Christoph Fürstaller
2007-Mar-14 01:16 UTC
[asterisk-users] strange things on call transfer
-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I'm setting up an Asterisk system which is connected to an Alcatel 4400 PBX. On the * I permit g729 and gsm as codecs. If I try to transfer a call by hitting the # key, I get this messages and nothing happens on the phone: WARNING[30110]: codec_ilbc.c:175 ilbctolin_framein: Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP (4)? But I'm not allowing ilbc as codec? If I prevent * from loading ilbc I tet this error and the call is hung up, when transfering: Mar 14 09:04:58 WARNING[30436]: channel.c:2380 set_format: Unable to find a codec translation path from ilbc to slin Mar 14 09:04:58 WARNING[30436]: channel.c:2380 set_format: Unable to find a codec translation path from ilbc to slin Mar 14 09:04:58 WARNING[30436]: channel.c:2752 ast_channel_make_compatible: No path to translate from Zap/31-1(72) to SIP/374-08199e50(1024) Mar 14 09:04:58 WARNING[30436]: channel.c:3632 ast_channel_bridge: Can't make Zap/31-1 and SIP/374-08199e50 compatible Mar 14 09:04:58 WARNING[30436]: res_features.c:1385 ast_bridge_call: Bridge failed on channels Zap/31-1 and SIP/374-08199e50 Attached there is a SIP debug from such a call (with ilbc loaded) I hope someone can help me. chris... -----BEGIN PGP SIGNATURE----- Version: GnuPG v2.0.3 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFF969jR0exH8dhr/YRAgP0AJ94ygGEPYHtGvLS7McUTrRAP1IkCgCgozv6 rfuVGufsb8wQT3Iwl0ipXNg=hrcE -----END PGP SIGNATURE----- -------------- next part -------------- <-- SIP read from 172.28.20.4:2051: INVITE sip:374@172.28.2.30;user=phone SIP/2.0 Via: SIP/2.0/UDP 172.28.20.4:2051;branch=z9hG4bK-0wdg2nh9bdfn;rport From: "Test User3" <sip:104@172.28.2.30>;tag=48rd3g03e1 To: <sip:374@172.28.2.30;user=phone> Call-ID: 3c30f7fb186a-n2ptt0oiuivb@snom300-0004132520F7 CSeq: 1 INVITE Max-Forwards: 70 Contact: <sip:104@172.28.20.4:2051;line=7p0ffzkf>;flow-id=1 P-Key-Flags: keys="3" User-Agent: snom300/6.2.3 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Session-Expires: 3600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 473 v=0 o=root 1775117380 1775117380 IN IP4 172.28.20.4 s=call c=IN IP4 172.28.20.4 t=0 0 m=audio 59502 RTP/AVP 0 8 9 2 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:8TM5Qape/dsy05garT9U1EHtmdo6iXy52b2Dc/Kc a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=encryption:optional a=sendrecv --- (18 headers 19 lines) --- Using INVITE request as basis request - 3c30f7fb186a-n2ptt0oiuivb@snom300-0004132520F7 Sending to 172.28.20.4 : 2051 (NAT) Reliably Transmitting (no NAT) to 172.28.20.4:2051: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 172.28.20.4:2051;branch=z9hG4bK-0wdg2nh9bdfn;rport;received=172.28.20.4 From: "Test User3" <sip:104@172.28.2.30>;tag=48rd3g03e1 To: <sip:374@172.28.2.30;user=phone>;tag=as49d79f99 Call-ID: 3c30f7fb186a-n2ptt0oiuivb@snom300-0004132520F7 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm="xxx.xx", nonce="0986769d" Content-Length: 0 --- Scheduling destruction of call '3c30f7fb186a-n2ptt0oiuivb@snom300-0004132520F7' in 15000 ms Found user '104' salxpbx1*CLI> <-- SIP read from 172.28.20.4:2051: ACK sip:374@172.28.2.30;user=phone SIP/2.0 Via: SIP/2.0/UDP 172.28.20.4:2051;branch=z9hG4bK-0wdg2nh9bdfn;rport From: "Test User3" <sip:104@172.28.2.30>;tag=48rd3g03e1 To: <sip:374@172.28.2.30;user=phone>;tag=as49d79f99 Call-ID: 3c30f7fb186a-n2ptt0oiuivb@snom300-0004132520F7 CSeq: 1 ACK Max-Forwards: 70 Contact: <sip:104@172.28.20.4:2051;line=7p0ffzkf>;flow-id=1 Content-Length: 0 --- (9 headers 0 lines) --- salxpbx1*CLI> <-- SIP read from 172.28.20.4:2051: INVITE sip:374@172.28.2.30;user=phone SIP/2.0 Via: SIP/2.0/UDP 172.28.20.4:2051;branch=z9hG4bK-8tixe2g5vsdv;rport From: "Test User3" <sip:104@172.28.2.30>;tag=48rd3g03e1 To: <sip:374@172.28.2.30;user=phone> Call-ID: 3c30f7fb186a-n2ptt0oiuivb@snom300-0004132520F7 CSeq: 2 INVITE Max-Forwards: 70 Contact: <sip:104@172.28.20.4:2051;line=7p0ffzkf>;flow-id=1 P-Key-Flags: keys="3" User-Agent: snom300/6.2.3 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Session-Expires: 3600;refresher=uas Min-SE: 90 Proxy-Authorization: Digest username="104",realm="xxx.xx",nonce="0986769d",uri="sip:374@172.28.2.30;user=phone",response="0e5cb3b8871eff4c6670a6f277ef3594",algorithm=md5 Content-Type: application/sdp Content-Length: 473 v=0 o=root 1775117380 1775117380 IN IP4 172.28.20.4 s=call c=IN IP4 172.28.20.4 t=0 0 m=audio 59502 RTP/AVP 0 8 9 2 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:8TM5Qape/dsy05garT9U1EHtmdo6iXy52b2Dc/Kc a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=encryption:optional a=sendrecv --- (19 headers 19 lines) --- Using INVITE request as basis request - 3c30f7fb186a-n2ptt0oiuivb@snom300-0004132520F7 Sending to 172.28.20.4 : 2051 (NAT) Found user '104' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 9 Found RTP audio format 2 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 101 Peer audio RTP is at port 172.28.20.4:59502 Peer video RTP is at port 172.28.20.4:65535 Found description format pcmu Found description format pcma Found description format g722 Found description format g726-32 Found description format gsm Found description format g729 Found description format g723 Found description format telephone-event Capabilities: us - 0x102 (gsm|g729), peer - audio=0x11f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0x102 (gsm|g729) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 374 in intern (domain 172.28.2.30;user=phone) list_route: hop: <sip:104@172.28.20.4:2051;line=7p0ffzkf> Transmitting (no NAT) to 172.28.20.4:2051: SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.28.20.4:2051;branch=z9hG4bK-8tixe2g5vsdv;rport;received=172.28.20.4 From: "Test User3" <sip:104@172.28.2.30>;tag=48rd3g03e1 To: <sip:374@172.28.2.30;user=phone> Call-ID: 3c30f7fb186a-n2ptt0oiuivb@snom300-0004132520F7 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:374@172.28.2.30> Content-Length: 0 Transmitting (no NAT) to 172.28.20.4:2051: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 172.28.20.4:2051;branch=z9hG4bK-8tixe2g5vsdv;rport;received=172.28.20.4 From: "Test User3" <sip:104@172.28.2.30>;tag=48rd3g03e1 To: <sip:374@172.28.2.30;user=phone>;tag=as31ca24ed Call-ID: 3c30f7fb186a-n2ptt0oiuivb@snom300-0004132520F7 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:374@172.28.2.30> Content-Length: 0 --- -- SIP/374-081b0378 answered SIP/104-0819a1f8 We're at 172.28.2.30 port 16206 Video is at 172.28.2.30 port 18568 Adding codec 0x100 (g729) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 172.28.20.4:2051: SIP/2.0 200 OK Via: SIP/2.0/UDP 172.28.20.4:2051;branch=z9hG4bK-8tixe2g5vsdv;rport;received=172.28.20.4 From: "Test User3" <sip:104@172.28.2.30>;tag=48rd3g03e1 To: <sip:374@172.28.2.30;user=phone>;tag=as31ca24ed Call-ID: 3c30f7fb186a-n2ptt0oiuivb@snom300-0004132520F7 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:374@172.28.2.30> Content-Type: application/sdp Content-Length: 260 v=0 o=root 29900 29900 IN IP4 172.28.2.30 s=session c=IN IP4 172.28.2.30 t=0 0 m=audio 16206 RTP/AVP 18 3 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- salxpbx1*CLI> <-- SIP read from 172.28.20.4:2051: ACK sip:374@172.28.2.30 SIP/2.0 Via: SIP/2.0/UDP 172.28.20.4:2051;branch=z9hG4bK-hjacxzcexdgk;rport From: "Test User3" <sip:104@172.28.2.30>;tag=48rd3g03e1 To: <sip:374@172.28.2.30;user=phone>;tag=as31ca24ed Call-ID: 3c30f7fb186a-n2ptt0oiuivb@snom300-0004132520F7 CSeq: 2 ACK Max-Forwards: 70 Contact: <sip:104@172.28.20.4:2051;line=7p0ffzkf>;flow-id=1 Content-Length: 0 --- (9 headers 0 lines) --- 12 headers, 0 lines Reliably Transmitting (no NAT) to 172.28.20.4:2051: OPTIONS sip:104@172.28.20.4:2051;line=7p0ffzkf SIP/2.0 Via: SIP/2.0/UDP 172.28.2.30:5060;branch=z9hG4bK7e2b5210;rport From: "asterisk" <sip:asterisk@xxx.xx>;tag=as68d9b69b To: <sip:104@172.28.20.4:2051;line=7p0ffzkf> Contact: <sip:asterisk@172.28.2.30> Call-ID: 30e124937558a11d3e9f397947b378b8@xxx.xx CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 14 Mar 2007 07:45:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- salxpbx1*CLI> <-- SIP read from 172.28.20.4:2051: SIP/2.0 200 OK Via: SIP/2.0/UDP 172.28.2.30:5060;branch=z9hG4bK7e2b5210;rport=5060 From: "asterisk" <sip:asterisk@xxx.xx>;tag=as68d9b69b To: <sip:104@172.28.20.4:2051;line=7p0ffzkf> Call-ID: 30e124937558a11d3e9f397947b378b8@xxx.xx CSeq: 102 OPTIONS Contact: <sip:104@172.28.20.4:2051;line=7p0ffzkf>;flow-id=1 User-Agent: snom300/6.2.3 Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Content-Length: 0 --- (14 headers 0 lines) --- Destroying call '30e124937558a11d3e9f397947b378b8@xxx.xx' Mar 14 08:45:27 WARNING[30110]: codec_ilbc.c:175 ilbctolin_framein: Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP (4)? Mar 14 08:45:27 WARNING[30110]: codec_ilbc.c:175 ilbctolin_framein: Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP (4)? Mar 14 08:45:27 WARNING[30110]: codec_ilbc.c:175 ilbctolin_framein: Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP (4)? Mar 14 08:45:27 WARNING[30110]: codec_ilbc.c:175 ilbctolin_framein: Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP (4)? <-- SIP read from 172.28.20.4:2051: BYE sip:374@172.28.2.30 SIP/2.0 Via: SIP/2.0/UDP 172.28.20.4:2051;branch=z9hG4bK-e9vrp3jilhg7;rport From: "Test User3" <sip:104@172.28.2.30>;tag=48rd3g03e1 To: <sip:374@172.28.2.30;user=phone>;tag=as31ca24ed Call-ID: 3c30f7fb186a-n2ptt0oiuivb@snom300-0004132520F7 CSeq: 3 BYE Max-Forwards: 70 Contact: <sip:104@172.28.20.4:2051;line=7p0ffzkf>;flow-id=1 User-Agent: snom300/6.2.3 RTP-RxStat: Total_Rx_Pkts=350,Rx_Pkts=350,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0 RTP-TxStat: Total_Tx_Pkts=355,Tx_Pkts=355,Remote_Tx_Pkts=0 Content-Length: 0 --- (12 headers 0 lines) --- Sending to 172.28.20.4 : 2051 (NAT) Transmitting (NAT) to 172.28.20.4:2051: SIP/2.0 200 OK Via: SIP/2.0/UDP 172.28.20.4:2051;branch=z9hG4bK-e9vrp3jilhg7;received=172.28.20.4;rport=2051 From: "Test User3" <sip:104@172.28.2.30>;tag=48rd3g03e1 To: <sip:374@172.28.2.30;user=phone>;tag=as31ca24ed Call-ID: 3c30f7fb186a-n2ptt0oiuivb@snom300-0004132520F7 CSeq: 3 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:374@172.28.2.30> Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing --- == Spawn extension (intern, h, 109) exited non-zero on 'SIP/104-0819a1f8' Destroying call '3c30f7fb186a-n2ptt0oiuivb@snom300-0004132520F7'
Hello, Can you tell about your configuration ( connection between Alacaltel 4400 and Asterisk ?) : what hardware ? what config files ? Thank you Minh VO MIVOC Systemes SAS FRANCE ----- Original Message ----- From: "Christoph F?rstaller" <christoph.fuerstaller@kurtkrenn.com> To: <asterisk-users@lists.digium.com> Sent: Wednesday, March 14, 2007 9:16 AM Subject: [asterisk-users] strange things on call transfer> -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Hi, > > I'm setting up an Asterisk system which is connected to an Alcatel 4400 > PBX. On the * I permit g729 and gsm as codecs. If I try to transfer a > call by hitting the # key, I get this messages and nothing happens on > the phone: > > WARNING[30110]: codec_ilbc.c:175 ilbctolin_framein: Huh? An ilbc frame > that isn't a multiple of 50 bytes long from RTP (4)? > > But I'm not allowing ilbc as codec? If I prevent * from loading ilbc I > tet this error and the call is hung up, when transfering: > > Mar 14 09:04:58 WARNING[30436]: channel.c:2380 set_format: Unable to > find a codec translation path from ilbc to slin > Mar 14 09:04:58 WARNING[30436]: channel.c:2380 set_format: Unable to > find a codec translation path from ilbc to slin > Mar 14 09:04:58 WARNING[30436]: channel.c:2752 > ast_channel_make_compatible: No path to translate from Zap/31-1(72) to > SIP/374-08199e50(1024) > Mar 14 09:04:58 WARNING[30436]: channel.c:3632 ast_channel_bridge: Can't > make Zap/31-1 and SIP/374-08199e50 compatible > Mar 14 09:04:58 WARNING[30436]: res_features.c:1385 ast_bridge_call: > Bridge failed on channels Zap/31-1 and SIP/374-08199e50 > > Attached there is a SIP debug from such a call (with ilbc loaded) > > I hope someone can help me. > > chris... > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v2.0.3 (GNU/Linux) > Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org > > iD8DBQFF969jR0exH8dhr/YRAgP0AJ94ygGEPYHtGvLS7McUTrRAP1IkCgCgozv6 > rfuVGufsb8wQT3Iwl0ipXNg> =hrcE > -----END PGP SIGNATURE----- > > ---------------------------------------------------------------------------------------> Orange vous informe que cet e-mail a ete controle par l'anti-virus mail. > Aucun virus connu a ce jour par nos services n'a ete detecte. > >---------------------------------------------------------------------------- ----> <-- SIP read from 172.28.20.4:2051: > INVITE sip:374@172.28.2.30;user=phone SIP/2.0 > Via: SIP/2.0/UDP 172.28.20.4:2051;branch=z9hG4bK-0wdg2nh9bdfn;rport > From: "Test User3" <sip:104@172.28.2.30>;tag=48rd3g03e1 > To: <sip:374@172.28.2.30;user=phone> > Call-ID: 3c30f7fb186a-n2ptt0oiuivb@snom300-0004132520F7 > CSeq: 1 INVITE > Max-Forwards: 70 > Contact: <sip:104@172.28.20.4:2051;line=7p0ffzkf>;flow-id=1 > P-Key-Flags: keys="3" > User-Agent: snom300/6.2.3 > Accept: application/sdp > Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,MESSAGE, INFO> Allow-Events: talk, hold, refer > Supported: timer, 100rel, replaces, callerid > Session-Expires: 3600;refresher=uas > Min-SE: 90 > Content-Type: application/sdp > Content-Length: 473 > > v=0 > o=root 1775117380 1775117380 IN IP4 172.28.20.4 > s=call > c=IN IP4 172.28.20.4 > t=0 0 > m=audio 59502 RTP/AVP 0 8 9 2 3 18 4 101 > a=crypto:1 AES_CM_128_HMAC_SHA1_32inline:8TM5Qape/dsy05garT9U1EHtmdo6iXy52b2Dc/Kc> a=rtpmap:0 pcmu/8000 > a=rtpmap:8 pcma/8000 > a=rtpmap:9 g722/8000 > a=rtpmap:2 g726-32/8000 > a=rtpmap:3 gsm/8000 > a=rtpmap:18 g729/8000 > a=rtpmap:4 g723/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > a=encryption:optional > a=sendrecv > > --- (18 headers 19 lines) --- > Using INVITE request as basis request -3c30f7fb186a-n2ptt0oiuivb@snom300-0004132520F7> Sending to 172.28.20.4 : 2051 (NAT) > Reliably Transmitting (no NAT) to 172.28.20.4:2051: > SIP/2.0 407 Proxy Authentication Required > Via: SIP/2.0/UDP172.28.20.4:2051;branch=z9hG4bK-0wdg2nh9bdfn;rport;received=172.28.20.4> From: "Test User3" <sip:104@172.28.2.30>;tag=48rd3g03e1 > To: <sip:374@172.28.2.30;user=phone>;tag=as49d79f99 > Call-ID: 3c30f7fb186a-n2ptt0oiuivb@snom300-0004132520F7 > CSeq: 1 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Proxy-Authenticate: Digest algorithm=MD5, realm="xxx.xx", nonce="0986769d" > Content-Length: 0 > > > --- > Scheduling destruction of call'3c30f7fb186a-n2ptt0oiuivb@snom300-0004132520F7' in 15000 ms> Found user '104' > salxpbx1*CLI> > <-- SIP read from 172.28.20.4:2051: > ACK sip:374@172.28.2.30;user=phone SIP/2.0 > Via: SIP/2.0/UDP 172.28.20.4:2051;branch=z9hG4bK-0wdg2nh9bdfn;rport > From: "Test User3" <sip:104@172.28.2.30>;tag=48rd3g03e1 > To: <sip:374@172.28.2.30;user=phone>;tag=as49d79f99 > Call-ID: 3c30f7fb186a-n2ptt0oiuivb@snom300-0004132520F7 > CSeq: 1 ACK > Max-Forwards: 70 > Contact: <sip:104@172.28.20.4:2051;line=7p0ffzkf>;flow-id=1 > Content-Length: 0 > > > --- (9 headers 0 lines) --- > salxpbx1*CLI> > <-- SIP read from 172.28.20.4:2051: > INVITE sip:374@172.28.2.30;user=phone SIP/2.0 > Via: SIP/2.0/UDP 172.28.20.4:2051;branch=z9hG4bK-8tixe2g5vsdv;rport > From: "Test User3" <sip:104@172.28.2.30>;tag=48rd3g03e1 > To: <sip:374@172.28.2.30;user=phone> > Call-ID: 3c30f7fb186a-n2ptt0oiuivb@snom300-0004132520F7 > CSeq: 2 INVITE > Max-Forwards: 70 > Contact: <sip:104@172.28.20.4:2051;line=7p0ffzkf>;flow-id=1 > P-Key-Flags: keys="3" > User-Agent: snom300/6.2.3 > Accept: application/sdp > Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,MESSAGE, INFO> Allow-Events: talk, hold, refer > Supported: timer, 100rel, replaces, callerid > Session-Expires: 3600;refresher=uas > Min-SE: 90 > Proxy-Authorization: Digestusername="104",realm="xxx.xx",nonce="0986769d",uri="sip:374@172.28.2.30;user =phone",response="0e5cb3b8871eff4c6670a6f277ef3594",algorithm=md5> Content-Type: application/sdp > Content-Length: 473 > > v=0 > o=root 1775117380 1775117380 IN IP4 172.28.20.4 > s=call > c=IN IP4 172.28.20.4 > t=0 0 > m=audio 59502 RTP/AVP 0 8 9 2 3 18 4 101 > a=crypto:1 AES_CM_128_HMAC_SHA1_32inline:8TM5Qape/dsy05garT9U1EHtmdo6iXy52b2Dc/Kc> a=rtpmap:0 pcmu/8000 > a=rtpmap:8 pcma/8000 > a=rtpmap:9 g722/8000 > a=rtpmap:2 g726-32/8000 > a=rtpmap:3 gsm/8000 > a=rtpmap:18 g729/8000 > a=rtpmap:4 g723/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > a=encryption:optional > a=sendrecv > > --- (19 headers 19 lines) --- > Using INVITE request as basis request -3c30f7fb186a-n2ptt0oiuivb@snom300-0004132520F7> Sending to 172.28.20.4 : 2051 (NAT) > Found user '104' > Found RTP audio format 0 > Found RTP audio format 8 > Found RTP audio format 9 > Found RTP audio format 2 > Found RTP audio format 3 > Found RTP audio format 18 > Found RTP audio format 4 > Found RTP audio format 101 > Peer audio RTP is at port 172.28.20.4:59502 > Peer video RTP is at port 172.28.20.4:65535 > Found description format pcmu > Found description format pcma > Found description format g722 > Found description format g726-32 > Found description format gsm > Found description format g729 > Found description format g723 > Found description format telephone-event > Capabilities: us - 0x102 (gsm|g729), peer - audio=0x11f(g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0x102 (gsm|g729)> Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1(telephone-event), combined - 0x1 (telephone-event)> Looking for 374 in intern (domain 172.28.2.30;user=phone) > list_route: hop: <sip:104@172.28.20.4:2051;line=7p0ffzkf> > Transmitting (no NAT) to 172.28.20.4:2051: > SIP/2.0 100 Trying > Via: SIP/2.0/UDP172.28.20.4:2051;branch=z9hG4bK-8tixe2g5vsdv;rport;received=172.28.20.4> From: "Test User3" <sip:104@172.28.2.30>;tag=48rd3g03e1 > To: <sip:374@172.28.2.30;user=phone> > Call-ID: 3c30f7fb186a-n2ptt0oiuivb@snom300-0004132520F7 > CSeq: 2 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Contact: <sip:374@172.28.2.30> > Content-Length: 0 > > > Transmitting (no NAT) to 172.28.20.4:2051: > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP172.28.20.4:2051;branch=z9hG4bK-8tixe2g5vsdv;rport;received=172.28.20.4> From: "Test User3" <sip:104@172.28.2.30>;tag=48rd3g03e1 > To: <sip:374@172.28.2.30;user=phone>;tag=as31ca24ed > Call-ID: 3c30f7fb186a-n2ptt0oiuivb@snom300-0004132520F7 > CSeq: 2 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Contact: <sip:374@172.28.2.30> > Content-Length: 0 > > > --- > -- SIP/374-081b0378 answered SIP/104-0819a1f8 > We're at 172.28.2.30 port 16206 > Video is at 172.28.2.30 port 18568 > Adding codec 0x100 (g729) to SDP > Adding codec 0x2 (gsm) to SDP > Adding non-codec 0x1 (telephone-event) to SDP > Reliably Transmitting (no NAT) to 172.28.20.4:2051: > SIP/2.0 200 OK > Via: SIP/2.0/UDP172.28.20.4:2051;branch=z9hG4bK-8tixe2g5vsdv;rport;received=172.28.20.4> From: "Test User3" <sip:104@172.28.2.30>;tag=48rd3g03e1 > To: <sip:374@172.28.2.30;user=phone>;tag=as31ca24ed > Call-ID: 3c30f7fb186a-n2ptt0oiuivb@snom300-0004132520F7 > CSeq: 2 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Contact: <sip:374@172.28.2.30> > Content-Type: application/sdp > Content-Length: 260 > > v=0 > o=root 29900 29900 IN IP4 172.28.2.30 > s=session > c=IN IP4 172.28.2.30 > t=0 0 > m=audio 16206 RTP/AVP 18 3 101 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > > --- > salxpbx1*CLI> > <-- SIP read from 172.28.20.4:2051: > ACK sip:374@172.28.2.30 SIP/2.0 > Via: SIP/2.0/UDP 172.28.20.4:2051;branch=z9hG4bK-hjacxzcexdgk;rport > From: "Test User3" <sip:104@172.28.2.30>;tag=48rd3g03e1 > To: <sip:374@172.28.2.30;user=phone>;tag=as31ca24ed > Call-ID: 3c30f7fb186a-n2ptt0oiuivb@snom300-0004132520F7 > CSeq: 2 ACK > Max-Forwards: 70 > Contact: <sip:104@172.28.20.4:2051;line=7p0ffzkf>;flow-id=1 > Content-Length: 0 > > > --- (9 headers 0 lines) --- > 12 headers, 0 lines > Reliably Transmitting (no NAT) to 172.28.20.4:2051: > OPTIONS sip:104@172.28.20.4:2051;line=7p0ffzkf SIP/2.0 > Via: SIP/2.0/UDP 172.28.2.30:5060;branch=z9hG4bK7e2b5210;rport > From: "asterisk" <sip:asterisk@xxx.xx>;tag=as68d9b69b > To: <sip:104@172.28.20.4:2051;line=7p0ffzkf> > Contact: <sip:asterisk@172.28.2.30> > Call-ID: 30e124937558a11d3e9f397947b378b8@xxx.xx > CSeq: 102 OPTIONS > User-Agent: Asterisk PBX > Max-Forwards: 70 > Date: Wed, 14 Mar 2007 07:45:26 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Content-Length: 0 > > > --- > salxpbx1*CLI> > <-- SIP read from 172.28.20.4:2051: > SIP/2.0 200 OK > Via: SIP/2.0/UDP 172.28.2.30:5060;branch=z9hG4bK7e2b5210;rport=5060 > From: "asterisk" <sip:asterisk@xxx.xx>;tag=as68d9b69b > To: <sip:104@172.28.20.4:2051;line=7p0ffzkf> > Call-ID: 30e124937558a11d3e9f397947b378b8@xxx.xx > CSeq: 102 OPTIONS > Contact: <sip:104@172.28.20.4:2051;line=7p0ffzkf>;flow-id=1 > User-Agent: snom300/6.2.3 > Accept-Language: en > Accept: application/sdp > Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK,MESSAGE, INFO> Allow-Events: talk, hold, refer > Supported: timer, 100rel, replaces, callerid > Content-Length: 0 > > > --- (14 headers 0 lines) --- > Destroying call '30e124937558a11d3e9f397947b378b8@xxx.xx' > Mar 14 08:45:27 WARNING[30110]: codec_ilbc.c:175 ilbctolin_framein: Huh?An ilbc frame that isn't a multiple of 50 bytes long from RTP (4)?> Mar 14 08:45:27 WARNING[30110]: codec_ilbc.c:175 ilbctolin_framein: Huh?An ilbc frame that isn't a multiple of 50 bytes long from RTP (4)?> Mar 14 08:45:27 WARNING[30110]: codec_ilbc.c:175 ilbctolin_framein: Huh?An ilbc frame that isn't a multiple of 50 bytes long from RTP (4)?> Mar 14 08:45:27 WARNING[30110]: codec_ilbc.c:175 ilbctolin_framein: Huh?An ilbc frame that isn't a multiple of 50 bytes long from RTP (4)?> > <-- SIP read from 172.28.20.4:2051: > BYE sip:374@172.28.2.30 SIP/2.0 > Via: SIP/2.0/UDP 172.28.20.4:2051;branch=z9hG4bK-e9vrp3jilhg7;rport > From: "Test User3" <sip:104@172.28.2.30>;tag=48rd3g03e1 > To: <sip:374@172.28.2.30;user=phone>;tag=as31ca24ed > Call-ID: 3c30f7fb186a-n2ptt0oiuivb@snom300-0004132520F7 > CSeq: 3 BYE > Max-Forwards: 70 > Contact: <sip:104@172.28.20.4:2051;line=7p0ffzkf>;flow-id=1 > User-Agent: snom300/6.2.3 > RTP-RxStat:Total_Rx_Pkts=350,Rx_Pkts=350,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0> RTP-TxStat: Total_Tx_Pkts=355,Tx_Pkts=355,Remote_Tx_Pkts=0 > Content-Length: 0 > > > --- (12 headers 0 lines) --- > Sending to 172.28.20.4 : 2051 (NAT) > Transmitting (NAT) to 172.28.20.4:2051: > SIP/2.0 200 OK > Via: SIP/2.0/UDP172.28.20.4:2051;branch=z9hG4bK-e9vrp3jilhg7;received=172.28.20.4;rport=2051> From: "Test User3" <sip:104@172.28.2.30>;tag=48rd3g03e1 > To: <sip:374@172.28.2.30;user=phone>;tag=as31ca24ed > Call-ID: 3c30f7fb186a-n2ptt0oiuivb@snom300-0004132520F7 > CSeq: 3 BYE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Contact: <sip:374@172.28.2.30> > Content-Length: 0 > X-Asterisk-HangupCause: Normal Clearing > > > --- > == Spawn extension (intern, h, 109) exited non-zero on'SIP/104-0819a1f8'> Destroying call '3c30f7fb186a-n2ptt0oiuivb@snom300-0004132520F7' > >---------------------------------------------------------------------------- ----> _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >