Thanks for the tip, I did just that, and now I am more confused. It does appear as though there is just one call ID (if my assumption that the "tag=" determines the call.
The first time it sends like this: <--- SIP read from UDP:x.x.x.x:5060 ---> INVITE sip:[email protected];user=phone SIP/2.0 Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bKe3e15c76913F8BDD From: "3271" <sip:3271@ y.y.y.y >;tag=990EE6B0-8E3DCEA7 To: <sip:3261@ y.y.y.y;user=phone> CSeq: 1 INVITE Call-ID: [email protected] Contact: <sip:3271@ x.x.x.x:5060> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundStationIP-SSIP_6000-UA/3.2.3.1734 Accept-Language: en Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 461 v=0 o=- 1271881915 1271881915 IN IP4 x.x.x.x s=Polycom IP Phone c=IN IP4 x.x.x.x t=0 0 a=sendrecv m=audio 2226 RTP/AVP 115 99 9 102 0 8 18 127 a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:99 SIREN14/16000 a=fmtp:99 bitrate=48000 a=rtpmap:9 G722/8000 a=rtpmap:102 G7221/16000 a=fmtp:102 bitrate=32000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:127 telephone-event/8000 Asterisk responds with a SIP/2.0 401 Unauthorized, the phone then comes back with this: <--- SIP read from UDP:x.x.x.x:5060 ---> INVITE sip:3261@ y.y.y.y;user=phone SIP/2.0 Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK6f7a6692AF94008 From: "3271" <sip:3271@ y.y.y.y >;tag=990EE6B0-8E3DCEA7 To: <sip:3261@ y.y.y.y;user=phone> CSeq: 2 INVITE Call-ID: 96a1fe9c-88f06c73-7e209322@ x.x.x.x Contact: <sip:3271@ x.x.x.x:5060> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundStationIP-SSIP_6000-UA/3.2.3.1734 Accept-Language: en Supported: 100rel,replaces Allow-Events: talk,hold,conference Authorization: Digest username="3271", realm="asterisk", nonce="393a1b1f", uri="sip:3261@ y.y.y.y;user=phone", response="c8223e261c252c12172982ee661ad307", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 461 v=0 o=- 1271881915 1271881915 IN IP4 x.x.x.x s=Polycom IP Phone c=IN IP4 x.x.x.x t=0 0 a=sendrecv m=audio 2226 RTP/AVP 115 99 9 102 0 8 18 127 a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:99 SIREN14/16000 a=fmtp:99 bitrate=48000 a=rtpmap:9 G722/8000 a=rtpmap:102 G7221/16000 a=fmtp:102 bitrate=32000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:127 telephone-event/8000 The difference is that the CSeq is now 2 and the following line is added: Authorization: Digest username="3271", realm="asterisk", nonce="393a1b1f", uri="sip:[email protected];user=phone", response="c8223e261c252c12172982ee661ad307", algorithm=MD5 So maybe I do have an issue in Asterisk, okay probably. Any clues as to how to debug? Let me know if need to post more information. Thanks. -Jay -----Original Message----- From: [email protected] [mailto:[email protected]] On Behalf Of Sean Brady Sent: Tuesday, April 20, 2010 4:57 AM To: [email protected] Subject: Re: [asterisk-users] Odd Issue With Polycom Phones On 04/19/2010 02:22 PM, Jay Vocaire wrote: > I have searched everywhere, but cannot seem to find anyone else talking about > this issue. Maybe I am just using the wrong search terms. > > I am running Asterisk 1.6.2 and multiple Polycom phones all with 3.2.3 (the > latest) firmware on them. > > I am having an issue with my 550's and my 6000's (but oddly enough, not my > 320's). Whenever a number is dialed on hook, and then the speakerphone > button is pressed, the number is dialed twice. If the handset is picked up, > or the "Dial" softkey is pressed, the call is only sent once. This leads me > to believe it is a phone issue, not a * config issue, but I have no way of > telling. > > I can verify that there are two call started via the snippet below: > <SNIP> > > The first hangup was triggered right away (without me doing anything), the > second hangup was me actually hanging up the calling phone. > > It does the same thing if I dial an outside line. > > Any idea where to start trying to solve this? Has anyone else seen it, and > can point me to the fix that I could not find with Google? > > Thanks. > > I would recommend that you enable debugging on the peer only and check to see if you see two invites come from the phone. Two invites with different call ID's would indicate it is indeed the phone making two calls. One would indicate that it MAY be an Asterisk issue. Are you using the latest Polycom firmware, btw? -- _____________________________________________________________________ -- 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 -- _____________________________________________________________________ -- 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
