Hi all!
This is my tricky problem.
I have a Trixbox PBX (celeron 3ghz with 1.5gb ram) that is working flawless with two VoipPhones Thomson ST2030 and 2 voip accounts (VoipStunt and VoipBuster).
Phones speak each other and voip channels are working fine.
Now I must conect Trixbox PBX (ip=77.0.0.166) with our internal Samsung PBX (ip=77.0.0.134).
There are the settings (in peer details) :
type=friend
host=77.0.0.134
qualify=yes
canreinvite=yes
context=from-internal
disallow=all
allow=alaw
nat=yes
srvlookup=yes
And this is the log (when from my voip extension 2001 i call my connected extension 2245 on Samsung PBX)
The 2245 standard phone ring, i answer, but there is no voice (i click keys on extension 2001 and DTMF are sended) (the viceversa wont work)
trixbox1*CLI>
– Executing [2245@from-internal:1] Macro(“SIP/2001-098943c0”, “user-callerid|SKIPTTL|”) in new stack
– Executing [s@macro-user-callerid:1] NoOp(“SIP/2001-098943c0”, “user-callerid: device 2001”) in new stack
– Executing [s@macro-user-callerid:2] Set(“SIP/2001-098943c0”, “AMPUSER=2001”) in new stack
– Executing [s@macro-user-callerid:3] GotoIf(“SIP/2001-098943c0”, “0?report”) in new stack
– Executing [s@macro-user-callerid:4] ExecIf(“SIP/2001-098943c0”, “1|Set|REALCALLERIDNUM=2001”) in new stack
– Executing [s@macro-user-callerid:5] NoOp(“SIP/2001-098943c0”, “REALCALLERIDNUM is 2001”) in new stack
– Executing [s@macro-user-callerid:6] Set(“SIP/2001-098943c0”, “AMPUSER=2001”) in new stack
– Executing [s@macro-user-callerid:7] Set(“SIP/2001-098943c0”, “AMPUSERCIDNAME=interno1”) in new stack
– Executing [s@macro-user-callerid:8] GotoIf(“SIP/2001-098943c0”, “0?report”) in new stack
– Executing [s@macro-user-callerid:9] Set(“SIP/2001-098943c0”, “AMPUSERCID=2001”) in new stack
– Executing [s@macro-user-callerid:10] Set(“SIP/2001-098943c0”, “CALLERID(all)=“interno1” <2001>”) in new stack
– Executing [s@macro-user-callerid:11] Set(“SIP/2001-098943c0”, “REALCALLERIDNUM=2001”) in new stack
– Executing [s@macro-user-callerid:12] ExecIf(“SIP/2001-098943c0”, “1|Set|CHANNEL(language)=it”) in new stack
– Executing [s@macro-user-callerid:13] NoOp(“SIP/2001-098943c0”, “TTL: ARG1: SKIPTTL”) in new stack
– Executing [s@macro-user-callerid:14] GotoIf(“SIP/2001-098943c0”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,23)
– Executing [s@macro-user-callerid:23] NoOp(“SIP/2001-098943c0”, “Using CallerID “interno1” <2001>”) in new stack
– Executing [2245@from-internal:2] Set(“SIP/2001-098943c0”, “_NODEST=”) in new stack
– Executing [2245@from-internal:3] Macro(“SIP/2001-098943c0”, “record-enable|2001|OUT|”) in new stack
– Executing [s@macro-record-enable:1] GotoIf(“SIP/2001-098943c0”, “0?2:4”) in new stack
– Goto (macro-record-enable,s,4)
– Executing [s@macro-record-enable:4] AGI(“SIP/2001-098943c0”, “recordingcheck|20081110-172714|1226334434.0”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20081110-172714|1226334434.0: Outbound recording not enabled
– AGI Script recordingcheck completed, returning 0
– Executing [s@macro-record-enable:5] NoOp(“SIP/2001-098943c0”, “No recording needed”) in new stack
– Executing [2245@from-internal:4] Macro(“SIP/2001-098943c0”, “dialout-trunk|1|2245||”) in new stack
– Executing [s@macro-dialout-trunk:1] Set(“SIP/2001-098943c0”, “DIAL_TRUNK=1”) in new stack
– Executing [s@macro-dialout-trunk:2] ExecIf(“SIP/2001-098943c0”, “0|Authenticate|”) in new stack
– Executing [s@macro-dialout-trunk:3] GotoIf(“SIP/2001-098943c0”, “0?disabletrunk|1”) in new stack
– Executing [s@macro-dialout-trunk:4] Set(“SIP/2001-098943c0”, “DIAL_NUMBER=2245”) in new stack
– Executing [s@macro-dialout-trunk:5] Set(“SIP/2001-098943c0”, “DIAL_TRUNK_OPTIONS=tTr”) in new stack
– Executing [s@macro-dialout-trunk:6] Set(“SIP/2001-098943c0”, “GROUP()=OUT_1”) in new stack
– Executing [s@macro-dialout-trunk:7] GotoIf(“SIP/2001-098943c0”, “0?nomax”) in new stack
– Executing [s@macro-dialout-trunk:8] GotoIf(“SIP/2001-098943c0”, “0?chanfull”) in new stack
– Executing [s@macro-dialout-trunk:9] GotoIf(“SIP/2001-098943c0”, “0?skipoutcid”) in new stack
– Executing [s@macro-dialout-trunk:10] Set(“SIP/2001-098943c0”, “DIAL_TRUNK_OPTIONS=”) in new stack
– Executing [s@macro-dialout-trunk:11] Macro(“SIP/2001-098943c0”, “outbound-callerid|1”) in new stack
– Executing [s@macro-outbound-callerid:1] ExecIf(“SIP/2001-098943c0”, “0|SetCallerPres|”) in new stack
– Executing [s@macro-outbound-callerid:2] GotoIf(“SIP/2001-098943c0”, “1?start”) in new stack
– Goto (macro-outbound-callerid,s,4)
– Executing [s@macro-outbound-callerid:4] NoOp(“SIP/2001-098943c0”, “REALCALLERIDNUM is 2001”) in new stack
– Executing [s@macro-outbound-callerid:5] GotoIf(“SIP/2001-098943c0”, “1?normcid”) in new stack
– Goto (macro-outbound-callerid,s,10)
– Executing [s@macro-outbound-callerid:10] Set(“SIP/2001-098943c0”, “USEROUTCID=”) in new stack
– Executing [s@macro-outbound-callerid:11] Set(“SIP/2001-098943c0”, “EMERGENCYCID=”) in new stack
– Executing [s@macro-outbound-callerid:12] Set(“SIP/2001-098943c0”, “TRUNKOUTCID=”) in new stack
– Executing [s@macro-outbound-callerid:13] GotoIf(“SIP/2001-098943c0”, “1?trunkcid”) in new stack
– Goto (macro-outbound-callerid,s,17)
– Executing [s@macro-outbound-callerid:17] GotoIf(“SIP/2001-098943c0”, “1?usercid”) in new stack
– Goto (macro-outbound-callerid,s,19)
– Executing [s@macro-outbound-callerid:19] GotoIf(“SIP/2001-098943c0”, “1?report”) in new stack
– Goto (macro-outbound-callerid,s,23)
– Executing [s@macro-outbound-callerid:23] NoOp(“SIP/2001-098943c0”, “CallerID set to “interno1” <2001>”) in new stack
– Executing [s@macro-dialout-trunk:12] AGI(“SIP/2001-098943c0”, “fixlocalprefix”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
== fixlocalprefix: Dialpattern 2. matched. 2245 -> 2245
– AGI Script fixlocalprefix completed, returning 0
– Executing [s@macro-dialout-trunk:13] Set(“SIP/2001-098943c0”, “OUTNUM=2245”) in new stack
– Executing [s@macro-dialout-trunk:14] Set(“SIP/2001-098943c0”, “custom=SIP/samsung”) in new stack
– Executing [s@macro-dialout-trunk:15] GotoIf(“SIP/2001-098943c0”, “1?gocall”) in new stack
– Goto (macro-dialout-trunk,s,17)
– Executing [s@macro-dialout-trunk:17] Macro(“SIP/2001-098943c0”, “dialout-trunk-predial-hook|”) in new stack
– Executing [s@macro-dialout-trunk:18] GotoIf(“SIP/2001-098943c0”, “0?bypass|1”) in new stack
– Executing [s@macro-dialout-trunk:19] GotoIf(“SIP/2001-098943c0”, “0?customtrunk”) in new stack
– Executing [s@macro-dialout-trunk:20] Dial(“SIP/2001-098943c0”, “SIP/samsung/2245|300|”) in new stack
Audio is at 77.0.2.166 port 12192
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 77.0.0.134:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 77.0.2.166:5060;branch=z9hG4bK5a80fa7b;rport
From: “interno1” sip:[email protected];tag=as0c5725b0
To: sip:[email protected]
Contact: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 10 Nov 2008 16:27:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 234
v=0
o=root 3137 3137 IN IP4 77.0.2.166
s=session
c=IN IP4 77.0.2.166
t=0 0
m=audio 12192 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
-- Called samsung/2245
trixbox1*CLI>
<— SIP read from 77.0.0.134:5060 —>
SIP/2.0 100 Trying
From: "interno1"sip:[email protected];tag=as0c5725b0
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 INVITE
Via: SIP/2.0/UDP 77.0.2.166:5060;rport=5060;branch=z9hG4bK5a80fa7b
Supported: 100rel,replaces
Contact: sip:[email protected]:5060
Content-Length: 0
<------------->
— (9 headers 0 lines) —
<— SIP read from 77.0.0.134:5060 —>
SIP/2.0 180 Ringing
From: "interno1"sip:[email protected];tag=as0c5725b0
To: sip:[email protected];tag=5bfe6c0-4d000086-13c4-116d93-72a6fd3a-116d93
Call-ID: [email protected]
CSeq: 102 INVITE
Via: SIP/2.0/UDP 77.0.2.166:5060;rport=5060;branch=z9hG4bK5a80fa7b
Supported: 100rel,replaces
Contact: sip:[email protected]:5060
Content-Length: 0
<------------->
— (9 headers 0 lines) —
== Parsing ‘/etc/asterisk/manager.conf’: Found
== Parsing ‘/etc/asterisk/manager_additional.conf’: Found
== Parsing ‘/etc/asterisk/manager_custom.conf’: Found
== Manager ‘admin’ logged on from 127.0.0.1
– SIP/samsung-0989ac68 is ringing
== Manager ‘admin’ logged off from 127.0.0.1
trixbox1*CLI>
<— SIP read from 77.0.0.134:5060 —>
SIP/2.0 200 OK
From: "interno1"sip:[email protected];tag=as0c5725b0
To: sip:[email protected];tag=5bfe6c0-4d000086-13c4-116d93-72a6fd3a-116d93
Call-ID: [email protected]
CSeq: 102 INVITE
Via: SIP/2.0/UDP 77.0.2.166:5060;rport=5060;branch=z9hG4bK5a80fa7b
Supported: 100rel,replaces
Contact: sip:[email protected]:5060
Content-Type: application/SDP
Content-Length: 197
v=0
o=SAMSUNG_SIP_GATEWAY 1142165518 0 IN IP4 77.0.0.134
s=SIP_CALL
c=IN IP4 77.0.0.191
t=0 0
m=audio 30058 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
<------------->
— (10 headers 9 lines) —
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 77.0.0.191:30058
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 77.0.0.191:30058
list_route: hop: sip:[email protected]:5060
set_destination: Parsing sip:[email protected]:5060 for address/port to send to
set_destination: set destination to 77.0.0.134, port 5060
Transmitting (NAT) to 77.0.0.134:5060:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 77.0.2.166:5060;branch=z9hG4bK29307bac;rport
From: “interno1” sip:[email protected];tag=as0c5725b0
To: sip:[email protected];tag=5bfe6c0-4d000086-13c4-116d93-72a6fd3a-116d93
Contact: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
-- SIP/samsung-0989ac68 answered SIP/2001-098943c0
== Parsing ‘/etc/asterisk/manager.conf’: Found
== Parsing ‘/etc/asterisk/manager_additional.conf’: Found
== Parsing ‘/etc/asterisk/manager_custom.conf’: Found
== Manager ‘admin’ logged on from 127.0.0.1
== Manager ‘admin’ logged off from 127.0.0.1
== Parsing ‘/etc/asterisk/manager.conf’: Found
== Parsing ‘/etc/asterisk/manager_additional.conf’: Found
== Parsing ‘/etc/asterisk/manager_custom.conf’: Found
== Manager ‘admin’ logged on from 127.0.0.1
== Manager ‘admin’ logged off from 127.0.0.1
Reliably Transmitting (NAT) to 77.0.0.134:5060:
OPTIONS sip:77.0.0.134 SIP/2.0
Via: SIP/2.0/UDP 77.0.2.166:5060;branch=z9hG4bK0bdcfd19;rport
From: “Unknown” sip:[email protected];tag=as45be884e
To: sip:77.0.0.134
Contact: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 10 Nov 2008 16:27:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
trixbox1*CLI>
<— SIP read from 77.0.0.134:5060 —>
SIP/2.0 200 OK
From: "Unknown"sip:[email protected];tag=as45be884e
To: sip:77.0.0.134;tag=5bfe960-4d000086-13c4-116d9f-2b5972a6-116d9f
Call-ID: [email protected]
CSeq: 102 OPTIONS
Allow: INVITE,ACK,CANCEL,BYE,PRACK,OPTIONS,UPDATE
Accept: application/sdp
Accept-Encoding: gzip
Supported: 100rel,replaces
Via: SIP/2.0/UDP 77.0.2.166:5060;rport=5060;branch=z9hG4bK0bdcfd19
Content-Length: 0
<------------->
— (11 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]’ Method: OPTIONS
…
Thank you all!
If you need other details ask me!
bye!
- polluz -