Re: [asterisk-users] Press release: Virtual Communication Clouds :: New feature in Asterisk 1.8
> Thought it was funny myself! Not as funny as the other one just posted > about the VoIP clients written in C# and .net under Microsoft though. Now > that was funny! It was not a joke. What is wrong with c#,.net? Best regards, Vitali Fomine -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OfficeSIP Communications Makes Its VoIP SIP Products Open Source
Press Release For Immediate Release OfficeSIP Communications http://www.officesip.com/ i...@officesip.com OfficeSIP Communications Makes Its VoIP SIP Products Open Source OfficeSIP Communications makes its two enterprise VoIP SIP clients officially open-source. OfficeSIP Softphone and OfficeSIP Messenger are now publicly available, and their source code published under the GPL license. The two products complete with the source code are available for immediate download at the company's Web site, officesip.org. OfficeSIP Communications is committed to continuous development of both SIP clients, and invites developers to join the project. The company believes that opening the source code to the community will benefit the development of the project, and will help it gain trust and popularity among its users. About OfficeSIP Softphone and OfficeSIP Messenger The two VoIP applications enable users to communicate via the industry-standard SIP protocol. OfficeSIP Softphone is a simple, straightforward SIP client enabling voice and video communications, while OfficeSIP Messenger offers enterprise customers the ability to communicate via text, voice and video chats for free. Compatible with Office Communications Server 2007, OfficeSIP Messenger delivers reliable performance combined with trouble-free deployment and management. OfficeSIP Messenger implements ICE, STUN, and TURN protocols to seamlessly traverse NAT and firewalls, and supports secure communications via the TLS protocol. OfficeSIP Softphone and OfficeSIP Messenger are written in C# in .NET framework. The two applications make use of Microsoft Unified Communications Client API SDK, ensuring the highest quality of audio and video communications. The use of underlying Microsoft platform ensures the greatest level of compatibility with a wide range of hardware devices such as webcams. OfficeSIP Softphone and OfficeSIP Messenger have been extensively tested, and offer the complete SIP functionality. About OfficeSIP Communications Established in 2007, OfficeSIP Communications has been developing open-source instant messaging and VoIP solutions for enterprises. The company established solid reputation among its customers, and gained expertise in meeting the communication needs of its corporate customers. # # # OfficeSIP Softphone and OfficeSIP Messenger along with their source code are available under the GPL license at http://www.officesip.org/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fw: OfficeSIP Softphone
Hello, Thank you for your help. I have enable tcp by using tcpenable and tcpbindaddr. The client can not connect w/o these settings. I am trying asterisk 1.6.0.10 (in trixbox), need I install something else? Best regards, Vitali Fomine - Original Message - From: Adrià Vidal To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, January 28, 2010 4:20 PM Subject: Re: [asterisk-users] Fw: OfficeSIP Softphone On Thu, Jan 28, 2010 at 1:56 PM, Vitali Fomine wrote: Hello, Yes, unfortunately, the sip client lib does not support udp. Best regards, Vitali Fomine Then check you are using an Asterisk patched for TCP. -- -- Adrià Vidal -- -- _ -- 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-- _ -- 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
Re: [asterisk-users] Fw: OfficeSIP Softphone
Hello, Yes, unfortunately, the sip client lib does not support udp. Best regards, Vitali Fomine - Original Message - From: Adrià Vidal To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, January 28, 2010 3:49 PM Subject: Re: [asterisk-users] Fw: OfficeSIP Softphone You are running an Asterisk version for SIP TCP ? your SIP UA seems talking SIP over TCP Via: SIP/2.0/TCP 192.168.1.15:56298 Max-Forwards: 70 From: ;tag=2baacde98c;epid=aa3c1b27a7 To: Call-ID: 28a90e7402da49159f343be9bc82b4d0 CSeq: 1 SERVICE Contact: ;proxy=replace;+sip.instance="" User-Agent: UCCAPI/2.0.6362.67 Authorization: Digest username="56", realm="asterisk", algorithm=MD5, uri="sip:mrasloc.trixbox1.lo...@trixbox1.local", nonce="36662fdf", response="d6f90f263010891a42b3f7d46113796a" Content-Type: application/msrtc-media-relay-auth+xml Content-Length: 395 -- -- Adrià Vidal -- -- _ -- 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-- _ -- 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
[asterisk-users] Fw: OfficeSIP Softphone
Hello, Could anyone help to review the log and issue? Where I could post asterisk bugreport? I could help with testing if someone try to fix this error. Best regards, Vitali Fomine - Original Message - From: "Vitali Fomine" To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Friday, January 22, 2010 3:33 PM Subject: Re: [asterisk-users] OfficeSIP Softphone 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 -- _ -- 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 login-invite.log Description: Binary data -- _ -- 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
Re: [asterisk-users] OfficeSIP Softphone
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 login-invite.log Description: Binary data -- _ -- 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
Re: [asterisk-users] OfficeSIP Softphone
>> 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:5...@trixbox1.local SIP/2.0 Via: SIP/2.0/TCP 192.168.1.15:58238 Max-Forwards: 70 From: ;tag=60b512cec9;epid=08fd7dc31f To: Call-ID: 16a3a30998874ae98538d221a2567fe1 CSeq: 1 INVITE Contact: ;proxy=replace;+sip.instance="" 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:5...@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: ;tag=60b512cec9;epid=08fd7dc31f To: ;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:5...@trixbox1.local SIP/2.0 Via: SIP/2.0/TCP 192.168.1.15:58238 Max-Forwards: 70 From: ;tag=60b512cec9;epid=08fd7dc31f To: ;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:5...@trixbox1.local", nonce="1746fc14", response="a3e2da49ec1a432115871eb965f4aad3" Content-Length: 0 01/22/2010|13:37:36.103 INVITE sip:5...@trixbox1.local SIP/2.0 Via: SIP/2.0/TCP 192.168.1.15:58238 Max-Forwards: 70 From: ;tag=60b512cec9;epid=08fd7dc31f To: Call-ID: 16a3a30998874ae98538d221a2567fe1 CSeq: 2 INVITE Contact: ;proxy=replace;+sip.instance="" 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:5...@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: ;tag=60b512cec9;epid=08fd7dc31f To: ;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:5...@trixbox1.local SIP/2.0 Via: SIP/2.0/TCP 192.168.1.15:58238 Max-Forwards: 70 From: ;tag=60b512cec9;epid=08fd7dc31f To: ;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:5...@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
[asterisk-users] OfficeSIP Softphone
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:5...@trixbox1.local SIP/2.0 Via: SIP/2.0/TCP 192.168.1.15:52774 Max-Forwards: 70 From: ;tag=39be813029;epid=f918608aea To: Call-ID: 738a7dd4d06d4c439c29fb703e491533 CSeq: 2 INVITE Contact: ;proxy=replace;+sip.instance="" 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:5...@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-rtvc1/9 a=rtpmap:34 H263/9 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: ;tag=39be813029;epid=f918608aea To: ;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:5...@trixbox1.local SIP/2.0 Via: SIP/2.0/TCP 192.168.1.15:52774 Max-Forwards: 70 From: ;tag=39be813029;epid=f918608aea To: ;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:5...@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
