— SIP read from UDP:192.168.1.1:5060 —>
SIP/2.0 180 Ringing
From: sip:[email protected];tag=as68e5ce64
To: sip:[email protected]:5060;tag=e49cc8-55050bfc-13c4-55013-7f447-43fbafc2-7f447
Call-ID: [email protected]
CSeq: 103 INVITE
Via: SIP/2.0/UDP 192.168.1.201:5060;rport=5060;branch=z9hG4bK35c3db48
Supported: replaces,timer
X-Serialnumber: 2.4T1537A0107841
User-Agent: SAH / 2.4 / v08.08.28.00
Allow: INVITE, ACK, BYE, NOTIFY, INFO, CANCEL, MESSAGE, OPTIONS
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 212

v=0
o=- 1520543188 1520543188 IN IP4 192.168.1.1
s=-
c=IN IP4 192.168.1.1
t=0 0
m=audio 10072 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=ptime:20
<------------->
[2018-03-08 22:04:15] VERBOSE[1926] chan_sip.c: — (13 headers 11 lines) —
[2018-03-08 22:04:15] VERBOSE[1926][C-0000002f] sip/route.c: sip_route_dump: route/path hop: sip:[email protected]:5060
[2018-03-08 22:04:15] VERBOSE[1926][C-0000002f] chan_sip.c: Found RTP audio format 8
[2018-03-08 22:04:15] VERBOSE[1926][C-0000002f] chan_sip.c: Found RTP audio format 101
[2018-03-08 22:04:15] VERBOSE[1926][C-0000002f] chan_sip.c: Found audio description format PCMA for ID 8
[2018-03-08 22:04:15] VERBOSE[1926][C-0000002f] chan_sip.c: Found audio description format telephone-event for ID 101
[2018-03-08 22:04:15] VERBOSE[1926][C-0000002f] chan_sip.c: Capabilities: us - (alaw|g722|g726|gsm|ulaw|g729), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
[2018-03-08 22:04:15] VERBOSE[1926][C-0000002f] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[2018-03-08 22:04:15] VERBOSE[1926][C-0000002f] chan_sip.c: Peer audio RTP is at port 192.168.1.1:10072
[2018-03-08 22:04:15] VERBOSE[2544][C-0000002f] app_dial.c: SIP/swisscom-0000009d is ringing
[2018-03-08 22:04:15] VERBOSE[2544][C-0000002f] chan_sip.c:
<— Transmitting (no NAT) to 192.168.1.202:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.202:5060;branch=z9hG4bK666f0652b473a73d839127b14c1e21e;received=192.168.1.202;rport=5060
From: “Gigaset” sip:[email protected];tag=2203342263
To: sip:[email protected];user=phone;tag=as4758f00e
Call-ID: 2928297543@192_168_1_202
CSeq: 3 INVITE
Server: FPBX-13.0.194.2(13.13.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Length: 0

<------------>
[2018-03-08 22:04:15] VERBOSE[2544][C-0000002f] app_dial.c: SIP/swisscom-0000009d is making progress passing it to SIP/302-0000009c
[2018-03-08 22:04:15] VERBOSE[2544][C-0000002f] chan_sip.c: Audio is at 11926
[2018-03-08 22:04:15] VERBOSE[2544][C-0000002f] chan_sip.c: Adding codec ulaw to SDP
[2018-03-08 22:04:15] VERBOSE[2544][C-0000002f] chan_sip.c: Adding codec alaw to SDP
[2018-03-08 22:04:15] VERBOSE[2544][C-0000002f] chan_sip.c: Adding codec g726 to SDP
[2018-03-08 22:04:15] VERBOSE[2544][C-0000002f] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[2018-03-08 22:04:15] VERBOSE[2544][C-0000002f] chan_sip.c:
<— Transmitting (no NAT) to 192.168.1.202:5060 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.202:5060;branch=z9hG4bK666f0652b473a73d839127b14c1e21e;received=192.168.1.202;rport=5060
From: “Gigaset” sip:[email protected];tag=2203342263
To: sip:[email protected];user=phone;tag=as4758f00e
Call-ID: 2928297543@192_168_1_202
CSeq: 3 INVITE
Server: FPBX-13.0.194.2(13.13.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 303

v=0
o=root 930563529 930563529 IN IP4 192.168.1.201
s=Asterisk PBX 13.13.1
c=IN IP4 192.168.1.201
t=0 0
m=audio 11926 RTP/AVP 0 8 2 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<------------>
[2018-03-08 22:04:18] VERBOSE[1926] chan_sip.c:
<— SIP read from UDP:192.168.1.1:5060 —>
SIP/2.0 180 Ringing
From: sip:[email protected];tag=as68e5ce64
To: sip:[email protected]:5060;tag=e49cc8-55050bfc-13c4-55013-7f447-43fbafc2-7f447
Call-ID: [email protected]
CSeq: 103 INVITE
Via: SIP/2.0/UDP 192.168.1.201:5060;rport=5060;branch=z9hG4bK35c3db48
Supported: replaces,timer
X-Serialnumber: 2.4T1537A0107841
User-Agent: SAH / 2.4 / v08.08.28.00
Allow: INVITE, ACK, BYE, NOTIFY, INFO, CANCEL, MESSAGE, OPTIONS
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 0

<------------->
[2018-03-08 22:04:18] VERBOSE[1926] chan_sip.c: — (13 headers 0 lines) —
[2018-03-08 22:04:18] VERBOSE[1926][C-0000002f] sip/route.c: sip_route_dump: route/path hop: sip:[email protected]:5060
[2018-03-08 22:04:18] VERBOSE[2544][C-0000002f] app_dial.c: SIP/swisscom-0000009d is ringing
[2018-03-08 22:04:22] VERBOSE[1926] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.1.205:5060:
OPTIONS sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK167bbc85
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as79b4b250
To: sip:[email protected]:5060
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-13.0.194.2(13.13.1)
Date: Thu, 08 Mar 2018 21:04:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

[2018-03-08 22:04:22] VERBOSE[1926] chan_sip.c:
<— SIP read from UDP:192.168.1.205:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK167bbc85
From: “Unknown” sip:[email protected];tag=as79b4b250
To: sip:[email protected]:5060;tag=ar68c5c341
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Supported: replaces
User-Agent: S850A GO/42.245.00.000.000
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY, REFER, UPDATE
Accept: application/sdp, application/dtmf-relay, message/sipfrag, application/simple-message-summary, application/url, multipart/mixed
Accept-Encoding: identity
Accept-Language: en
Content-Length: 0

<------------->
[2018-03-08 22:04:22] VERBOSE[1926] chan_sip.c: — (13 headers 0 lines) —
[2018-03-08 22:04:22] VERBOSE[1926] chan_sip.c: Really destroying SIP dialog ‘[email protected]:5060’ Method: OPTIONS
[2018-03-08 22:04:24] NOTICE[1926] chan_sip.c: – Re-registration for [email protected]
[2018-03-08 22:04:24] VERBOSE[1926] chan_sip.c: REGISTER 12 headers, 0 lines
[2018-03-08 22:04:24] VERBOSE[1926] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.1:5060:
REGISTER sip:192.168.1.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK440cdc2d;rport
Max-Forwards: 70
From: sip:[email protected];tag=as752e2d2b
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 10016 REGISTER
Supported: replaces, timer
User-Agent: FPBX-13.0.194.2(13.13.1)
Authorization: Digest username=“220”, realm=“Home”, algorithm=MD5, uri=“sip:b2b.domain”, nonce=“457af6ebe4b0087c8f8114”, response=“125297b149a0af71499f60afcd3db7fe”, qop=auth, cnonce=“4a10db85”, nc=00000002
Expires: 120
Contact: sip:[email protected]:5060
Content-Length: 0

[2018-03-08 22:04:24] VERBOSE[1926] chan_sip.c: Retransmitting #1 (NAT) to 192.168.1.1:5060:
REGISTER sip:192.168.1.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK440cdc2d;rport
Max-Forwards: 70
From: sip:[email protected];tag=as752e2d2b
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 10016 REGISTER
Supported: replaces, timer
User-Agent: FPBX-13.0.194.2(13.13.1)
Authorization: Digest username=“220”, realm=“Home”, algorithm=MD5, uri=“sip:b2b.domain”, nonce=“457af6ebe4b0087c8f8114”, response=“125297b149a0af71499f60afcd3db7fe”, qop=auth, cnonce=“4a10db85”, nc=00000002
Expires: 120
Contact: sip:[email protected]:5060
Content-Length: 0

[2018-03-08 22:04:25] VERBOSE[1926] chan_sip.c: Retransmitting #2 (NAT) to 192.168.1.1:5060:
REGISTER sip:192.168.1.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK440cdc2d;rport
Max-Forwards: 70
From: sip:[email protected];tag=as752e2d2b
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 10016 REGISTER
Supported: replaces, timer
User-Agent: FPBX-13.0.194.2(13.13.1)
Authorization: Digest username=“220”, realm=“Home”, algorithm=MD5, uri=“sip:b2b.domain”, nonce=“457af6ebe4b0087c8f8114”, response=“125297b149a0af71499f60afcd3db7fe”, qop=auth, cnonce=“4a10db85”, nc=00000002
Expires: 120
Contact: sip:[email protected]:5060
Content-Length: 0

[2018-03-08 22:04:27] VERBOSE[1926] chan_sip.c: Retransmitting #3 (NAT) to 192.168.1.1:5060:
REGISTER sip:192.168.1.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK440cdc2d;rport
Max-Forwards: 70
From: sip:[email protected];tag=as752e2d2b
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 10016 REGISTER
Supported: replaces, timer
User-Agent: FPBX-13.0.194.2(13.13.1)
Authorization: Digest username=“220”, realm=“Home”, algorithm=MD5, uri=“sip:b2b.domain”, nonce=“457af6ebe4b0087c8f8114”, response=“125297b149a0af71499f60afcd3db7fe”, qop=auth, cnonce=“4a10db85”, nc=00000002
Expires: 120
Contact: sip:[email protected]:5060
Content-Length: 0

[2018-03-08 22:04:31] VERBOSE[1926] chan_sip.c:
<— SIP read from UDP:192.168.1.202:5060 —>
CANCEL sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.202:5060;branch=z9hG4bK666f0652b473a73d839127b14c1e21e;rport
From: “Gigaset” sip:[email protected];tag=2203342263
To: sip:[email protected];user=phone
Call-ID: 2928297543@192_168_1_202
CSeq: 3 CANCEL
Contact: sip:[email protected]:5060
Authorization: Digest username=“302”, realm=“asterisk”, algorithm=MD5, uri="sip:[email protected];user=phone", nonce=“3c798139”, response="e73c989cf5fe17e6205563ae363f131d"
Max-Forwards: 70
User-Agent: E630A GO/42.245.00.000.000
Content-Length: 0

<------------->
[2018-03-08 22:04:31] VERBOSE[1926] chan_sip.c: — (11 headers 0 lines) —
[2018-03-08 22:04:31] VERBOSE[1926][C-0000002f] chan_sip.c: Sending to 192.168.1.202:5060 (no NAT)
[2018-03-08 22:04:31] VERBOSE[1926][C-0000002f] chan_sip.c:
<— Reliably Transmitting (no NAT) to 192.168.1.202:5060 —>
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.1.202:5060;branch=z9hG4bK666f0652b473a73d839127b14c1e21e;received=192.168.1.202;rport=5060
From: “Gigaset” sip:[email protected];tag=2203342263
To: sip:[email protected];user=phone;tag=as4758f00e
Call-ID: 2928297543@192_168_1_202
CSeq: 3 INVITE
Server: FPBX-13.0.194.2(13.13.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
[2018-03-08 22:04:31] VERBOSE[1926][C-0000002f] chan_sip.c:
<— Transmitting (no NAT) to 192.168.1.202:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.202:5060;branch=z9hG4bK666f0652b473a73d839127b14c1e21e;received=192.168.1.202;rport=5060
From: “Gigaset” sip:[email protected];tag=2203342263
To: sip:[email protected];user=phone;tag=as4758f00e
Call-ID: 2928297543@192_168_1_202
CSeq: 3 CANCEL
Server: FPBX-13.0.194.2(13.13.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
[2018-03-08 22:04:31] VERBOSE[2544][C-0000002f] chan_sip.c: Scheduling destruction of SIP dialog ‘[email protected]’ in 6400 ms (Method: INVITE)
[2018-03-08 22:04:31] VERBOSE[2544][C-0000002f] chan_sip.c: Reliably Transmitting (NAT) to 192.168.1.1:5060:
CANCEL sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK35c3db48;rport
Max-Forwards: 70
From: sip:[email protected];tag=as68e5ce64
To: sip:[email protected]:5060
Call-ID: [email protected]
CSeq: 103 CANCEL
User-Agent: FPBX-13.0.194.2(13.13.1)
Content-Length: 0

[2018-03-08 22:04:31] VERBOSE[2544][C-0000002f] chan_sip.c: Scheduling destruction of SIP dialog ‘[email protected]’ in 6400 ms (Method: INVITE)
[2018-03-08 22:04:31] VERBOSE[2544][C-0000002f] app_macro.c: Spawn extension (macro-dialout-trunk, s, 23) exited non-zero on ‘SIP/302-0000009c’ in macro ‘dialout-trunk’
[2018-03-08 22:04:31] VERBOSE[2544][C-0000002f] pbx.c: Spawn extension (from-internal, 003934XXXXXXXX, 6) exited non-zero on ‘SIP/302-0000009c’
[2018-03-08 22:04:31] VERBOSE[2544][C-0000002f] pbx.c: Executing [h@from-internal:1] Macro(“SIP/302-0000009c”, “hangupcall”) in new stack
[2018-03-08 22:04:31] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@macro-hangupcall:1] GotoIf(“SIP/302-0000009c”, “1?theend”) in new stack
[2018-03-08 22:04:31] VERBOSE[2544][C-0000002f] pbx_builtins.c: Goto (macro-hangupcall,s,3)
[2018-03-08 22:04:31] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@macro-hangupcall:3] ExecIf(“SIP/302-0000009c”, “0?Set(CDR(recordingfile)=)”) in new stack
[2018-03-08 22:04:31] VERBOSE[2544][C-0000002f] pbx.c: Executing [s@macro-hangupcall:4] Hangup(“SIP/302-0000009c”, “”) in new stack
[2018-03-08 22:04:31] VERBOSE[2544][C-0000002f] app_macro.c: Spawn extension (macro-hangupcall, s, 4) exited non-zero on ‘SIP/302-0000009c’ in macro ‘hangupcall’
[2018-03-08 22:04:31] VERBOSE[2544][C-0000002f] pbx.c: Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/302-0000009c’
[2018-03-08 22:04:31] VERBOSE[1926] chan_sip.c:
<— SIP read from UDP:192.168.1.202:5060 —>
ACK sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.202:5060;branch=z9hG4bK666f0652b473a73d839127b14c1e21e;rport
From: “Gigaset” sip:[email protected];tag=2203342263
To: sip:[email protected];user=phone;tag=as4758f00e
Call-ID: 2928297543@192_168_1_202
CSeq: 3 ACK
Contact: sip:[email protected]:5060
Authorization: Digest username=“302”, realm=“asterisk”, algorithm=MD5, uri="sip:[email protected];user=phone", nonce=“3c798139”, response="753a05b96cb7c802df1642869e0af7d3"
Max-Forwards: 70
User-Agent: E630A GO/42.245.00.000.000
Content-Length: 0

<------------->
[2018-03-08 22:04:31] VERBOSE[1926] chan_sip.c: — (11 headers 0 lines) —
[2018-03-08 22:04:31] VERBOSE[1926] chan_sip.c: Really destroying SIP dialog ‘2928297543@192_168_1_202’ Method: ACK
[2018-03-08 22:04:31] VERBOSE[1926] chan_sip.c: Retransmitting #4 (NAT) to 192.168.1.1:5060:
REGISTER sip:192.168.1.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK440cdc2d;rport
Max-Forwards: 70
From: sip:[email protected];tag=as752e2d2b
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 10016 REGISTER
Supported: replaces, timer
User-Agent: FPBX-13.0.194.2(13.13.1)
Authorization: Digest username=“220”, realm=“Home”, algorithm=MD5, uri=“sip:b2b.domain”, nonce=“457af6ebe4b0087c8f8114”, response=“125297b149a0af71499f60afcd3db7fe”, qop=auth, cnonce=“4a10db85”, nc=00000002
Expires: 120
Contact: sip:[email protected]:5060
Content-Length: 0

[2018-03-08 22:04:31] VERBOSE[1926] chan_sip.c: Retransmitting #1 (NAT) to 192.168.1.1:5060:
CANCEL sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK35c3db48;rport
Max-Forwards: 70
From: sip:[email protected];tag=as68e5ce64
To: sip:[email protected]:5060
Call-ID: [email protected]
CSeq: 103 CANCEL
User-Agent: FPBX-13.0.194.2(13.13.1)
Content-Length: 0

[2018-03-08 22:04:31] VERBOSE[1926] chan_sip.c: Retransmitting #2 (NAT) to 192.168.1.1:5060:
CANCEL sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK35c3db48;rport
Max-Forwards: 70
From: sip:[email protected];tag=as68e5ce64
To: sip:[email protected]:5060
Call-ID: [email protected]
CSeq: 103 CANCEL
User-Agent: FPBX-13.0.194.2(13.13.1)
Content-Length: 0

[2018-03-08 22:04:32] VERBOSE[1926] chan_sip.c: Retransmitting #3 (NAT) to 192.168.1.1:5060:
CANCEL sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK35c3db48;rport
Max-Forwards: 70
From: sip:[email protected];tag=as68e5ce64
To: sip:[email protected]:5060
Call-ID: [email protected]
CSeq: 103 CANCEL
User-Agent: FPBX-13.0.194.2(13.13.1)
Content-Length: 0

[2018-03-08 22:04:32] VERBOSE[1926] chan_sip.c: Really destroying SIP dialog ‘3922016170@192_168_1_202’ Method: REGISTER
[2018-03-08 22:04:32] VERBOSE[1926] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.1.204:5060:
OPTIONS sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK0efaa099
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as349aed02
To: sip:[email protected]:5060
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-13.0.194.2(13.13.1)
Date: Thu, 08 Mar 2018 21:04:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

[2018-03-08 22:04:32] VERBOSE[1926] chan_sip.c:
<— SIP read from UDP:192.168.1.204:5060 —>
SIP/2.0 200 OK
To: sip:[email protected]:5060;tag=7d0dad5f4438d851i0
From: “Unknown” sip:[email protected];tag=as349aed02
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK0efaa099
Server: Cisco/SPA301-7.5.2
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, UPDATE
Supported: replaces

<------------->
[2018-03-08 22:04:32] VERBOSE[1926] chan_sip.c: — (10 headers 0 lines) —
[2018-03-08 22:04:32] VERBOSE[1926] chan_sip.c: Really destroying SIP dialog ‘[email protected]:5060’ Method: OPTIONS
[2018-03-08 22:04:32] VERBOSE[1926] chan_sip.c: Really destroying SIP dialog ‘2711781970@192_168_1_202’ Method: REGISTER
[2018-03-08 22:04:33] VERBOSE[1926] chan_sip.c: Retransmitting #4 (NAT) to 192.168.1.1:5060:
CANCEL sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK35c3db48;rport
Max-Forwards: 70
From: sip:[email protected];tag=as68e5ce64
To: sip:[email protected]:5060
Call-ID: [email protected]
CSeq: 103 CANCEL
User-Agent: FPBX-13.0.194.2(13.13.1)
Content-Length: 0

[2018-03-08 22:04:34] VERBOSE[1926] chan_sip.c: Retransmitting #5 (NAT) to 192.168.1.1:5060:
CANCEL sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK35c3db48;rport
Max-Forwards: 70
From: sip:[email protected];tag=as68e5ce64
To: sip:[email protected]:5060
Call-ID: [email protected]
CSeq: 103 CANCEL
User-Agent: FPBX-13.0.194.2(13.13.1)
Content-Length: 0

[2018-03-08 22:04:35] VERBOSE[1926] chan_sip.c: Retransmitting #5 (NAT) to 192.168.1.1:5060:
REGISTER sip:192.168.1.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK440cdc2d;rport
Max-Forwards: 70
From: sip:[email protected];tag=as752e2d2b
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 10016 REGISTER
Supported: replaces, timer
User-Agent: FPBX-13.0.194.2(13.13.1)
Authorization: Digest username=“220”, realm=“Home”, algorithm=MD5, uri=“sip:b2b.domain”, nonce=“457af6ebe4b0087c8f8114”, response=“125297b149a0af71499f60afcd3db7fe”, qop=auth, cnonce=“4a10db85”, nc=00000002
Expires: 120
Contact: sip:[email protected]:5060
Content-Length: 0

[2018-03-08 22:04:37] VERBOSE[1926] chan_sip.c: Retransmitting #6 (NAT) to 192.168.1.1:5060:
CANCEL sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK35c3db48;rport
Max-Forwards: 70
From: sip:[email protected];tag=as68e5ce64
To: sip:[email protected]:5060
Call-ID: [email protected]
CSeq: 103 CANCEL
User-Agent: FPBX-13.0.194.2(13.13.1)
Content-Length: 0

[2018-03-08 22:04:37] WARNING[1926] chan_sip.c: Retransmission timeout reached on transmission [email protected] for seqno 103 (Non-critical Request) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6399ms with no response
[2018-03-08 22:04:37] WARNING[1926] chan_sip.c: Timeout on [email protected] on non-critical invite transaction.
[2018-03-08 22:04:39] VERBOSE[1926] chan_sip.c: Retransmitting #6 (NAT) to 192.168.1.1:5060:
REGISTER sip:192.168.1.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK440cdc2d;rport
Max-Forwards: 70
From: sip:[email protected];tag=as752e2d2b
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 10016 REGISTER
Supported: replaces, timer
User-Agent: FPBX-13.0.194.2(13.13.1)
Authorization: Digest username=“220”, realm=“Home”, algorithm=MD5, uri=“sip:b2b.domain”, nonce=“457af6ebe4b0087c8f8114”, response=“125297b149a0af71499f60afcd3db7fe”, qop=auth, cnonce=“4a10db85”, nc=00000002
Expires: 120
Contact: sip:[email protected]:5060
Content-Length: 0

[2018-03-08 22:04:39] VERBOSE[1926] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.1.203:5060:
OPTIONS sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK1086245b
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as280787ef
To: sip:[email protected]:5060
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-13.0.194.2(13.13.1)
Date: Thu, 08 Mar 2018 21:04:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

[2018-03-08 22:04:39] VERBOSE[1926] chan_sip.c:
<— SIP read from UDP:192.168.1.203:5060 —>
SIP/2.0 200 OK
To: sip:[email protected]:5060;tag=b2910c0842a4665bi0
From: “Unknown” sip:[email protected];tag=as280787ef
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK1086245b
Server: Cisco/SPA112-1.3.5(004p)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces

<------------->
[2018-03-08 22:04:39] VERBOSE[1926] chan_sip.c: — (10 headers 0 lines) —
[2018-03-08 22:04:39] VERBOSE[1926] chan_sip.c: Really destroying SIP dialog ‘[email protected]:5060’ Method: OPTIONS
[2018-03-08 22:04:39] VERBOSE[1926] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.1.203:5061:
OPTIONS sip:[email protected]:5061 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK2464bfcc
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as3861e6a1
To: sip:[email protected]:5061
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-13.0.194.2(13.13.1)
Date: Thu, 08 Mar 2018 21:04:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

[2018-03-08 22:04:40] VERBOSE[1926] chan_sip.c:
<— SIP read from UDP:192.168.1.203:5061 —>
SIP/2.0 200 OK
To: sip:[email protected]:5061;tag=ca44d6a0ea555ca3i1
From: “Unknown” sip:[email protected];tag=as3861e6a1
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.1.201:5060;branch=z9hG4bK2464bfcc
Server: Cisco/SPA112-1.3.5(004p)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces

<------------->
[2018-03-08 22:04:40] VERBOSE[1926] chan_sip.c: — (10 headers 0 lines) —
[2018-03-08 22:04:40] VERBOSE[1926] chan_sip.c: Really destroying SIP dialog ‘[email protected]:5060’ Method: OPTIONS
[2018-03-08 22:04:42] VERBOSE[1926] chan_sip.c:
<— SIP read from UDP:192.168.1.202:5060 —>
INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.202:5060;branch=z9hG4bK3f9eff0e938ce148c4c9820813bcdc08;rport
From: “Gigaset” sip:[email protected];tag=1393343259
To: sip:[email protected];user=phone
Call-ID: 3899627947@192_168_1_202
CSeq: 2 INVITE
Contact: sip:[email protected]:5060
Max-Forwards: 70
User-Agent: E630A GO/42.245.00.000.000
Supported: replaces
Allow-Events: message-summary, refer, ua-profile, talk, check-sync
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY, REFER, UPDATE
Content-Type: application/sdp
Content-Length: 381

v=0
o=302 5006 9 IN IP4 192.168.1.202
s=Mapping
c=IN IP4 192.168.1.202
t=0 0
m=audio 5006 RTP/AVP 9 8 0 96 97 2 18 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:96 G726-32/8000
a=rtpmap:97 AAL2-G726-32/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
<------------->
[2018-03-08 22:04:42] VERBOSE[1926] chan_sip.c: — (14 headers 17 lines) —
[2018-03-08 22:04:42] VERBOSE[1926] chan_sip.c: Sending to 192.168.1.202:5060 (NAT)
[2018-03-08 22:04:42] VERBOSE[1926][C-00000030] chan_sip.c: Sending to 192.168.1.202:5060 (NAT)
[2018-03-08 22:04:42] VERBOSE[1926][C-00000030] chan_sip.c: Using INVITE request as basis request - 3899627947@192_168_1_202
[2018-03-08 22:04:42] VERBOSE[1926][C-00000030] chan_sip.c: Found peer ‘302’ for ‘302’ from 192.168.1.202:5060
[2018-03-08 22:04:42] VERBOSE[1926][C-00000030] chan_sip.c:
<— Reliably Transmitting (no NAT) to 192.168.1.202:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.202:5060;branch=z9hG4bK3f9eff0e938ce148c4c9820813bcdc08;received=192.168.1.202;rport=5060
From: “Gigaset” sip:[email protected];tag=1393343259
To: sip:[email protected];user=phone;tag=as3b896d59
Call-ID: 3899627947@192_168_1_202
CSeq: 2 INVITE
Server: FPBX-13.0.194.2(13.13.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="7c3689a6"
Content-Length: 0

<------------>
[2018-03-08 22:04:42] VERBOSE[1926][C-00000030] chan_sip.c: Scheduling destruction of SIP dialog ‘3899627947@192_168_1_202’ in 6400 ms (Method: INVITE)
[2018-03-08 22:04:42] VERBOSE[1926] chan_sip.c:
<— SIP read from UDP:192.168.1.202:5060 —>
ACK sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.202:5060;branch=z9hG4bK3f9eff0e938ce148c4c9820813bcdc08;rport
From: “Gigaset” sip:[email protected];tag=1393343259
To: sip:[email protected];user=phone;tag=as3b896d59
Call-ID: 3899627947@192_168_1_202
CSeq: 2 ACK
Contact: sip:[email protected]:5060
Max-Forwards: 70
User-Agent: E630A GO/42.245.00.000.000
Content-Length: 0

dicko · March 9, 2018, 9:20pm

You will have to check with Swisscom, they replied with a 401 (Unauthorized) in response to your invite to
sip:003934XXXXXXXX

Stewart1 · March 10, 2018, 7:06am

The log shows that at 22:04:15, the router sends a 180 Ringing with audio. Possibly the phone was really ringing, or there may have been an out of range announcement or similar. The audio appears to have been passed correctly to the Gigaset. What (if anything) was heard?

However, after that, the router fails to respond to REGISTER requests and when the Gigaset user hangs up at 22:04:31, Asterisks starts sending CANCEL requests to the router, which are also ignored.

At face value, this is a problem with the router, but it may have been triggered by something that is avoidable.

Can you configure the Gigaset to register with the router directly? If so, does it have the same problem on calls to IT mobile?

Also, are you required to use the router as a proxy (if you can register directly to the Swisscom server, it may avoid this issue)?

Finally, I believe that at 22:04:42, the call was manually retried at the Gigaset. Is that correct? If so, then I disagree with dicko’s diagnosis: The 401 Unauthorized was a normal challenge from Asterisk to the Gigaset for the second call (which the Gigaset presumably replied to but it was after the end of the log section that you posted).

Paolodic · March 11, 2018, 11:22am

Hello Freinds,
I have some news:

I can confirm that the called number rings just once, no possibility to answer.
The call was retried shortlz after, maybe I have cutted the log at the wrong place…
I also configured the gigaset directly to the router, bypassing freepbx. The result is exactlz the same* --> So it is NOT a problem on the pbx, but it is on the provider.
I have contacted them and I espect something next week.
I will also get the credentials for accessing the SIP account directly and try to reconfigure the trunk with these.

I will keep zou informed.

Thanks
Paolo

Paolodic · March 18, 2018, 4:54pm

Hello Friends,
here some updates.
They told me that there were too* many trials in setting up the calls (too many codecs?) so I did also test with only g711 alaw with SIP phone point directly to the router, but the result was the same: fail.

After getting the SIP Credentials from the provider (not the ones generated on the router), I have configured a SIP phone to point directly to the provider and no more to my router. The result is that the call is successful.
I wanted then to configure the Trunk of Raspbx to point directly to the provider with the SIP credentials, but then I realized that they have upgraded the firmware of my router to a new version (they told me that there was a beta for testing, but they decided to install it…). So before configure the trunk I tried another call and it was also successful!
At this point, I configured the trunk, just to see if it was registering. It was OK. Then I give up (no more time) and didn’t configure inbound and outbound routes.

At present everything works ok.

Thank you very much for the suggestions and support.
Paolo

system · March 18, 2019, 4:54pm

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