{\rtf1\ansi\ansicpg1252\deff0\deflang1033{\fonttbl{\f0\fswiss\fcharset0 Arial;}} {\*\generator Msftedit 5.41.15.1507;}\viewkind4\uc1\pard\f0\fs20 Only G729 is allowed through the IAX trunk\par \par -- Executing Macro("SIP/901-b7e12c98", "DialOut|00402042997|60|T|g1") in new stack\par -- Executing GotoIf("SIP/901-b7e12c98", "0?21:2") in new stack\par -- Goto (macro-DialOut,s,2)\par -- Executing GotoIf("SIP/901-b7e12c98", "0?31:3") in new stack\par -- Goto (macro-DialOut,s,3)\par -- Executing GotoIf("SIP/901-b7e12c98", "0?41:4") in new stack\par -- Goto (macro-DialOut,s,4)\par -- Executing GotoIf("SIP/901-b7e12c98", "1?51:5") in new stack\par -- Goto (macro-DialOut,s,51)\par -- Executing Macro("SIP/901-b7e12c98", "DialTrunk|00402042997|60|T") in new stack\par -- Executing Set("SIP/901-b7e12c98", "CALLERID(number)=894230900") in new stack\par -- Executing Set("SIP/901-b7e12c98", "SIP_CODEC=g729") in new stack\par -- Executing Dial("SIP/901-b7e12c98", "IAX2/IAXTRUNK01/00402042997|60|T") in new stack\par Oct 5 08:57:48 WARNING[8637]: chan_iax2.c:8003 iax2_request: Unable to create translator path for unknown to ulaw on IAX2/IAXTRUNK01-16384\par -- Hungup 'IAX2/IAXTRUNK01-16384'\par Oct 5 08:57:48 NOTICE[8637]: app_dial.c:1049 dial_exec_full: Unable to create channel of type 'IAX2' (cause 0 - Unknown)\par \par /var/log/asterisk/messages\par Oct 5 09:46:14 WARNING[10816] chan_iax2.c: Unable to create translator path for unknown to ulaw on IAX2/IAXTRUNK01-16384\par Oct 5 09:46:14 NOTICE[10816] app_dial.c: Unable to create channel of type 'IAX2' (cause 0 - Unknown)\par Oct 5 09:47:14 WARNING[10850] chan_iax2.c: Unable to create translator path for unknown to ulaw on IAX2/IAXTRUNK01-16385\par Oct 5 09:47:14 NOTICE[10850] app_dial.c: Unable to create channel of type 'IAX2' (cause 0 - Unknown)\par \par \par After allowing G711ulaw through the IAX2 trunk\par \par \par -- Executing Macro("SIP/901-08527208", "DialOut|00402042997|60|T|g1") in new stack\par -- Executing GotoIf("SIP/901-08527208", "0?21:2") in new stack\par -- Goto (macro-DialOut,s,2)\par -- Executing GotoIf("SIP/901-08527208", "0?31:3") in new stack\par -- Goto (macro-DialOut,s,3)\par -- Executing GotoIf("SIP/901-08527208", "0?41:4") in new stack\par -- Goto (macro-DialOut,s,4)\par -- Executing GotoIf("SIP/901-08527208", "1?51:5") in new stack\par -- Goto (macro-DialOut,s,51)\par -- Executing Macro("SIP/901-08527208", "DialTrunk|00402042997|60|T") in new stack\par -- Executing Set("SIP/901-08527208", "CALLERID(number)=894230900") in new stack\par -- Executing Set("SIP/901-08527208", "SIP_CODEC=g729") in new stack\par -- Executing Dial("SIP/901-08527208", "IAX2/IAXTRUNK01/00402042997|60|T") in new stack\par -- Called IAXTRUNK01/00402042997\par -- Call accepted by 192.168.240.52 (format g729)\par -- Format for call is g729\par Oct 5 09:00:11 WARNING[8719]: channel.c:2341 set_format: Unable to find a codec translation path from g729 to ulaw\par Oct 5 09:00:11 WARNING[8719]: channel.c:2341 set_format: Unable to find a codec translation path from g729 to ulaw\par Oct 5 09:00:18 WARNING[8719]: channel.c:2341 set_format: Unable to find a codec translation path from g729 to ulaw\par Oct 5 09:00:18 WARNING[8764]: chan_sip.c:2552 sip_write: Asked to transmit frame type 256, while native formats is 4 (read/write = 4/4)\par -- IAX2/IAXTRUNK01-16385 is making progress passing it to SIP/901-08527208\par Oct 5 09:00:18 NOTICE[8764]: chan_sip.c:2510 try_suggested_sip_codec: Changing codec to 'g729' for this call because of $\{SIP_CODEC) variable\par Oct 5 09:00:18 WARNING[8764]: chan_sip.c:2552 sip_write: Asked to transmit frame type 256, while native formats is 4 (read/write = 4/4)\par Oct 5 09:00:18 WARNING[8764]: chan_sip.c:2552 sip_write: Asked to transmit frame type 256, while native formats is 4 (read/write = 4/4)\par Oct 5 09:00:18 WARNING[8764]: chan_sip.c:2552 sip_write: Asked to transmit frame type 256, while native formats is 4 (read/write = 4/4)\par Oct 5 09:00:18 WARNING[8764]: chan_sip.c:2552 sip_write: Asked to transmit frame type 256, while native formats is 4 (read/write = 4/4)\par Oct 5 09:00:18 WARNING[8764]: chan_sip.c:2552 sip_write: Asked to transmit frame type 256, while native formats is 4 (read/write = 4/4)\par Oct 5 09:00:18 WARNING[8764]: chan_sip.c:2552 sip_write: Asked to transmit frame type 256, while native formats is 4 (read/write = 4/4)\par Oct 5 09:00:18 WARNING[8764]: chan_sip.c:2552 sip_write: Asked to transmit frame type 256, while native formats is 4 (read/write = 4/4)\par Oct 5 09:00:18 WARNING[8764]: chan_sip.c:2552 sip_write: Asked to transmit frame type 256, while native formats is 4 (read/write = 4/4)\par Oct 5 09:00:18 WARNING[8764]: chan_sip.c:2552 sip_write: Asked to transmit frame type 256, while native formats is 4 (read/write = 4/4)\par Oct 5 09:00:18 WARNING[8764]: chan_sip.c:2552 sip_write: Asked to transmit frame type 256, while native formats is 4 (read/write = 4/4)\par Oct 5 09:00:18 WARNING[8764]: channel.c:2341 set_format: Unable to find a codec translation path from g729 to ulaw\par Oct 5 09:00:18 WARNING[8764]: channel.c:2341 set_format: Unable to find a codec translation path from g729 to ulaw\par -- IAX2/IAXTRUNK01-16385 answered SIP/901-08527208\par Oct 5 09:00:22 NOTICE[8764]: chan_sip.c:2510 try_suggested_sip_codec: Changing codec to 'g729' for this call because of $\{SIP_CODEC) variable\par Oct 5 09:00:22 NOTICE[8764]: chan_sip.c:2510 try_suggested_sip_codec: Changing codec to 'g729' for this call because of $\{SIP_CODEC) variable\par \par \par With sip debug and iax2 debug turned on.\par \par <-- SIP read from 192.168.240.162:2051: \par INVITE sip:00402042997@192.168.240.53;user=phone SIP/2.0\par Via: SIP/2.0/UDP 192.168.240.162:2051;branch=z9hG4bK-yhccsv6wrdbp;rport\par From: "Island test" ;tag=2lmw5oebdb\par To: \par Call-ID: 3c2673165f37-agv4vpcj3y9g@snom360-000413230EE8\par CSeq: 1 INVITE\par Max-Forwards: 70\par Contact: ;flow-id=1\par P-Key-Flags: resolution="31x13", keys="4"\par User-Agent: snom360/6.2.3\par Accept: application/sdp\par Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO\par Allow-Events: talk, hold, refer\par Supported: timer, 100rel, replaces, callerid\par Session-Expires: 3600;refresher=uas\par Min-SE: 90\par Content-Type: application/sdp\par Content-Length: 359\par \par v=0\par o=root 1058298768 1058298768 IN IP4 192.168.240.162\par s=call\par c=IN IP4 192.168.240.162\par t=0 0\par m=audio 63164 RTP/AVP 18 0 101\par a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:pbkRnuLG99vM64pntZaFExu4O3nDayNla/5dXyXZ\par =rtpmap:18 g729/8000\par a=rtpmap:0 pcmu/8000\par a=rtpmap:101 telephone-event/8000\par a=fmtp:101 0-16\par a=ptime:20\par a=encryption:optional\par a=sendrecv\par \par --- (18 headers 14 lines)---\par Using INVITE request as basis request - 3c2673165f37-agv4vpcj3y9g@snom360-000413230EE8\par Sending to 192.168.240.162 : 2051 (NAT)\par Found user '901'\par Found RTP audio format 18\par Found RTP audio format 0\par Found RTP audio format 101\par Peer audio RTP is at port 192.168.240.162:63164\par Found description format g729\par Found description format pcmu\par Found description format telephone-event\par Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing), combined - 0x104 (ulaw|g729)\par Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)\par Looking for 00402042997 in internal-sip (domain 192.168.240.53)\par list_route: hop: \par Transmitting (no NAT) to 192.168.240.162:2051:\par SIP/2.0 100 Trying\par Via: SIP/2.0/UDP 192.168.240.162:2051;branch=z9hG4bK-yhccsv6wrdbp;rport;received=192.168.240.162\par From: "Island test" ;tag=2lmw5oebdb\par To: \par Call-ID: 3c2673165f37-agv4vpcj3y9g@snom360-000413230EE8\par CSeq: 1 INVITE\par User-Agent: Asterisk PBX\par Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY\par Contact: \par Content-Length: 0\par \par \par ---\par -- Executing Macro("SIP/901-09d7d3a8", "DialOut|00402042997|60|T|g1") in new stack\par -- Executing GotoIf("SIP/901-09d7d3a8", "0?21:2") in new stack\par -- Goto (macro-DialOut,s,2)\par -- Executing GotoIf("SIP/901-09d7d3a8", "0?31:3") in new stack\par -- Goto (macro-DialOut,s,3)\par -- Executing GotoIf("SIP/901-09d7d3a8", "0?41:4") in new stack\par -- Goto (macro-DialOut,s,4)\par -- Executing GotoIf("SIP/901-09d7d3a8", "1?51:5") in new stack\par -- Goto (macro-DialOut,s,51)\par -- Executing Macro("SIP/901-09d7d3a8", "DialTrunk|00402042997|60|T") in new stack\par -- Executing Set("SIP/901-09d7d3a8", "CALLERID(number)=894230900") in new stack\par -- Executing Dial("SIP/901-09d7d3a8", "IAX2/IAXTRUNK01/00402042997|60|T") in new stack\par Oct 5 09:52:00 WARNING[11103]: chan_iax2.c:8003 iax2_request: Unable to create translator path for unknown to ulaw on IAX2/IAXTRUNK01-16384\par Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: HANGUP \par Timestamp: 00011ms SCall: 16384 DCall: 00000 [192.168.240.52:4569]\par CAUSE CODE : 0\par \par -- Hungup 'IAX2/IAXTRUNK01-16384'\par Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL \par Timestamp: 00000ms SCall: 00000 DCall: 16384 [192.168.240.52:4569]\par Oct 5 09:52:00 NOTICE[11103]: app_dial.c:1049 dial_exec_full: Unable to create channel of type 'IAX2' (cause 0 - Unknown)\par == Everyone is busy/congested at this time (1:0/0/1)\par localhost*CLI> \par <-- SIP read from 192.168.240.162:2051: \par CANCEL sip:00402042997@192.168.240.53;user=phone SIP/2.0\par Via: SIP/2.0/UDP 192.168.240.162:2051;branch=z9hG4bK-yhccsv6wrdbp;rport\par From: "Island test" ;tag=2lmw5oebdb\par To: \par Call-ID: 3c2673165f37-agv4vpcj3y9g@snom360-000413230EE8\par CSeq: 1 CANCEL\par Max-Forwards: 70\par Contact: ;flow-id=1\par Reason: SIP;cause=487;text="Request terminated by user"\par Content-Length: 0\par \par \par --- (10 headers 0 lines)---\par Sending to 192.168.240.162 : 2051 (NAT)\par Reliably Transmitting (NAT) to 192.168.240.162:2051:\par SIP/2.0 487 Request Terminated\par Via: SIP/2.0/UDP 192.168.240.162:2051;branch=z9hG4bK-yhccsv6wrdbp;received=192.168.240.162;rport=2051\par From: "Island test" ;tag=2lmw5oebdb\par To: ;tag=as11509ee9\par Call-ID: 3c2673165f37-agv4vpcj3y9g@snom360-000413230EE8\par CSeq: 1 INVITE\par User-Agent: Asterisk PBX\par Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY\par Contact: \par Content-Length: 0\par \par \par ---\par Transmitting (NAT) to 192.168.240.162:2051:\par SIP/2.0 200 OK\par Via: SIP/2.0/UDP 192.168.240.162:2051;branch=z9hG4bK-yhccsv6wrdbp;received=192.168.240.162;rport=2051\par From: "Island test" ;tag=2lmw5oebdb\par To: ;tag=as11509ee9\par Call-ID: 3c2673165f37-agv4vpcj3y9g@snom360-000413230EE8\par CSeq: 1 CANCEL\par User-Agent: Asterisk PBX\par Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY\par Contact: \par Content-Length: 0\par }