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:[email protected] SIP/2.0 Via: SIP/2.0/TCP 192.168.1.15:52774 Max-Forwards: 70 From: <sip:[email protected]>;tag=39be813029;epid=f918608aea To: <sip:[email protected]> Call-ID: 738a7dd4d06d4c439c29fb703e491533 CSeq: 2 INVITE Contact: <sip:[email protected]: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:[email protected]", 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:[email protected]>;tag=39be813029;epid=f918608aea To: <sip:[email protected]>;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:[email protected] SIP/2.0 Via: SIP/2.0/TCP 192.168.1.15:52774 Max-Forwards: 70 From: <sip:[email protected]>;tag=39be813029;epid=f918608aea To: <sip:[email protected]>;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:[email protected]", 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
