RobertO Posted January 17, 2008 Report Share Posted January 17, 2008 When an external caller ( A ) calls an internal destination ( B ) the phone ( B ) shows the right caller ID from ( A ). After an attended transerfer to an other internal Destination ( C ) the phone ( C ) shows the Caller ID from Phone ( B ). Is this by design or is there a way to control this behavior. In my opinion this should be optimal: Phone ( C ) see in the ringstate Caller ID ( B ) and after a successful transfer the phone ( C ) see caller ID ( A ) Quote Link to comment Share on other sites More sharing options...
Kristan Posted January 17, 2008 Report Share Posted January 17, 2008 It depends on the type of phone and what it supports - see my post here : http://pbxnsip.invisionzone.com/index.php?showtopic=598 Snoms do, I think Linksys do, polycoms don't. Quote Link to comment Share on other sites More sharing options...
RobertO Posted January 20, 2008 Author Report Share Posted January 20, 2008 Ok, I have understood where the problem is. But is there a solution? We are using Snom 360 (Fw7.1.30) with factory default plus PNP-Settings and the latest PBXnSIP Software. Is there a setting at the phones or the pbx that I have to change? The only thing I found is "Change names in To/From-headers:" in the domain-Settings. And I have not found any description for this setting. Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted January 20, 2008 Report Share Posted January 20, 2008 Ok, I have understood where the problem is. But is there a solution? We are using Snom 360 (Fw7.1.30) with factory default plus PNP-Settings and the latest PBXnSIP Software. That should work. Do you see a INFO packet during the call with an attachment that contains the changed caller-ID information? Do you have the SIP packets that are being exchanged between the PBX and phone that should display the changed ID? Quote Link to comment Share on other sites More sharing options...
RobertO Posted February 7, 2008 Author Report Share Posted February 7, 2008 Here is the SIP Protocol from Phone ( C ) There is an INFO message but the Caller ID from Phone ( A ) ist never seen... Phone ( A ) External Phone ( B ) IP 192.168.232.117 Phone ( C ) IP 192.168.232.90 PBX IP 192.168.232.10 An other phenomen is that the button of the called person is blinking. Any Idea???? <---------- Snip ----- Received from tls:192.168.232.10:5061 at 7/2/2008 20:09:53:370 (1038 bytes): INVITE sip:4406@192.168.232.90:2051;transport=tls;line=f03ilqr1 SIP/2.0 Via: SIP/2.0/TLS 192.168.232.10:5061;branch=z9hG4bK-4ab17271000e1823454fdfbdecd95195;rport From: "ROE-Demo " <sip:4451@sip.domain.de>;tag=3173 To: "ROE " <sip:4406@sip.domain.de> Call-ID: 02622e7d@pbx CSeq: 16185 INVITE Max-Forwards: 70 Contact: <sip:4406@192.168.232.10:5061;transport=tls> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/2.1.5.2357 Alert-Info: <http://127.0.0.1/Bellcore-dr2> Content-Type: application/sdp Content-Length: 378 v=0 o=- 40095 40095 IN IP4 192.168.232.10 s=- c=IN IP4 192.168.232.10 t=0 0 m=audio 58138 RTP/AVP 8 0 9 2 3 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:MHLtx19IYi7nF6NjlHBXJIZRbYLG6MgJ5LnW5sbG a=rtpmap:8 pcma/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv -------------------------------------------------------------------------------- Sent to tls:192.168.232.10:5061 at 7/2/2008 20:09:53:420 (532 bytes): SIP/2.0 180 Ringing Via: SIP/2.0/TLS 192.168.232.10:5061;branch=z9hG4bK-4ab17271000e1823454fdfbdecd95195;rport=5061 From: "ROE-Demo " <sip:4451@sip.domain.de>;tag=3173 To: "ROE " <sip:4406@sip.domain.de>;tag=wmc0u3besw Call-ID: 02622e7d@pbx CSeq: 16185 INVITE Contact: <sip:4406@192.168.232.90:2051;transport=tls;line=f03ilqr1>;flow-id=1 Require: 100rel RSeq: 1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Content-Length: 0 -------------------------------------------------------------------------------- Received from tls:192.168.232.10:5061 at 7/2/2008 20:09:53:679 (500 bytes): MESSAGE sip:4406@192.168.232.90:2051;transport=tls;line=f03ilqr1 SIP/2.0 Via: SIP/2.0/TLS 192.168.232.10:5061;branch=z9hG4bK-1f3c1d62036b0d14ee3486e175f7cbc9;rport From: "ROE " <sip:4406@sip.domain.de>;tag=65162 To: "ROE " <sip:4406@sip.domain.de> Call-ID: iu5xtytk@pbx CSeq: 12216 MESSAGE Max-Forwards: 70 Contact: <sip:192.168.232.10:5061;transport=tls> Subject: buttons Content-Type: application/x-buttons Content-Length: 53 k=40 c=pickup x=ext i=4451 n=*60114 a=invite -------------------------------------------------------------------------------- Sent to tls:192.168.232.10:5061 at 7/2/2008 20:09:53:810 (270 bytes): SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.232.10:5061;branch=z9hG4bK-1f3c1d62036b0d14ee3486e175f7cbc9;rport=5061 From: "ROE " <sip:4406@sip.domain.de>;tag=65162 To: "ROE " <sip:4406@sip.domain.de> Call-ID: iu5xtytk@pbx CSeq: 12216 MESSAGE Content-Length: 0 -------------------------------------------------------------------------------- Received from tls:192.168.232.10:5061 at 7/2/2008 20:09:53:817 (433 bytes): PRACK sip:4406@192.168.232.90:2051;transport=tls;line=f03ilqr1 SIP/2.0 Via: SIP/2.0/TLS 192.168.232.10:5061;branch=z9hG4bK-12d35d0e49ec9c6110c9ffcb59878593;rport From: "ROE-Demo " <sip:4451@sip.domain.de>;tag=3173 To: "ROE " <sip:4406@sip.domain.de>;tag=wmc0u3besw Call-ID: 02622e7d@pbx CSeq: 16186 PRACK Max-Forwards: 70 Contact: <sip:4406@192.168.232.10:5061;transport=tls> RAck: 1 16185 INVITE Content-Length: 0 -------------------------------------------------------------------------------- Sent to tls:192.168.232.10:5061 at 7/2/2008 20:09:53:827 (366 bytes): SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.232.10:5061;branch=z9hG4bK-12d35d0e49ec9c6110c9ffcb59878593;rport=5061 From: "ROE-Demo " <sip:4451@sip.domain.de>;tag=3173 To: "ROE " <sip:4406@sip.domain.de>;tag=wmc0u3besw Call-ID: 02622e7d@pbx CSeq: 16186 PRACK Contact: <sip:4406@192.168.232.90:2051;transport=tls;line=f03ilqr1>;flow-id=1 Content-Length: 0 -------------------------------------------------------------------------------- Sent to tls:192.168.232.10:5061 at 7/2/2008 20:09:57:708 (1025 bytes): SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.232.10:5061;branch=z9hG4bK-4ab17271000e1823454fdfbdecd95195;rport=5061 From: "ROE-Demo " <sip:4451@sip.domain.de>;tag=3173 To: "ROE " <sip:4406@sip.domain.de>;tag=wmc0u3besw Call-ID: 02622e7d@pbx CSeq: 16185 INVITE Contact: <sip:4406@192.168.232.90:2051;transport=tls;line=f03ilqr1>;flow-id=1 User-Agent: snom360/7.1.30 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, callerid Content-Type: application/sdp Content-Length: 417 v=0 o=root 1426542497 1426542498 IN IP4 192.168.232.90 s=call c=IN IP4 192.168.232.90 t=0 0 m=audio 52886 RTP/AVP 8 0 9 2 3 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:rs98bfKUTQhtHDhr6tNCmv1ZrIP7RmFHplOK0Y66 a=rtpmap:8 pcma/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=encryption:optional a=sendrecv -------------------------------------------------------------------------------- Received from tls:192.168.232.10:5061 at 7/2/2008 20:09:57:958 (407 bytes): ACK sip:4406@192.168.232.90:2051;transport=tls;line=f03ilqr1 SIP/2.0 Via: SIP/2.0/TLS 192.168.232.10:5061;branch=z9hG4bK-2acacff56c07853ba001d3fe4cf46e3c;rport From: "ROE-Demo " <sip:4451@sip.domain.de>;tag=3173 To: "ROE " <sip:4406@sip.domain.de>;tag=wmc0u3besw Call-ID: 02622e7d@pbx CSeq: 16185 ACK Max-Forwards: 70 Contact: <sip:4406@192.168.232.10:5061;transport=tls> Content-Length: 0 -------------------------------------------------------------------------------- Received from tls:192.168.232.10:5061 at 7/2/2008 20:09:57:963 (468 bytes): MESSAGE sip:4406@192.168.232.90:2051;transport=tls;line=f03ilqr1 SIP/2.0 Via: SIP/2.0/TLS 192.168.232.10:5061;branch=z9hG4bK-c538840cd1bf6ec221877a463f16c28f;rport From: "ROE " <sip:4406@sip.domain.de>;tag=35457 To: "ROE " <sip:4406@sip.domain.de> Call-ID: srjdn69q@pbx CSeq: 19739 MESSAGE Max-Forwards: 70 Contact: <sip:192.168.232.10:5061;transport=tls> Subject: buttons Content-Type: application/x-buttons Content-Length: 21 k=40 c=on x=ext -------------------------------------------------------------------------------- Sent to tls:192.168.232.10:5061 at 7/2/2008 20:09:57:1000 (270 bytes): SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.232.10:5061;branch=z9hG4bK-c538840cd1bf6ec221877a463f16c28f;rport=5061 From: "ROE " <sip:4406@sip.domain.de>;tag=35457 To: "ROE " <sip:4406@sip.domain.de> Call-ID: srjdn69q@pbx CSeq: 19739 MESSAGE Content-Length: 0 -------------------------------------------------------------------------------- Received from tls:192.168.232.10:5061 at 7/2/2008 20:10:01:175 (498 bytes): MESSAGE sip:4406@192.168.232.90:2051;transport=tls;line=f03ilqr1 SIP/2.0 Via: SIP/2.0/TLS 192.168.232.10:5061;branch=z9hG4bK-0d3ac7578e7fdfe873e61040634b3c07;rport From: "ROE " <sip:4406@sip.domain.de>;tag=15184 To: "ROE " <sip:4406@sip.domain.de> Call-ID: jlb2mrq5@pbx CSeq: 34430 MESSAGE Max-Forwards: 70 Contact: <sip:192.168.232.10:5061;transport=tls> Subject: buttons Content-Type: application/x-buttons Content-Length: 51 k=40 c=hold x=ext i=4451 n=*60115 a=invite -------------------------------------------------------------------------------- Sent to tls:192.168.232.10:5061 at 7/2/2008 20:10:01:215 (270 bytes): SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.232.10:5061;branch=z9hG4bK-0d3ac7578e7fdfe873e61040634b3c07;rport=5061 From: "ROE " <sip:4406@sip.domain.de>;tag=15184 To: "ROE " <sip:4406@sip.domain.de> Call-ID: jlb2mrq5@pbx CSeq: 34430 MESSAGE Content-Length: 0 -------------------------------------------------------------------------------- Received from tls:192.168.232.10:5061 at 7/2/2008 20:10:03:551 (528 bytes): INFO sip:4406@192.168.232.90:2051;transport=tls;line=f03ilqr1 SIP/2.0 Via: SIP/2.0/TLS 192.168.232.10:5061;branch=z9hG4bK-6dc49b483f259e6b15537191cf31cbb2;rport From: "ROE-Demo " <sip:4451@sip.domain.de>;tag=3173 To: "ROE " <sip:4406@sip.domain.de>;tag=wmc0u3besw Call-ID: 02622e7d@pbx CSeq: 16187 INFO Max-Forwards: 70 Contact: <sip:4406@192.168.232.10:5061;transport=tls> Content-Type: message/sipfrag Content-Length: 87 From: "ROE-Demo " <sip:4451@sip.domain.de> To: "ROE " <sip:4406@sip.domain.de> -------------------------------------------------------------------------------- Sent to tls:192.168.232.10:5061 at 7/2/2008 20:10:03:594 (365 bytes): SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.232.10:5061;branch=z9hG4bK-6dc49b483f259e6b15537191cf31cbb2;rport=5061 From: "ROE-Demo " <sip:4451@sip.domain.de>;tag=3173 To: "ROE " <sip:4406@sip.domain.de>;tag=wmc0u3besw Call-ID: 02622e7d@pbx CSeq: 16187 INFO Contact: <sip:4406@192.168.232.90:2051;transport=tls;line=f03ilqr1>;flow-id=1 Content-Length: 0 -------------------------------------------------------------------------------- Received from tls:192.168.232.10:5061 at 7/2/2008 20:10:03:602 (1008 bytes): INVITE sip:4406@192.168.232.90:2051;transport=tls;line=f03ilqr1 SIP/2.0 Via: SIP/2.0/TLS 192.168.232.10:5061;branch=z9hG4bK-5259f7fb7d7dc0c792e37d0e4f2fdeec;rport From: "ROE-Demo " <sip:4451@sip.domain.de>;tag=3173 To: "ROE " <sip:4406@sip.domain.de>;tag=wmc0u3besw Call-ID: 02622e7d@pbx CSeq: 16188 INVITE Max-Forwards: 70 Contact: <sip:4406@192.168.232.10:5061;transport=tls> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/2.1.5.2357 Content-Type: application/sdp Content-Length: 378 v=0 o=- 40095 40096 IN IP4 192.168.232.10 s=- c=IN IP4 192.168.232.10 t=0 0 m=audio 58138 RTP/AVP 8 0 9 2 3 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:MHLtx19IYi7nF6NjlHBXJIZRbYLG6MgJ5LnW5sbG a=rtpmap:8 pcma/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv -------------------------------------------------------------------------------- Sent to tls:192.168.232.10:5061 at 7/2/2008 20:10:03:674 (1025 bytes): SIP/2.0 200 Ok Via: SIP/2.0/TLS 192.168.232.10:5061;branch=z9hG4bK-5259f7fb7d7dc0c792e37d0e4f2fdeec;rport=5061 From: "ROE-Demo " <sip:4451@sip.domain.de>;tag=3173 To: "ROE " <sip:4406@sip.domain.de>;tag=wmc0u3besw Call-ID: 02622e7d@pbx CSeq: 16188 INVITE Contact: <sip:4406@192.168.232.90:2051;transport=tls;line=f03ilqr1>;flow-id=1 User-Agent: snom360/7.1.30 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, callerid Content-Type: application/sdp Content-Length: 417 v=0 o=root 1426542497 1426542499 IN IP4 192.168.232.90 s=call c=IN IP4 192.168.232.90 t=0 0 m=audio 52886 RTP/AVP 8 0 9 2 3 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:rs98bfKUTQhtHDhr6tNCmv1ZrIP7RmFHplOK0Y66 a=rtpmap:8 pcma/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=encryption:optional a=sendrecv -------------------------------------------------------------------------------- Received from tls:192.168.232.10:5061 at 7/2/2008 20:10:03:826 (407 bytes): ACK sip:4406@192.168.232.90:2051;transport=tls;line=f03ilqr1 SIP/2.0 Via: SIP/2.0/TLS 192.168.232.10:5061;branch=z9hG4bK-2acacff56c07853ba001d3fe4cf46e3c;rport From: "ROE-Demo " <sip:4451@sip.domain.de>;tag=3173 To: "ROE " <sip:4406@sip.domain.de>;tag=wmc0u3besw Call-ID: 02622e7d@pbx CSeq: 16188 ACK Max-Forwards: 70 Contact: <sip:4406@192.168.232.10:5061;transport=tls> Content-Length: 0 Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted February 7, 2008 Report Share Posted February 7, 2008 Well the problem is probably here that the Re-INVITE is killing it. RFC 4916 makes sense, but it is not supported by the phone. See http://wiki.pbxnsip.com/index.php/Indicati...ge_of_Caller-ID. Quote Link to comment Share on other sites More sharing options...
RobertO Posted February 7, 2008 Author Report Share Posted February 7, 2008 We are using Snom 360 phones with Firmware 7.1.30 and PBXnSIP (2.1.5.2357 (Win32)). The Snom Phones are factory default and have only the PNP settings applied. So I think nothing very exotic. As I know snom phones support RFC 4916. But I am not shure! I have no more Ideas what to do? Please help!!!! Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted February 8, 2008 Report Share Posted February 8, 2008 As I know snom phones support RFC 4916. But I am not shure! Nope. The supported header does not include the magic word "from-change". See http://www.ietf.org/rfc/rfc4916.txt. Quote Link to comment Share on other sites More sharing options...
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