search for: polycomsoundpointip

...192.168.1.30>;tag=53ED9FBF-D06765E2 To: <sip:2000@192.168.1.30;user=phone> CSeq: 1 INVITE Call-ID: a9092ab-b63e7115-89ce2c58@192.168.1.37 Contact: <sip:2004@192.168.1.37:5060> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.2.0054 Supported: 100rel,replace Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 225 v=0 o=- 1124335166 1124335166 IN IP4 192.168.1.37 s=Polycom IP Phone c=IN IP4 192.168.1.37 t=0 0 a=sendrecv m=audio 2224 RTP/AVP 18 0 101 a=rtpmap:...

..."1050" <sip:1050 at XXX.XXX.232.175>;tag=as140f4415 CSeq: 1 ACK Call-ID: 75235b51626f63440c264f6b70dc5718 at XXX.XXX.232.175 Contact: <sip:1051 at XXX.XXX.232.66:8986> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_331-UA/3.2.2.0477 Accept-Language: en Max-Forwards: 70 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- <--- SIP read from XXX.XXX.232.66:8986 ---> REFER sip:1050 at XXX.XXX.232.175 SIP/2.0 Via: SIP/2.0/UDP XXX.XXX.232.66:8986;branch=z9hG4bKd2064841D6E30B4A From: &q...

...168.1.150>;tag=87113650-18E1B969 To: <sip:10 at 192.168.1.150;user=phone> CSeq: 1 INVITE Call-ID: e6055f35-926fca76-78e0dcbf at 192.168.1.99 Contact: <sip:522 at 192.168.1.99> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_330-UA/2.1.2.0049 Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 249 v=0 o=- 1207070053 1207070053 IN IP4 192.168.1.99 s=Polycom IP Phone c=IN IP4 192.168.1.99 t=0 0 m=audio 2228 RTP/AVP 0 8 18 101 a=sendrecv a=rtpm...

...t : Addr->IP : 74.CENSORED.213 Port 5060 Defaddr->IP : 0.0.0.0 Port 5060 Reg. exten : Def. Username: 17865221569 SIP Options : (none) Codecs : 0x104 (ulaw|g729) Codec Order : (g729:20,ulaw:20) Auto-Framing: No Status : OK (130 ms) Useragent : PolycomSoundPointIP-SPIP_501-UA/3.0.1.0032 Reg. Contact : sip:17865221569 at 192.168.1.54 app5*CLI> core show version Asterisk 1.4.21.1 built by root @ app5 on a i686 running Linux on 2008-07-09 01:41:43 UTC

...192.168.1.8>;tag=25AB8538-7BACFE71 To: <sip:10 at 192.168.1.8;user=phone> CSeq: 1 INVITE Call-ID: a4b0cbb4-9737882e-815856ff at 192.168.1.89 Contact: <sip:404 at 192.168.1.89> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.0.0.0258 Supported: 1?00rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 249 v=0 o=- 1226069152 1226069152 IN IP4 192.168.1.89 s=Polycom IP Phone c=IN IP4 192.168.1.89 t=0 0 m=audio 2244 RTP/AVP 0 8 18 101 a=sendrecv a=rtp...

...gt;;tag=D4964260-95FB99E3 To: <sip:9990@hostname.company.domain;user=phone> CSeq: 1 INVITE Call-ID: a857d7ac-36f29d46-4d6ef889@10.253.4.50 Contact: <sip:eden-1000a@10.253.4.50> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.0.3.0127 Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 245 v=0 o=- 978307756 978307756 IN IP4 10.253.4.50 s=Polycom IP Phone c=IN IP4 10.253.4.50 t=0 0 m=audio 2228 RTP/AVP 0 18 8 101 a=sendrecv a=rtpmap:0...

...168.4.5:5060;branch=z9hG4bK15b008bf From: "3124048745" <sip:3124048745@192.168.4.5>;tag=as5e966399 To: <sip:3004@192.168.4.204>;tag=EAA91427-3070A3C8 CSeq: 102 INVITE Call-ID: 6f12df2509a5292d6775e2143f75f93a@192.168.4.5 Contact: <sip:3004@192.168.4.204:5060> User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.4 Content-Length: 0 9 headers, 0 lines Sip read: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.4.5:5060;branch=z9hG4bK15b008bf From: "3124048745" <sip:3124048745@192.168.4.5>;tag=as5e966399 To: <sip:3004@192.168.4.204>;tag=EAA91427-3070A3C8 CSeq: 102 INVITE...

...=2A2425B5-B64A4132. To: <sip:2944029@ipt.oneeighty.com>. CSeq: 1 REGISTER. Call-ID: 56150889-214b0f7f-e02e6d9c@216.187.128.72. Contact: <sip:2944029@216.187.128.72>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER". User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.3.0067. Max-Forwards: 70. Expires: 3600. Content-Length: 0. . # U 216.187.140.233:5060 -> 216.187.128.72:5060 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 216.187.128.72;branch=z9hG4bKca28b8d3BC755D46. From: "Sandy Sauvageau" <sip:2944029@ipt.oneeighty.com>;tag=2A2...

Got my first round of IP500s in today. Anybody have any example sip.cfg files they'd like to share? Tim Jackson Network Engineer Angelina County, Texas (936)639-4827 office (936)414-6723 mobile -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040928/a923e094/attachment.htm

...a: SIP/2.0/UDP 192.168.1.15:5060;branch=z9hG4bK0f90d445 From: "Frank Rizzo" <sip:5285@192.168.1.15>;tag=as6a2a2007 To: <sip:192.168.1.181>;tag=C4BAB225-8696114 CSeq: 102 INVITE Call-ID: 08cf93d712ecba2703837fed6f933068@192.168.1.15 Contact:<sip:192.168.1.181> User-Agent: PolycomSoundPointIP-UA/1.0.4 Content-Length: 0 9 headers, 0 lines Sip read: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.1.15:5060;branch=z9hG4bK0f90d445 From: "Frank Rizzo" <sip:5285@192.168.1.15>;tag=as6a2a2007 To: <sip:192.168.1.181>;tag=C4BAB225-8696114 CSeq: 102 INVITE Call-ID: 08cf93d712e...

Hi, I configured Buddy Watch function on my Polycom IP 601. It works well, until I make a reload of Asterisk. After reload, It can't monitor any lines and I have to restart the phone to reactivate this function. Is this a specific problem of asterisk-1.2.3? How can I solve it? Thank in advance, regards, Marco.

....edu>;tag=DDF0722-FFF8D457 To: <sip:4000@tcm1.shsu.edu;user=phone> CSeq: 1 INVITE Call-ID: dd12cb03-99065278-efdfa12d@10.20.136.130 Contact: <sip:0004F2100526_2@10.20.136.130> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_430-UA/2.0.1.0291 Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 251 v=0 o=- 1161637564 1161637564 IN IP4 10.20.136.130 s=Polycom IP Phone c=IN IP4 10.20.136.130 t=0 0 a=sendrecv m=audio 2240 RTP/AVP 0 8 18 101 a=rt...

...rport From: <sip:2944026@ua1.ipt.oneeighty.com>;tag=as6fd80d1b To: "Front Desk" <sip:2944030@ua1.ipt.oneeighty.com>;tag=3B576862-120A3007 CSeq: 114 NOTIFY Call-ID: 5d22385e-d097d068-93894e55@xxx.187.128.95 Contact: <sip:2944030@xxx.187.128.95> Event: presence User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.6.0036 Content-Length: 0 Doug.

Before I play around with this again in 1.2.1, regexten is still essentially broken, correct? The misconception seems to be that it allows you to execute a command upon registration from a SIP UA. Even the O'Reilly TFOT book erroneously states that this is what it is for. After reading the developer discussion though, it definitely seems to be broken. Is it fixed yet? Doug.

...Via: SIP/2.0/UDP 10.20.1.79:5060;branch=z9hG4bK271f9f8f From: "asterisk" <sip:asterisk@10.20.1.79>;tag=as319b6dd5 To: <sip:1630@10.20.2.16>;tag=A368FBB2-B6EAC0CF CSeq: 104 BYE Call-ID: 758a7b251c545f4652227c4d74d2d12d@10.20.1.79 Contact: <sip:1630@10.20.2.16> User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.2.0041 Content-Length: 0 Aside from the MoH and the error, all procedes as it should; eventually, the call is either answered or goes to voicemail. Any ideas as to what I'm doing wrong? Thanks, -Ken

Hello All, I am having an odd problem with Armin's chan-capi_cm on builds higher than 7955. It would seem that this happens on anything higher than 7955. What is happening is the isdn is ringing, then asterisk does a goto-if and just hangs. Asterisk itself is ok, but the isdn then rings out or busys out on the other side. Outgoing works fine, this only seems to effect incoming. I

...: <sip:5622832456@67.110.252.13:5060> CSeq: 1 REGISTER Call-ID: f25ece25-9e450ecf-437df5a2@67.110.253.129 Contact: <sip:5622832456@67.110.253.129:5060;transport=udp>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.0 Max-Forwards: 70 Expires: 300 Content-Length: 0 11 headers, 0 lines Using latest request as basis request Sending to 67.110.253.129 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 67.110.253.129:5060;branch=z9hG4bK63df903b2EF1BB58 From: "138po...

...9>;tag=569A308-31C12E4D To: <sip:groupcallpickup@192.168.0.9> CSeq: 1 SUBSCRIBE Call-ID: d4b32c74-68b2cfb6-70db113@192.168.0.234 Contact: <sip:201@192.168.0.234> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER Event: dialog User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.5.2.0054 Accept: application/dialog-info+xml Max-Forwards: 70 Expires: 0 Content-Length: 0 14 headers, 0 lines Using latest SUBSCRIBE request as basis request Sending to 192.168.0.234 : 5060 (non-NAT) Found peer '201' Transmitting (no NAT): SIP/2.0 407 Proxy Authentication R...

...-D8618427 To: <sip:0100@<CENSORED>;user=phone> CSeq: 2 INVITE Call-ID: d2c5011e-1d718f7-9203b936@<CENSORED POLYCOM IP> Contact: <sip:3109@<CENSORED POLYCOM IP>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_450-UA/4.0.7.2514 Accept-Language: en Supported: 100rel,replaces Allow-Events: conference,talk,hold Authorization: Digest username="3109", realm="asterisk", nonce="<CENSORED>", uri="sip:0100@<CENSORED>:5060;user=phone", response="<CEN...

On Tuesday 25 June 2019 at 16:49:23, Doug Lytle wrote: > We have Polycom phones (I'm using a VVX601, the destination is a VVX301). > We're also on Asterisk 13. > > I forwarded my call to the VVX301 and then dialed my phones DID. Surely that is "call forwarding", which is quite different from either a blind or attended transfer? A transfer involves a call coming