Hi Guys
I need some help with making an outgoing call:
FreePbx 11
Centos 6.3
I have installed an configured a SIP Trunk with my ISP and its registered.
I have created an extension and that’s registered.
I need to know what i need to do get a call out this Trunk.
In SA we have 10 digit numbers and all start with “0” so i need to know how to configure the dial plan or is there a wild card i can use.
Also were do i configure this is the GUI.
Regards
Denzel
You need to set up an Outbound Route from the Freepbx Outbound Routes menu. You can use a dial pattern of 0. to match any number beginning with 0. If you want to limit to only 10 digit non international calls, you could use a dial pattern of 0ZXXXXXXXX. There are help popups on the freepbx Outbound Routes menu which give useful information about writing dial patterns.
Hi John
Thanks for your reply.
I see that trunk had some issues and now the sip show peers shows registered
Host dnsmgr Username Refresh State Reg.Time
xx.xxx.x.xx:5060 N username 45 Registered Sat, 11 May 2013 18:43:16
1 SIP registrations.
I add this to this 0ZXXXXXXXX to the new outbound route
0 = prepend+prefix =Z|match pattern is XXXXXXXX
this is how the GUI has it.
I still cant get a call out. here is sip trace: (i just noticed that there is not trace of this sip information going to the ISP my is an ip starting with 41.xx.xx.xx)
Connected to Asterisk 11.3.0 currently running on localhost (pid = 1614)
<— SIP read from UDP:192.168.10.202:6654 —>
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.10.202:6654;branch=z9hG4bK-d8754z-99b4c65d9959270d-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:[email protected]:6654
To: sip:[email protected]
From: "Denzel Marimuthu"sip:[email protected];tag=10ddd132
Call-ID: ZTEzYWVjNTY2YmQzNGYwMGU3NzFlMjRiMmIwNzgyYTU
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite release 4.5.2 stamp 70142
Content-Length: 307
v=0
o=- 13012764492936112 1 IN IP4 192.168.10.202
s=X-Lite 4 release 4.5.2 stamp 70142
c=IN IP4 192.168.10.202
t=0 0
m=audio 55800 RTP/AVP 125 9 8 0 100 101
a=rtpmap:125 opus/48000/2
a=fmtp:125 useinbandfec=1
a=rtpmap:100 speex/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
— (13 headers 12 lines) —
Sending to 192.168.10.202:6654 (no NAT)
Using INVITE request as basis request - ZTEzYWVjNTY2YmQzNGYwMGU3NzFlMjRiMmIwNzgyYTU
Found peer ‘1000’ for ‘1000’ from 192.168.10.202:6654
<— Reliably Transmitting (no NAT) to 192.168.10.202:6654 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.10.202:6654;branch=z9hG4bK-d8754z-99b4c65d9959270d-1—d8754z-;received=192.168.10.202;rport=6654
From: "Denzel Marimuthu"sip:[email protected];tag=10ddd132
To: sip:[email protected];tag=as7e28b2e8
Call-ID: ZTEzYWVjNTY2YmQzNGYwMGU3NzFlMjRiMmIwNzgyYTU
CSeq: 1 INVITE
Server: FPBX2.11.0rc1(11.3.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="278cced4"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘ZTEzYWVjNTY2YmQzNGYwMGU3NzFlMjRiMmIwNzgyYTU’ in 6400 ms (Method: INVITE)
<— SIP read from UDP:192.168.10.202:6654 —>
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.10.202:6654;branch=z9hG4bK-d8754z-99b4c65d9959270d-1—d8754z-;rport
Max-Forwards: 70
To: sip:[email protected];tag=as7e28b2e8
From: "Denzel Marimuthu"sip:[email protected];tag=10ddd132
Call-ID: ZTEzYWVjNTY2YmQzNGYwMGU3NzFlMjRiMmIwNzgyYTU
CSeq: 1 ACK
Content-Length: 0
<------------->
— (8 headers 0 lines) —
<— SIP read from UDP:192.168.10.202:6654 —>
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.10.202:6654;branch=z9hG4bK-d8754z-83cc535f0074c071-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:[email protected]:6654
To: sip:[email protected]
From: “Denzel Marimuthu"sip:[email protected];tag=10ddd132
Call-ID: ZTEzYWVjNTY2YmQzNGYwMGU3NzFlMjRiMmIwNzgyYTU
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite release 4.5.2 stamp 70142
Authorization: Digest username=“1000”,realm=“asterisk”,nonce=“278cced4”,uri="sip:[email protected]”,response=“a881ff03b1ed49d095e3792149b11a5c”,algorithm=MD5
Content-Length: 307
v=0
o=- 13012764492936112 1 IN IP4 192.168.10.202
s=X-Lite 4 release 4.5.2 stamp 70142
c=IN IP4 192.168.10.202
t=0 0
m=audio 55800 RTP/AVP 125 9 8 0 100 101
a=rtpmap:125 opus/48000/2
a=fmtp:125 useinbandfec=1
a=rtpmap:100 speex/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
— (14 headers 12 lines) —
Sending to 192.168.10.202:6654 (no NAT)
Using INVITE request as basis request - ZTEzYWVjNTY2YmQzNGYwMGU3NzFlMjRiMmIwNzgyYTU
Found peer ‘1000’ for ‘1000’ from 192.168.10.202:6654
Found RTP audio format 125
Found RTP audio format 9
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 100
Found RTP audio format 101
Found unknown media description format opus for ID 125
Found audio description format speex for ID 100
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw), peer - audio=(ulaw|alaw|speex16|g722)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.10.202:55800
Looking for 00832898786 in from-internal (domain 192.168.10.16)
list_route: hop: sip:[email protected]:6654
<— Transmitting (no NAT) to 192.168.10.202:6654 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.10.202:6654;branch=z9hG4bK-d8754z-83cc535f0074c071-1—d8754z-;received=192.168.10.202;rport=6654
From: "Denzel Marimuthu"sip:[email protected];tag=10ddd132
To: sip:[email protected]
Call-ID: ZTEzYWVjNTY2YmQzNGYwMGU3NzFlMjRiMmIwNzgyYTU
CSeq: 2 INVITE
Server: FPBX2.11.0rc1(11.3.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Length: 0
<------------>
Audio is at 11334
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<— Transmitting (no NAT) to 192.168.10.202:6654 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.10.202:6654;branch=z9hG4bK-d8754z-83cc535f0074c071-1—d8754z-;received=192.168.10.202;rport=6654
From: "Denzel Marimuthu"sip:[email protected];tag=10ddd132
To: sip:[email protected];tag=as45c2855e
Call-ID: ZTEzYWVjNTY2YmQzNGYwMGU3NzFlMjRiMmIwNzgyYTU
CSeq: 2 INVITE
Server: FPBX2.11.0rc1(11.3.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 259
v=0
o=root 985877677 985877677 IN IP4 192.168.10.16
s=Asterisk PBX 11.3.0
c=IN IP4 192.168.10.16
t=0 0
m=audio 11334 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------>
Really destroying SIP dialog ‘0d97971c27ed53e9644cc15449436685@[::1]’ Method: REGISTER
<— SIP read from UDP:192.168.10.202:6654 —>
<------------->
<— Reliably Transmitting (no NAT) to 192.168.10.202:6654 —>
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.10.202:6654;branch=z9hG4bK-d8754z-83cc535f0074c071-1—d8754z-;received=192.168.10.202;rport=6654
From: "Denzel Marimuthu"sip:[email protected];tag=10ddd132
To: sip:[email protected];tag=as45c2855e
Call-ID: ZTEzYWVjNTY2YmQzNGYwMGU3NzFlMjRiMmIwNzgyYTU
CSeq: 2 INVITE
Server: FPBX2.11.0rc1(11.3.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------>
[2013-05-11 18:48:23] WARNING[5914][C-00000034]: channel.c:4816 ast_prod: Prodding channel ‘SIP/1000-00000042’ failed
<— SIP read from UDP:192.168.10.202:6654 —>
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.10.202:6654;branch=z9hG4bK-d8754z-83cc535f0074c071-1—d8754z-;rport
Max-Forwards: 70
To: sip:[email protected];tag=as45c2855e
From: "Denzel Marimuthu"sip:[email protected];tag=10ddd132
Call-ID: ZTEzYWVjNTY2YmQzNGYwMGU3NzFlMjRiMmIwNzgyYTU
CSeq: 2 ACK
Content-Length: 0
<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘ZTEzYWVjNTY2YmQzNGYwMGU3NzFlMjRiMmIwNzgyYTU’ Method: ACK
Regards
I don’t believe you need a prepend or prefix, just match pattern 0ZXXXXXXXX
That’s true so long as your provider requires to receive the number as dialed without any specific prefix (like + 00 and country code).
HI John
Thanks for that, so i decided i will stop being lazy and read the popup and i understand the “X” is used for digits 0 to 9 which is fine for me as my numbers are 10 digits long and start with 0. Now i get a No Nat issue! my SP again says the trunk is registered but they can see a call coming sip trace shows in passing the correct number but not leaving the asterisk box.
Regards
Can you show the verbose log from asterisk of what is happening. Can you capture a SIP trace (like you did before)?
Hi John
Your assistance is leading me to positiveness 
So im getting an outbound call!! YAY!!! I found issues with the Outbound Trunk Settings : this is my trunk settings now :
host=xx.xx.xx.xx
username=######
type=peer
secret=********
qualify=yes
insecure=invite&port
disallow=all
allow=gsm&ulaw&alaw&g279&line
So now the call connects but as soon as i speak the call drops.
Some other good news : i have remote stations logging into the Asterisk server but again audio is the issue.
“SIP TRACE OF CALL GETTING DROPPED AFTER ±30 SECONDS”
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Executing [0715864789@from-internal:1] Macro(“SIP/1000-0000000c”, “user-callerid,LIMIT,”) in new stack
– Executing [s@macro-user-callerid:1] Set(“SIP/1000-0000000c”, “TOUCH_MONITOR=1368508967.12”) in new stack
– Executing [s@macro-user-callerid:2] Set(“SIP/1000-0000000c”, “AMPUSER=1000”) in new stack
– Executing [s@macro-user-callerid:3] GotoIf(“SIP/1000-0000000c”, “0?report”) in new stack
– Executing [s@macro-user-callerid:4] ExecIf(“SIP/1000-0000000c”, “1?Set(REALCALLERIDNUM=1000)”) in new stack
– Executing [s@macro-user-callerid:5] Set(“SIP/1000-0000000c”, “AMPUSER=1000”) in new stack
– Executing [s@macro-user-callerid:6] Set(“SIP/1000-0000000c”, “AMPUSERCIDNAME=Denzel”) in new stack
– Executing [s@macro-user-callerid:7] GotoIf(“SIP/1000-0000000c”, “0?report”) in new stack
– Executing [s@macro-user-callerid:8] Set(“SIP/1000-0000000c”, “AMPUSERCID=1000”) in new stack
– Executing [s@macro-user-callerid:9] Set(“SIP/1000-0000000c”, “__DIAL_OPTIONS=Ttr”) in new stack
– Executing [s@macro-user-callerid:10] Set(“SIP/1000-0000000c”, “CALLERID(all)=“Denzel” <1000>”) in new stack
– Executing [s@macro-user-callerid:11] GotoIf(“SIP/1000-0000000c”, “0?limit”) in new stack
– Executing [s@macro-user-callerid:12] ExecIf(“SIP/1000-0000000c”, “1?Set(GROUP(concurrency_limit)=1000)”) in new stack
– Executing [s@macro-user-callerid:13] GosubIf(“SIP/1000-0000000c”, “7?sub-ccss,s,1(from-internal,0715864789)”) in new stack
– Executing [s@sub-ccss:1] ExecIf(“SIP/1000-0000000c”, “0?Return()”) in new stack
– Executing [s@sub-ccss:2] Set(“SIP/1000-0000000c”, “CCSS_SETUP=TRUE”) in new stack
– Executing [s@sub-ccss:3] GosubIf(“SIP/1000-0000000c”, “0?monitor_config,1(from-internal,0715864789):monitor_default,1(from-internal,0715864789)”) in new stack
– Executing [monitor_default@sub-ccss:1] GotoIf(“SIP/1000-0000000c”, “0?is_exten”) in new stack
– Executing [monitor_default@sub-ccss:2] StackPop(“SIP/1000-0000000c”, “”) in new stack
– Executing [monitor_default@sub-ccss:3] Return(“SIP/1000-0000000c”, “FALSE”) in new stack
– Executing [s@macro-user-callerid:14] ExecIf(“SIP/1000-0000000c”, “0?Set(CHANNEL(language)=)”) in new stack
– Executing [s@macro-user-callerid:15] GotoIf(“SIP/1000-0000000c”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,28)
– Executing [s@macro-user-callerid:28] Set(“SIP/1000-0000000c”, “CALLERID(number)=1000”) in new stack
– Executing [s@macro-user-callerid:29] Set(“SIP/1000-0000000c”, “CALLERID(name)=Denzel”) in new stack
– Executing [s@macro-user-callerid:30] Set(“SIP/1000-0000000c”, “CDR(cnum)=1000”) in new stack
– Executing [s@macro-user-callerid:31] Set(“SIP/1000-0000000c”, “CDR(cnam)=Denzel”) in new stack
– Executing [s@macro-user-callerid:32] Set(“SIP/1000-0000000c”, “CHANNEL(language)=en”) in new stack
– Executing [0715864789@from-internal:2] Set(“SIP/1000-0000000c”, “MOHCLASS=default”) in new stack
– Executing [0715864789@from-internal:3] Set(“SIP/1000-0000000c”, “_NODEST=”) in new stack
– Executing [0715864789@from-internal:4] Gosub(“SIP/1000-0000000c”, “sub-record-check,s,1(out,0715864789,)”) in new stack
– Executing [s@sub-record-check:1] Set(“SIP/1000-0000000c”, “REC_POLICY_MODE_SAVE=”) in new stack
– Executing [s@sub-record-check:2] GotoIf(“SIP/1000-0000000c”, “1?check”) in new stack
– Goto (sub-record-check,s,7)
– Executing [s@sub-record-check:7] Set(“SIP/1000-0000000c”, “__MON_FMT=wav”) in new stack
– Executing [s@sub-record-check:8] GotoIf(“SIP/1000-0000000c”, “1?next”) in new stack
– Goto (sub-record-check,s,11)
– Executing [s@sub-record-check:11] ExecIf(“SIP/1000-0000000c”, “0?Return()”) in new stack
– Executing [s@sub-record-check:12] ExecIf(“SIP/1000-0000000c”, “0?Set(__REC_POLICY_MODE=)”) in new stack
– Executing [s@sub-record-check:13] GotoIf(“SIP/1000-0000000c”, “0?out,1”) in new stack
– Executing [s@sub-record-check:14] Set(“SIP/1000-0000000c”, “__REC_STATUS=INITIALIZED”) in new stack
– Executing [s@sub-record-check:15] Set(“SIP/1000-0000000c”, “NOW=1368508967”) in new stack
– Executing [s@sub-record-check:16] Set(“SIP/1000-0000000c”, “__DAY=14”) in new stack
– Executing [s@sub-record-check:17] Set(“SIP/1000-0000000c”, “__MONTH=05”) in new stack
– Executing [s@sub-record-check:18] Set(“SIP/1000-0000000c”, “__YEAR=2013”) in new stack
– Executing [s@sub-record-check:19] Set(“SIP/1000-0000000c”, “__TIMESTR=20130514-072247”) in new stack
– Executing [s@sub-record-check:20] Set(“SIP/1000-0000000c”, “__FROMEXTEN=1000”) in new stack
– Executing [s@sub-record-check:21] Set(“SIP/1000-0000000c”, “__CALLFILENAME=out-0715864789-1000-20130514-072247-1368508967.12”) in new stack
– Executing [s@sub-record-check:22] Goto(“SIP/1000-0000000c”, “out,1”) in new stack
– Goto (sub-record-check,out,1)
– Executing [out@sub-record-check:1] ExecIf(“SIP/1000-0000000c”, “1?Set(__REC_POLICY_MODE=dontcare)”) in new stack
– Executing [out@sub-record-check:2] GosubIf(“SIP/1000-0000000c”, “0?record,1(exten,0715864789,1000)”) in new stack
– Executing [out@sub-record-check:3] Return(“SIP/1000-0000000c”, “”) in new stack
– Executing [0715864789@from-internal:5] Macro(“SIP/1000-0000000c”, “dialout-trunk,1,0715864789,off”) in new stack
– Executing [s@macro-dialout-trunk:1] Set(“SIP/1000-0000000c”, “DIAL_TRUNK=1”) in new stack
– Executing [s@macro-dialout-trunk:2] GosubIf(“SIP/1000-0000000c”, “0?sub-pincheck,s,1()”) in new stack
– Executing [s@macro-dialout-trunk:3] GotoIf(“SIP/1000-0000000c”, “0?disabletrunk,1”) in new stack
– Executing [s@macro-dialout-trunk:4] Set(“SIP/1000-0000000c”, “DIAL_NUMBER=0715864789”) in new stack
– Executing [s@macro-dialout-trunk:5] Set(“SIP/1000-0000000c”, “DIAL_TRUNK_OPTIONS=Ttr”) in new stack
– Executing [s@macro-dialout-trunk:6] Set(“SIP/1000-0000000c”, “OUTBOUND_GROUP=OUT_1”) in new stack
– Executing [s@macro-dialout-trunk:7] GotoIf(“SIP/1000-0000000c”, “1?nomax”) in new stack
– Goto (macro-dialout-trunk,s,9)
– Executing [s@macro-dialout-trunk:9] GotoIf(“SIP/1000-0000000c”, “0?skipoutcid”) in new stack
– Executing [s@macro-dialout-trunk:10] Set(“SIP/1000-0000000c”, “DIAL_TRUNK_OPTIONS=Tt”) in new stack
– Executing [s@macro-dialout-trunk:11] Macro(“SIP/1000-0000000c”, “outbound-callerid,1”) in new stack
– Executing [s@macro-outbound-callerid:1] ExecIf(“SIP/1000-0000000c”, “0?Set(CALLERPRES()=)”) in new stack
– Executing [s@macro-outbound-callerid:2] ExecIf(“SIP/1000-0000000c”, “0?Set(REALCALLERIDNUM=1000)”) in new stack
– Executing [s@macro-outbound-callerid:3] GotoIf(“SIP/1000-0000000c”, “1?normcid”) in new stack
– Goto (macro-outbound-callerid,s,6)
– Executing [s@macro-outbound-callerid:6] Set(“SIP/1000-0000000c”, “USEROUTCID=”) in new stack
– Executing [s@macro-outbound-callerid:7] Set(“SIP/1000-0000000c”, “EMERGENCYCID=”) in new stack
– Executing [s@macro-outbound-callerid:8] Set(“SIP/1000-0000000c”, “TRUNKOUTCID=”) in new stack
– Executing [s@macro-outbound-callerid:9] GotoIf(“SIP/1000-0000000c”, “1?trunkcid”) in new stack
– Goto (macro-outbound-callerid,s,14)
– Executing [s@macro-outbound-callerid:14] ExecIf(“SIP/1000-0000000c”, “0?Set(CALLERID(all)=)”) in new stack
– Executing [s@macro-outbound-callerid:15] ExecIf(“SIP/1000-0000000c”, “0?Set(CALLERID(all)=)”) in new stack
– Executing [s@macro-outbound-callerid:16] ExecIf(“SIP/1000-0000000c”, “0?Set(CALLERID(all)=)”) in new stack
– Executing [s@macro-outbound-callerid:17] ExecIf(“SIP/1000-0000000c”, “0?Set(CALLERPRES()=prohib_passed_screen)”) in new stack
– Executing [s@macro-outbound-callerid:18] Set(“SIP/1000-0000000c”, “CDR(outbound_cnum)=1000”) in new stack
– Executing [s@macro-outbound-callerid:19] Set(“SIP/1000-0000000c”, “CDR(outbound_cnam)=Denzel”) in new stack
– Executing [s@macro-dialout-trunk:12] GosubIf(“SIP/1000-0000000c”, “0?sub-flp-1,s,1()”) in new stack
– Executing [s@macro-dialout-trunk:13] Set(“SIP/1000-0000000c”, “OUTNUM=0715864789”) in new stack
– Executing [s@macro-dialout-trunk:14] Set(“SIP/1000-0000000c”, “custom=SIP/Out_Bella”) in new stack
– Executing [s@macro-dialout-trunk:15] ExecIf(“SIP/1000-0000000c”, “0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)Tt)”) in new stack
– Executing [s@macro-dialout-trunk:16] ExecIf(“SIP/1000-0000000c”, “0?Set(DIAL_TRUNK_OPTIONS=TtM(confirm))”) in new stack
– Executing [s@macro-dialout-trunk:17] Macro(“SIP/1000-0000000c”, “dialout-trunk-predial-hook,”) in new stack
– Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit(“SIP/1000-0000000c”, “”) in new stack
– Executing [s@macro-dialout-trunk:18] GotoIf(“SIP/1000-0000000c”, “0?bypass,1”) in new stack
– Executing [s@macro-dialout-trunk:19] ExecIf(“SIP/1000-0000000c”, “1?Set(CONNECTEDLINE(num,i)=0715864789)”) in new stack
– Executing [s@macro-dialout-trunk:20] ExecIf(“SIP/1000-0000000c”, “1?Set(CONNECTEDLINE(name,i)=CID:1000)”) in new stack
– Executing [s@macro-dialout-trunk:21] GotoIf(“SIP/1000-0000000c”, “0?customtrunk”) in new stack
– Executing [s@macro-dialout-trunk:22] Dial(“SIP/1000-0000000c”, “SIP/Out_Bella/0715864789,300,Tt”) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Called SIP/Out_Bella/0715864789
– SIP/Out_Bella-0000000d is making progress passing it to SIP/1000-0000000c
– SIP/Out_Bella-0000000d is ringing
– SIP/Out_Bella-0000000d is making progress passing it to SIP/1000-0000000c
– SIP/Out_Bella-0000000d answered SIP/1000-0000000c
[2013-05-14 07:23:12] WARNING[1744]: CHAN_SIP.C:4169 RETRANS_PKT: RETRANSMISSION TIMEOUT REACHED ON TRANSMISSION YJHIODNHNWU3NDY0YZAZZDQYMGIXYWQYOTIXNDLJZDK FOR SEQNO 2 (CRITICAL RESPONSE) – SEE HTTPS://WIKI.ASTERISK.ORG/WIKI/DISPLAY/AST/SIP+RETRANSMISSIONS
PACKET TIMED OUT AFTER 16640MS WITH NO RESPONSE
[2013-05-14 07:23:12] WARNING[1744]: CHAN_SIP.C:4198 RETRANS_PKT: HANGING UP CALL YJHIODNHNWU3NDY0YZAZZDQYMGIXYWQYOTIXNDLJZDK - NO REPLY TO OUR CRITICAL PACKET (SEE HTTPS://WIKI.ASTERISK.ORG/WIKI/DISPLAY/AST/SIP+RETRANSMISSIONS).
– EXECUTING [H@MACRO-DIALOUT-TRUNK:1] MACRO(“SIP/1000-0000000C”, “HANGUPCALL,”) IN NEW STACK
– EXECUTING [S@MACRO-HANGUPCALL:1] GOTOIF(“SIP/1000-0000000C”, “1?THEEND”) IN NEW STACK
– GOTO (MACRO-HANGUPCALL,S,3)
– EXECUTING [S@MACRO-HANGUPCALL:3] EXECIF(“SIP/1000-0000000C”, “0?SET(CDR(RECORDINGFILE)=)”) IN NEW STACK
– EXECUTING [S@MACRO-HANGUPCALL:4] HANGUP(“SIP/1000-0000000C”, “”) IN NEW STACK
== SPAWN EXTENSION (MACRO-HANGUPCALL, S, 4) EXITED NON-ZERO ON ‘SIP/1000-0000000C’ IN MACRO ‘HANGUPCALL’
== SPAWN EXTENSION (MACRO-DIALOUT-TRUNK, H, 1) EXITED NON-ZERO ON ‘SIP/1000-0000000C’
== SPAWN EXTENSION (MACRO-DIALOUT-TRUNK, S, 22) EXITED NON-ZERO ON ‘SIP/1000-0000000C’ IN MACRO ‘DIALOUT-TRUNK’
== SPAWN EXTENSION (FROM-INTERNAL, 0715864789, 5) EXITED NON-ZERO ON 'SIP/1000-0000000C’
LOCALHOST*CLI>
Regards
Denz
Hi John
I have fixed the issue of the call dropping after 30 seconds, i went into the extension config and made nat=yes. So i have audio over all calls and extension logging remotely can make calls out the pstn(SIP trunk).
I have managed to get an inbound call hitting the pbx but i get a code translation error, unable to translate (g729) to (ulaw) ? the does ring on the far end but no audio…
Regards
If you have run out of G729 licenses that might be your problem.
Hi John
So i bought a G729 Licence from Digum and installed as per there setup, call routes from external to inside eg: extension 1000 but no audio and for some reason two calls come to the extension at once although im only dialing that external number once. Below is the "show codecs:
100001 audio g723 (G.723.1)
100002 audio gsm (GSM)
100003 audio ulaw (G.711 u-law)
100004 audio alaw (G.711 A-law)
100011 audio g726 (G.726 RFC3551)
100006 audio adpcm (ADPCM)
100019 audio slin (16 bit Signed Linear PCM)
100007 audio lpc10 (LPC10)
100008 audio g729 (G.729A)
100009 audio speex (SpeeX)
100016 audio speex16 (SpeeX 16khz)
100010 audio ilbc (iLBC)
100005 audio g726aal2 (G.726 AAL2)
100012 audio g722 (G722)
100021 audio slin16 (16 bit Signed Linear PCM (16kHz))
300001 image jpeg (JPEG image)
300002 image png (PNG image)
200001 video h261 (H.261 Video)
200002 video h263 (H.263 Video)
200003 video h263p (H.263+ Video)
200004 video h264 (H.264 Video)
200005 video mpeg4 (MPEG4 Video)
400001 text red (T.140 Realtime Text with redundancy)
400002 text t140 (Passthrough T.140 Realtime Text)
100013 audio siren7 (ITU G.722.1 (Siren7, licensed from Polycom))
100014 audio siren14 (ITU G.722.1 Annex C, (Siren14, licensed from Polycom))
100017 audio testlaw (G.711 test-law)
100015 audio g719 (ITU G.719)
100028 audio speex32 (SpeeX 32khz)
100020 audio slin12 (16 bit Signed Linear PCM (12kHz))
100022 audio slin24 (16 bit Signed Linear PCM (24kHz))
100023 audio slin32 (16 bit Signed Linear PCM (32kHz))
100024 audio slin44 (16 bit Signed Linear PCM (44kHz))
100025 audio slin48 (16 bit Signed Linear PCM (48kHz))
100026 audio slin96 (16 bit Signed Linear PCM (96kHz))
100027 audio slin192 (16 bit Signed Linear PCM (192kHz))
and
core show translation
?