Hello, I am developing the free SIP softphone (audio+video) for Windows. And I have some issues with asterisk 1.6 compatibility. I am new in asterisk, so I guess, I have no enough skills to config asterisk properly. I have enable tcp transport mode and register client, but can not make a call. The server report 491 Request Pending on invite message. Why server report the error? Here is link to the softphone: http://www.officesip.com/download/officesip-softphone-1.0.msi Best regards, Vitali Fomine INVITE sip:58 at trixbox1.local SIP/2.0 Via: SIP/2.0/TCP 192.168.1.15:52774 Max-Forwards: 70 From: <sip:56 at trixbox1.local>;tag=39be813029;epid=f918608aea To: <sip:58 at trixbox1.local> Call-ID: 738a7dd4d06d4c439c29fb703e491533 CSeq: 2 INVITE Contact: <sip:56 at trixbox1.local:52774;maddr=192.168.1.15;transport=tcp>;proxy=replace;+sip.instance="<urn:uuid:431F37E0-5CE9-5995-B8AB-3F65F0D9795A>" User-Agent: UCCAPI/2.0.6362.67 Supported: timer Supported: ms-sender Supported: ms-early-media Supported: Replaces ms-keep-alive: UAC;hop-hop=yes Authorization: Digest username="56", realm="asterisk", algorithm=MD5, uri="sip:58 at trixbox1.local", nonce="601d7934", response="36c795437dc4088ac5947f923e8dbb0f" Content-Type: application/sdp Content-Length: 2146 v=0 o=- 0 0 IN IP4 192.168.1.15 s=session c=IN IP4 192.168.1.15 b=CT:99980 t=0 0 m=audio 46080 RTP/AVP 114 111 112 115 116 4 8 0 97 101 k=base64:D3YQHD+33y6crQYg5HKB5+xk+uzWWjx1Nqu92I0yqiNXO3u4Neq5AsqMPOA0 a=candidate:y0T/m81+hR/z4InuWh6XB7ix0Az1iTjwcaqYENbLia4 1 JzFksprV3LGT4h5qLzK+gA UDP 0.830 192.168.1.15 46080 a=candidate:y0T/m81+hR/z4InuWh6XB7ix0Az1iTjwcaqYENbLia4 2 JzFksprV3LGT4h5qLzK+gA UDP 0.830 192.168.1.15 8960 a=candidate:JNZVPl8eF6E19v/HMq1oWE/6ScDVqz05i2XXLGhNhjw 1 qIgZJbPCW/xDOWMMN5ZWJQ UDP 0.840 192.168.56.1 17792 a=candidate:JNZVPl8eF6E19v/HMq1oWE/6ScDVqz05i2XXLGhNhjw 2 qIgZJbPCW/xDOWMMN5ZWJQ UDP 0.840 192.168.56.1 30080 a=cryptoscale:1 client AES_CM_128_HMAC_SHA1_80 inline:5zuBoVK+RVi6Yw/Po02VsVrbZVQLPVy4VxColZpZ|2^31|1:1 a=crypto:2 AES_CM_128_HMAC_SHA1_80 inline:bB2j++RDWPo/sLbSBVijJy8lKyy/dd2bEKyxdC+k|2^31|1:1 a=maxptime:200 a=rtcp:8960 a=rtpmap:114 x-msrta/16000 a=fmtp:114 bitrate=29000 a=rtpmap:111 SIREN/16000 a=fmtp:111 bitrate=16000 a=rtpmap:112 G7221/16000 a=fmtp:112 bitrate=24000 a=rtpmap:115 x-msrta/8000 a=fmtp:115 bitrate=11800 a=rtpmap:116 AAL2-G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:97 RED/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=encryption:optional m=video 34432 RTP/AVP 121 34 k=base64:ve6wgVJQaeIkcDokUVyKXuQM2JzIBIoyiJUDPcH27R89T80GhLRVF+JPZPtI a=candidate:VNyfYW0mHvCAJsHrBO+jt+l0bZHFIuiSfHfcjYORL8U 1 Xy99337KGr+R6TEcalNuaQ UDP 0.850 192.168.1.15 34432 a=candidate:VNyfYW0mHvCAJsHrBO+jt+l0bZHFIuiSfHfcjYORL8U 2 Xy99337KGr+R6TEcalNuaQ UDP 0.850 192.168.1.15 12032 a=candidate:h4gtcS9V5HP54MByMbgC3wJpFfg1WciRe/VqLZ6qiFc 1 gjQT1OVzQRIyzR2eq1i7kw UDP 0.860 192.168.56.1 52608 a=candidate:h4gtcS9V5HP54MByMbgC3wJpFfg1WciRe/VqLZ6qiFc 2 gjQT1OVzQRIyzR2eq1i7kw UDP 0.860 192.168.56.1 37120 a=cryptoscale:1 client AES_CM_128_HMAC_SHA1_80 inline:fcLhKq245/mep3k6sYBdnnusNq8mfwAN6aXBpbot|2^31|1:1 a=crypto:2 AES_CM_128_HMAC_SHA1_80 inline:q2Atyjw2+REUuFXxkQYrE3nuzsVT5xFkt+2xcaDD|2^31|1:1 a=maxptime:200 a=rtcp:12032 a=rtpmap:121 x-rtvc1/90000 a=rtpmap:34 H263/90000 a=encryption:optional SIP/2.0 491 Request Pending Via: SIP/2.0/TCP 192.168.1.15:52774;received=192.168.1.15 From: <sip:56 at trixbox1.local>;tag=39be813029;epid=f918608aea To: <sip:58 at trixbox1.local>;tag=as5c7a7ed8 Call-ID: 738a7dd4d06d4c439c29fb703e491533 CSeq: 2 INVITE User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 ACK sip:58 at trixbox1.local SIP/2.0 Via: SIP/2.0/TCP 192.168.1.15:52774 Max-Forwards: 70 From: <sip:56 at trixbox1.local>;tag=39be813029;epid=f918608aea To: <sip:58 at trixbox1.local>;tag=as5c7a7ed8 Call-ID: 738a7dd4d06d4c439c29fb703e491533 CSeq: 2 ACK User-Agent: UCCAPI/2.0.6362.67 Authorization: Digest username="56", realm="asterisk", algorithm=MD5, uri="sip:58 at trixbox1.local", nonce="601d7934", response="35dca1911b5bb614b1cadfda53e7d8f4" Content-Length: 0
22 jan 2010 kl. 10.13 skrev Vitali Fomine:> Hello, > > I am developing the free SIP softphone (audio+video) for Windows. And I have > some issues with asterisk 1.6 compatibility. I am new in asterisk, so I > guess, I have no enough skills to config asterisk properly. I have enable > tcp transport mode and register client, but can not make a call. The server > report 491 Request Pending on invite message. Why server report the error?The server reports this when we already have an INVITE to handle. Please check that you did not transmit two invites without waiting for a response and sending an ACK from your softphone. /O> > Here is link to the softphone: > http://www.officesip.com/download/officesip-softphone-1.0.msi > > Best regards, > Vitali Fomine > > > INVITE sip:58 at trixbox1.local SIP/2.0 > Via: SIP/2.0/TCP 192.168.1.15:52774 > Max-Forwards: 70 > From: <sip:56 at trixbox1.local>;tag=39be813029;epid=f918608aea > To: <sip:58 at trixbox1.local> > Call-ID: 738a7dd4d06d4c439c29fb703e491533 > CSeq: 2 INVITE > Contact: > <sip:56 at trixbox1.local:52774;maddr=192.168.1.15;transport=tcp>;proxy=replace;+sip.instance="<urn:uuid:431F37E0-5CE9-5995-B8AB-3F65F0D9795A>" > User-Agent: UCCAPI/2.0.6362.67 > Supported: timer > Supported: ms-sender > Supported: ms-early-media > Supported: Replaces > ms-keep-alive: UAC;hop-hop=yes > Authorization: Digest username="56", realm="asterisk", algorithm=MD5, > uri="sip:58 at trixbox1.local", nonce="601d7934", > response="36c795437dc4088ac5947f923e8dbb0f" > Content-Type: application/sdp > Content-Length: 2146 > > v=0 > o=- 0 0 IN IP4 192.168.1.15 > s=session > c=IN IP4 192.168.1.15 > b=CT:99980 > t=0 0 > m=audio 46080 RTP/AVP 114 111 112 115 116 4 8 0 97 101 > k=base64:D3YQHD+33y6crQYg5HKB5+xk+uzWWjx1Nqu92I0yqiNXO3u4Neq5AsqMPOA0 > a=candidate:y0T/m81+hR/z4InuWh6XB7ix0Az1iTjwcaqYENbLia4 1 > JzFksprV3LGT4h5qLzK+gA UDP 0.830 192.168.1.15 46080 > a=candidate:y0T/m81+hR/z4InuWh6XB7ix0Az1iTjwcaqYENbLia4 2 > JzFksprV3LGT4h5qLzK+gA UDP 0.830 192.168.1.15 8960 > a=candidate:JNZVPl8eF6E19v/HMq1oWE/6ScDVqz05i2XXLGhNhjw 1 > qIgZJbPCW/xDOWMMN5ZWJQ UDP 0.840 192.168.56.1 17792 > a=candidate:JNZVPl8eF6E19v/HMq1oWE/6ScDVqz05i2XXLGhNhjw 2 > qIgZJbPCW/xDOWMMN5ZWJQ UDP 0.840 192.168.56.1 30080 > a=cryptoscale:1 client AES_CM_128_HMAC_SHA1_80 > inline:5zuBoVK+RVi6Yw/Po02VsVrbZVQLPVy4VxColZpZ|2^31|1:1 > a=crypto:2 AES_CM_128_HMAC_SHA1_80 > inline:bB2j++RDWPo/sLbSBVijJy8lKyy/dd2bEKyxdC+k|2^31|1:1 > a=maxptime:200 > a=rtcp:8960 > a=rtpmap:114 x-msrta/16000 > a=fmtp:114 bitrate=29000 > a=rtpmap:111 SIREN/16000 > a=fmtp:111 bitrate=16000 > a=rtpmap:112 G7221/16000 > a=fmtp:112 bitrate=24000 > a=rtpmap:115 x-msrta/8000 > a=fmtp:115 bitrate=11800 > a=rtpmap:116 AAL2-G726-32/8000 > a=rtpmap:4 G723/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:97 RED/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=encryption:optional > m=video 34432 RTP/AVP 121 34 > k=base64:ve6wgVJQaeIkcDokUVyKXuQM2JzIBIoyiJUDPcH27R89T80GhLRVF+JPZPtI > a=candidate:VNyfYW0mHvCAJsHrBO+jt+l0bZHFIuiSfHfcjYORL8U 1 > Xy99337KGr+R6TEcalNuaQ UDP 0.850 192.168.1.15 34432 > a=candidate:VNyfYW0mHvCAJsHrBO+jt+l0bZHFIuiSfHfcjYORL8U 2 > Xy99337KGr+R6TEcalNuaQ UDP 0.850 192.168.1.15 12032 > a=candidate:h4gtcS9V5HP54MByMbgC3wJpFfg1WciRe/VqLZ6qiFc 1 > gjQT1OVzQRIyzR2eq1i7kw UDP 0.860 192.168.56.1 52608 > a=candidate:h4gtcS9V5HP54MByMbgC3wJpFfg1WciRe/VqLZ6qiFc 2 > gjQT1OVzQRIyzR2eq1i7kw UDP 0.860 192.168.56.1 37120 > a=cryptoscale:1 client AES_CM_128_HMAC_SHA1_80 > inline:fcLhKq245/mep3k6sYBdnnusNq8mfwAN6aXBpbot|2^31|1:1 > a=crypto:2 AES_CM_128_HMAC_SHA1_80 > inline:q2Atyjw2+REUuFXxkQYrE3nuzsVT5xFkt+2xcaDD|2^31|1:1 > a=maxptime:200 > a=rtcp:12032 > a=rtpmap:121 x-rtvc1/90000 > a=rtpmap:34 H263/90000 > a=encryption:optional > > > SIP/2.0 491 Request Pending > Via: SIP/2.0/TCP 192.168.1.15:52774;received=192.168.1.15 > From: <sip:56 at trixbox1.local>;tag=39be813029;epid=f918608aea > To: <sip:58 at trixbox1.local>;tag=as5c7a7ed8 > Call-ID: 738a7dd4d06d4c439c29fb703e491533 > CSeq: 2 INVITE > User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40 > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces, timer > Content-Length: 0 > > > ACK sip:58 at trixbox1.local SIP/2.0 > Via: SIP/2.0/TCP 192.168.1.15:52774 > Max-Forwards: 70 > From: <sip:56 at trixbox1.local>;tag=39be813029;epid=f918608aea > To: <sip:58 at trixbox1.local>;tag=as5c7a7ed8 > Call-ID: 738a7dd4d06d4c439c29fb703e491533 > CSeq: 2 ACK > User-Agent: UCCAPI/2.0.6362.67 > Authorization: Digest username="56", realm="asterisk", algorithm=MD5, > uri="sip:58 at trixbox1.local", nonce="601d7934", > response="35dca1911b5bb614b1cadfda53e7d8f4" > Content-Length: 0 > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users--- * Olle E Johansson - oej at edvina.net * Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden
>> report 491 Request Pending on invite message. Why server report the >> error? > The server reports this when we already have an INVITE to handle. > Please check that you did not transmit two invites without waiting for > a response and sending an ACK from your softphone.Yes, here is two INVITEs (I have missed first invite before), but the server respond 401 on first invite and softphone send ACK. Here is softphone log. Unfortunately, I do not know how to enable (where to find) log of SIP messages at server? I have find one more issue, the server sends two replies on register message, first with 200 and second one 403 with same CSeq. I am not sure is that relayted to INVITE issue. But the asterisk show user as connected-unmonitored in control panel (trixbox). Best regards, Vitali Fomine 01/22/2010|13:37:36.097 INVITE sip:58 at trixbox1.local SIP/2.0 Via: SIP/2.0/TCP 192.168.1.15:58238 Max-Forwards: 70 From: <sip:56 at trixbox1.local>;tag=60b512cec9;epid=08fd7dc31f To: <sip:58 at trixbox1.local> Call-ID: 16a3a30998874ae98538d221a2567fe1 CSeq: 1 INVITE Contact: <sip:56 at trixbox1.local:58238;maddr=192.168.1.15;transport=tcp>;proxy=replace;+sip.instance="<urn:uuid:D37BDEE9-53F8-5FD6-8CE0-51307C216D28>" User-Agent: UCCAPI/2.0.6362.67 Supported: timer Supported: ms-sender Supported: ms-early-media Supported: Replaces ms-keep-alive: UAC;hop-hop=yes Authorization: Digest username="56", realm="asterisk", algorithm=MD5, uri="sip:58 at trixbox1.local", nonce="1746fc14", response="c57700c482c83cbeb411398d92f94113" Content-Type: application/sdp Content-Length: 2147 v=0 o=- 0 0 IN IP4 192.168.1.15 s=session [...session description removed...] 01/22/2010|13:37:36.102 SIP/2.0 401 Unauthorized Via: SIP/2.0/TCP 192.168.1.15:58238;received=192.168.1.15 From: <sip:56 at trixbox1.local>;tag=60b512cec9;epid=08fd7dc31f To: <sip:58 at trixbox1.local>;tag=as63c5f412 Call-ID: 16a3a30998874ae98538d221a2567fe1 CSeq: 1 INVITE User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="39ce8842" Content-Length: 0 01/22/2010|13:37:36.102 ACK sip:58 at trixbox1.local SIP/2.0 Via: SIP/2.0/TCP 192.168.1.15:58238 Max-Forwards: 70 From: <sip:56 at trixbox1.local>;tag=60b512cec9;epid=08fd7dc31f To: <sip:58 at trixbox1.local>;tag=as63c5f412 Call-ID: 16a3a30998874ae98538d221a2567fe1 CSeq: 1 ACK User-Agent: UCCAPI/2.0.6362.67 Authorization: Digest username="56", realm="asterisk", algorithm=MD5, uri="sip:58 at trixbox1.local", nonce="1746fc14", response="a3e2da49ec1a432115871eb965f4aad3" Content-Length: 0 01/22/2010|13:37:36.103 INVITE sip:58 at trixbox1.local SIP/2.0 Via: SIP/2.0/TCP 192.168.1.15:58238 Max-Forwards: 70 From: <sip:56 at trixbox1.local>;tag=60b512cec9;epid=08fd7dc31f To: <sip:58 at trixbox1.local> Call-ID: 16a3a30998874ae98538d221a2567fe1 CSeq: 2 INVITE Contact: <sip:56 at trixbox1.local:58238;maddr=192.168.1.15;transport=tcp>;proxy=replace;+sip.instance="<urn:uuid:D37BDEE9-53F8-5FD6-8CE0-51307C216D28>" User-Agent: UCCAPI/2.0.6362.67 Supported: timer Supported: ms-sender Supported: ms-early-media Supported: Replaces ms-keep-alive: UAC;hop-hop=yes Authorization: Digest username="56", realm="asterisk", algorithm=MD5, uri="sip:58 at trixbox1.local", nonce="39ce8842", response="5925e73eeaf067412c6b1c73cf520d0e" Content-Type: application/sdp Content-Length: 2147 v=0 o=- 0 0 IN IP4 192.168.1.15 s=session c=IN IP4 192.168.1.15 [...removed..] 01/22/2010|13:37:36.106 SIP/2.0 491 Request Pending Via: SIP/2.0/TCP 192.168.1.15:58238;received=192.168.1.15 From: <sip:56 at trixbox1.local>;tag=60b512cec9;epid=08fd7dc31f To: <sip:58 at trixbox1.local>;tag=as63c5f412 Call-ID: 16a3a30998874ae98538d221a2567fe1 CSeq: 2 INVITE User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 01/22/2010|13:37:36.106 ACK sip:58 at trixbox1.local SIP/2.0 Via: SIP/2.0/TCP 192.168.1.15:58238 Max-Forwards: 70 From: <sip:56 at trixbox1.local>;tag=60b512cec9;epid=08fd7dc31f To: <sip:58 at trixbox1.local>;tag=as63c5f412 Call-ID: 16a3a30998874ae98538d221a2567fe1 CSeq: 2 ACK User-Agent: UCCAPI/2.0.6362.67 Authorization: Digest username="56", realm="asterisk", algorithm=MD5, uri="sip:58 at trixbox1.local", nonce="39ce8842", response="2ebb3595b1af94e67f7e880478c82171" Content-Length: 0
22 jan 2010 kl. 11.51 skrev Vitali Fomine:>>> report 491 Request Pending on invite message. Why server report the >>> error? >> The server reports this when we already have an INVITE to handle. >> Please check that you did not transmit two invites without waiting for >> a response and sending an ACK from your softphone. > > Yes, here is two INVITEs (I have missed first invite before), but the server > respond 401 on first invite and softphone send ACK. Here is softphone log. > Unfortunately, I do not know how to enable (where to find) log of SIP > messages at server?I would like to see that log with Asterisk messages in between, so I understand when Asterisk receives the ACK. If Asterisk receives the ACK *after* the second INVITE I understand it.> > I have find one more issue, the server sends two replies on register > message, first with 200 and second one 403 with same CSeq. I am not sure is > that relayted to INVITE issue. But the asterisk show user as > connected-unmonitored in control panel (trixbox).I would like to see this as well, from an Asterisk CLI log perspective with "sip debug" turned on. /O> > Best regards, > Vitali Fomine > > 01/22/2010|13:37:36.097 > INVITE sip:58 at trixbox1.local SIP/2.0 > Via: SIP/2.0/TCP 192.168.1.15:58238 > Max-Forwards: 70 > From: <sip:56 at trixbox1.local>;tag=60b512cec9;epid=08fd7dc31f > To: <sip:58 at trixbox1.local> > Call-ID: 16a3a30998874ae98538d221a2567fe1 > CSeq: 1 INVITE > Contact: > <sip:56 at trixbox1.local:58238;maddr=192.168.1.15;transport=tcp>;proxy=replace;+sip.instance="<urn:uuid:D37BDEE9-53F8-5FD6-8CE0-51307C216D28>" > User-Agent: UCCAPI/2.0.6362.67 > Supported: timer > Supported: ms-sender > Supported: ms-early-media > Supported: Replaces > ms-keep-alive: UAC;hop-hop=yes > Authorization: Digest username="56", realm="asterisk", algorithm=MD5, > uri="sip:58 at trixbox1.local", nonce="1746fc14", > response="c57700c482c83cbeb411398d92f94113" > Content-Type: application/sdp > Content-Length: 2147 > > v=0 > o=- 0 0 IN IP4 192.168.1.15 > s=session > [...session description removed...] > > > 01/22/2010|13:37:36.102 > SIP/2.0 401 Unauthorized > Via: SIP/2.0/TCP 192.168.1.15:58238;received=192.168.1.15 > From: <sip:56 at trixbox1.local>;tag=60b512cec9;epid=08fd7dc31f > To: <sip:58 at trixbox1.local>;tag=as63c5f412 > Call-ID: 16a3a30998874ae98538d221a2567fe1 > CSeq: 1 INVITE > User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40 > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces, timer > WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="39ce8842" > Content-Length: 0 > > > > 01/22/2010|13:37:36.102 > ACK sip:58 at trixbox1.local SIP/2.0 > Via: SIP/2.0/TCP 192.168.1.15:58238 > Max-Forwards: 70 > From: <sip:56 at trixbox1.local>;tag=60b512cec9;epid=08fd7dc31f > To: <sip:58 at trixbox1.local>;tag=as63c5f412 > Call-ID: 16a3a30998874ae98538d221a2567fe1 > CSeq: 1 ACK > User-Agent: UCCAPI/2.0.6362.67 > Authorization: Digest username="56", realm="asterisk", algorithm=MD5, > uri="sip:58 at trixbox1.local", nonce="1746fc14", > response="a3e2da49ec1a432115871eb965f4aad3" > Content-Length: 0 > > > > 01/22/2010|13:37:36.103 > INVITE sip:58 at trixbox1.local SIP/2.0 > Via: SIP/2.0/TCP 192.168.1.15:58238 > Max-Forwards: 70 > From: <sip:56 at trixbox1.local>;tag=60b512cec9;epid=08fd7dc31f > To: <sip:58 at trixbox1.local> > Call-ID: 16a3a30998874ae98538d221a2567fe1 > CSeq: 2 INVITE > Contact: > <sip:56 at trixbox1.local:58238;maddr=192.168.1.15;transport=tcp>;proxy=replace;+sip.instance="<urn:uuid:D37BDEE9-53F8-5FD6-8CE0-51307C216D28>" > User-Agent: UCCAPI/2.0.6362.67 > Supported: timer > Supported: ms-sender > Supported: ms-early-media > Supported: Replaces > ms-keep-alive: UAC;hop-hop=yes > Authorization: Digest username="56", realm="asterisk", algorithm=MD5, > uri="sip:58 at trixbox1.local", nonce="39ce8842", > response="5925e73eeaf067412c6b1c73cf520d0e" > Content-Type: application/sdp > Content-Length: 2147 > > v=0 > o=- 0 0 IN IP4 192.168.1.15 > s=session > c=IN IP4 192.168.1.15 > [...removed..] > > > 01/22/2010|13:37:36.106 > SIP/2.0 491 Request Pending > Via: SIP/2.0/TCP 192.168.1.15:58238;received=192.168.1.15 > From: <sip:56 at trixbox1.local>;tag=60b512cec9;epid=08fd7dc31f > To: <sip:58 at trixbox1.local>;tag=as63c5f412 > Call-ID: 16a3a30998874ae98538d221a2567fe1 > CSeq: 2 INVITE > User-Agent: Asterisk PBX 1.6.0.10-FONCORE-r40 > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces, timer > Content-Length: 0 > > > 01/22/2010|13:37:36.106 > ACK sip:58 at trixbox1.local SIP/2.0 > Via: SIP/2.0/TCP 192.168.1.15:58238 > Max-Forwards: 70 > From: <sip:56 at trixbox1.local>;tag=60b512cec9;epid=08fd7dc31f > To: <sip:58 at trixbox1.local>;tag=as63c5f412 > Call-ID: 16a3a30998874ae98538d221a2567fe1 > CSeq: 2 ACK > User-Agent: UCCAPI/2.0.6362.67 > Authorization: Digest username="56", realm="asterisk", algorithm=MD5, > uri="sip:58 at trixbox1.local", nonce="39ce8842", > response="2ebb3595b1af94e67f7e880478c82171" > Content-Length: 0 > > > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users--- * Olle E Johansson - oej at edvina.net * Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden
Hello,> I would like to see this as well, from an Asterisk CLI log perspective > with "sip debug" turned on.The .log file for login and invite is attached, I have use asterisk -vr command. Is it correct?>> Yes, here is two INVITEs (I have missed first invite before), but the >> server >> respond 401 on first invite and softphone send ACK. Here is softphone >> log. > If Asterisk receives the ACK *after* the second INVITE I understand it.The softphone uses single tcp connection, so messages must arrive in same order as them was sent. Best regards, Vitali Fomine -------------- next part -------------- A non-text attachment was scrubbed... Name: login-invite.log Type: application/octet-stream Size: 20170 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20100122/69186dbc/attachment.obj