reco Posted February 18, 2009 Report Share Posted February 18, 2009 i had a working intercom setup. after the upgrade to 3.2 the phones only ring but do not auto answer domain intercom feature code *90 extension i try to intercom 34 extension intercom permission on both phones: * calling *9034 makes the phone only ring 3.2.0.3143 (Darwin) snom 320 7.3.14 phone setting intercom_enabled!: on intercom_connect_type!: intercom_connect_type_handsfree what am i missing? thanx Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted February 18, 2009 Report Share Posted February 18, 2009 what am i missing? Hmm. Works here... Also on version 7.3.14. Do you see "No permission for intercom to ..." in the log (level 5)? Quote Link to comment Share on other sites More sharing options...
nate Posted February 20, 2009 Report Share Posted February 20, 2009 I'm having a hard time getting the intercom feature to work on snom handsets. for example, extension 600, i dial *90600 it should auto pick up that phone. instead, it just rings the extension. Any one have any input? Quote Link to comment Share on other sites More sharing options...
reco Posted February 20, 2009 Author Report Share Posted February 20, 2009 all log levels to 9 and every option turned on. nothing with ``intercom`` or ``permission`` in the logfiles. again the feature code for intercom is: *90 the target extension: 34 the origin extension: 33 on EXT 33 I dial: *9034 -> ext 34 just rings. settings from the target extension: intercom_enabled!: on intercom_connect_type!: intercom_connect_type_handsfree no clue what's happening here... Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted February 21, 2009 Report Share Posted February 21, 2009 settings from the target extension: intercom_enabled!: on intercom_connect_type!: intercom_connect_type_handsfree The settings on the phone should not make the big difference. Can you get the SIP INVITE from the phones web trace log? Quote Link to comment Share on other sites More sharing options...
Bill H Posted February 21, 2009 Report Share Posted February 21, 2009 I discovered this too on the CS-410 with the 3144 version. Dial *90# and it will ask you to "Please enter the extension number" Then it will work like an intercom. Why has this been changed? Quote Link to comment Share on other sites More sharing options...
mattlandis Posted February 21, 2009 Report Share Posted February 21, 2009 I discovered this too on the CS-410 with the 3144 version. Dial *90# and it will ask you to "Please enter the extension number" Then it will work like an intercom. Why has this been changed? For us when I dial *90XXX (xxx=ext.) is just rings the phone. (or do it the way you explianed) This WORKED for us in the previous version. (we are running windows version) we are using provisioned snom360s. tx matt Quote Link to comment Share on other sites More sharing options...
reco Posted February 24, 2009 Author Report Share Posted February 24, 2009 glad you guys see the same problems. looking forward for a fix reco Quote Link to comment Share on other sites More sharing options...
pbx support Posted February 25, 2009 Report Share Posted February 25, 2009 glad you guys see the same problems.looking forward for a fix reco I tested it here and when the "Intercom to the following extensions:" is empty, it rings the phone and when I set it to "*" it intercoms (snom 320 & 360) Quote Link to comment Share on other sites More sharing options...
mattlandis Posted February 25, 2009 Report Share Posted February 25, 2009 i have * in the "Intercom to the following extensions:" for the extension initiating the intercom and the phone getting the intercom just rings... snom 360 there remains a problem. tx matt Quote Link to comment Share on other sites More sharing options...
pbx support Posted February 25, 2009 Report Share Posted February 25, 2009 i have * in the "Intercom to the following extensions:" for the extension initiating the intercom and the phone getting the intercom just rings...snom 360 there remains a problem. tx matt What OS? I tested it on Windows and CS410. Both seem to work ok. BTW, what is the called UA? (snom, polycom, cisco?) Quote Link to comment Share on other sites More sharing options...
pbx support Posted February 25, 2009 Report Share Posted February 25, 2009 What OS? I tested it on Windows and CS410. Both seem to work ok. BTW, what is the called UA? (snom, polycom, cisco?) Checkout http://wiki.snom.com/Settings/answer_after_policy. Also, http://wiki.snom.com/Settings/intercom_enabled Quote Link to comment Share on other sites More sharing options...
mattlandis Posted February 25, 2009 Report Share Posted February 25, 2009 What OS? I tested it on Windows and CS410. Both seem to work ok. BTW, what is the called UA? (snom, polycom, cisco?) -all phones involved are snom 360 -this worked before upgrading to the latest version -version=windows. tx matt Quote Link to comment Share on other sites More sharing options...
nate Posted March 3, 2009 Report Share Posted March 3, 2009 I am still having this problem... so does anyone have a fix? on the phones, i have ' answer after policy' always. enable intercom on i don't know what these settings were prior to latest firmware update. this was a very nice feature. Can someone please help? Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted March 3, 2009 Report Share Posted March 3, 2009 I am still having this problem... so does anyone have a fix? on the phones, i have ' answer after policy' always. enable intercom on i don't know what these settings were prior to latest firmware update. this was a very nice feature. Can someone please help? I would factory-reset the phone and provision it automatically. There is no need to change the configuration; the PBX will signal the intercom feature in the INVITE. Quote Link to comment Share on other sites More sharing options...
nate Posted March 3, 2009 Report Share Posted March 3, 2009 Ok, so i reset the phone back to factory defaults. it still does the same thing. the phone won't auto anwer, and now, i can't login to the web interface on the phone. it prompts me for a login, and my extension doesn't work, neither does admin and no password. Quote Link to comment Share on other sites More sharing options...
reco Posted March 3, 2009 Author Report Share Posted March 3, 2009 hi pbxnsip, also just reset both phones again dialed *9034 phone was ringing... i think we need to digg deeper. what do you need? thanx reco Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted March 3, 2009 Report Share Posted March 3, 2009 what do you need? The INVITE from the phone's log (the one that receives the INVITE) would be interesting! Quote Link to comment Share on other sites More sharing options...
reco Posted March 4, 2009 Author Report Share Posted March 4, 2009 here the phone log: Received from tls:10.0.24.2:5061 at 3/3/2009 23:05:02:428 (956 bytes): INVITE sip:34@10.0.24.23:2056;transport=tls;line=1hjzdmsd SIP/2.0 Via: SIP/2.0/TLS 10.0.24.2:5061;branch=z9hG4bK-5d2ca0c5b6214ab9378d950f0d8c5694;rport From: "Christof Haemmerle" <sip:33@nex9.com>;tag=1030602364 To: "Tina Preschitz" <sip:34@nex9.com> Call-ID: 31be71b6@pbx CSeq: 24165 INVITE Max-Forwards: 70 Contact: <sip:34@10.0.24.2:5061;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/3.2.0.3143 Content-Type: application/sdp Content-Length: 354 v=0 o=- 1469141092 1469141092 IN IP4 10.0.24.2 s=- c=IN IP4 10.0.24.2 t=0 0 m=audio 62894 RTP/AVP 0 8 3 2 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:JnUQo6gyigtj8YS0iNRIXM89hzVkINALVaPk6rUH a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv Sent to tls:10.0.24.2:5061 at 3/3/2009 23:05:02:472 (525 bytes): SIP/2.0 180 Ringing Via: SIP/2.0/TLS 10.0.24.2:5061;branch=z9hG4bK-5d2ca0c5b6214ab9378d950f0d8c5694;rport=5061 From: "Christof Haemmerle" <sip:33@nex9.com>;tag=1030602364 To: "Tina Preschitz" <sip:34@nex9.com>;tag=r4696z55pr Call-ID: 31be71b6@pbx CSeq: 24165 INVITE Contact: <sip:34@10.0.24.23:2056;transport=tls;line=1hjzdmsd>;reg-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:10.0.24.2:5061 at 3/3/2009 23:05:02:613 (420 bytes): PRACK sip:34@10.0.24.23:2056;transport=tls;line=1hjzdmsd SIP/2.0 Via: SIP/2.0/TLS 10.0.24.2:5061;branch=z9hG4bK-3bb0c2280d9f577775208f94e989a13e;rport From: "Christof Haemmerle" <sip:33@nex9.com>;tag=1030602364 To: "Tina Preschitz" <sip:34@nex9.com>;tag=r4696z55pr Call-ID: 31be71b6@pbx CSeq: 24166 PRACK Max-Forwards: 70 Contact: <sip:34@10.0.24.2:5061;transport=tls> RAck: 1 24165 INVITE Content-Length: 0 Sent to tls:10.0.24.2:5061 at 3/3/2009 23:05:02:624 (359 bytes): SIP/2.0 200 Ok Via: SIP/2.0/TLS 10.0.24.2:5061;branch=z9hG4bK-3bb0c2280d9f577775208f94e989a13e;rport=5061 From: "Christof Haemmerle" <sip:33@nex9.com>;tag=1030602364 To: "Tina Preschitz" <sip:34@nex9.com>;tag=r4696z55pr Call-ID: 31be71b6@pbx CSeq: 24166 PRACK Contact: <sip:34@10.0.24.23:2056;transport=tls;line=1hjzdmsd>;reg-id=1 Content-Length: 0 Received from tls:10.0.24.2:5061 at 3/3/2009 23:05:04:091 (337 bytes): CANCEL sip:34@10.0.24.23:2056;transport=tls;line=1hjzdmsd SIP/2.0 Via: SIP/2.0/TLS 10.0.24.2:5061;branch=z9hG4bK-5d2ca0c5b6214ab9378d950f0d8c5694;rport From: "Christof Haemmerle" <sip:33@nex9.com>;tag=1030602364 To: "Tina Preschitz" <sip:34@nex9.com> Call-ID: 31be71b6@pbx CSeq: 24165 CANCEL Max-Forwards: 70 Content-Length: 0 Sent to tls:10.0.24.2:5061 at 3/3/2009 23:05:04:099 (288 bytes): SIP/2.0 200 OK Via: SIP/2.0/TLS 10.0.24.2:5061;branch=z9hG4bK-5d2ca0c5b6214ab9378d950f0d8c5694;rport=5061 From: "Christof Haemmerle" <sip:33@nex9.com>;tag=1030602364 To: "Tina Preschitz" <sip:34@nex9.com>;tag=r4696z55pr Call-ID: 31be71b6@pbx CSeq: 24165 CANCEL Content-Length: 0 Sent to tls:10.0.24.2:5061 at 3/3/2009 23:05:04:111 (376 bytes): SIP/2.0 487 Request Terminated Via: SIP/2.0/TLS 10.0.24.2:5061;branch=z9hG4bK-5d2ca0c5b6214ab9378d950f0d8c5694;rport=5061 From: "Christof Haemmerle" <sip:33@nex9.com>;tag=1030602364 To: "Tina Preschitz" <sip:34@nex9.com>;tag=r4696z55pr Call-ID: 31be71b6@pbx CSeq: 24165 INVITE Contact: <sip:34@10.0.24.23:2056;transport=tls;line=1hjzdmsd>;reg-id=1 Content-Length: 0 Received from tls:10.0.24.2:5061 at 3/3/2009 23:05:04:219 (394 bytes): ACK sip:34@10.0.24.23:2056;transport=tls;line=1hjzdmsd SIP/2.0 Via: SIP/2.0/TLS 10.0.24.2:5061;branch=z9hG4bK-5d2ca0c5b6214ab9378d950f0d8c5694;rport From: "Christof Haemmerle" <sip:33@nex9.com>;tag=1030602364 To: "Tina Preschitz" <sip:34@nex9.com>;tag=r4696z55pr Call-ID: 31be71b6@pbx CSeq: 24165 ACK Max-Forwards: 70 Contact: <sip:34@10.0.24.2:5061;transport=tls> Content-Length: 0 here the pbxnsip log [9] 20090303225858: SIP Rx tls:10.0.24.24:2056: INVITE sip:*9034@nex9.com;user=phone SIP/2.0 Via: SIP/2.0/TLS 10.0.24.24:2056;branch=z9hG4bK-vwhdetk7intn;rport From: "Christof Haemmerle" <sip:33@nex9.com>;tag=f6yifu0fcj To: <sip:*9034@nex9.com;user=phone> Call-ID: 3c2712760257-tf8ewlridkyh CSeq: 1 INVITE Max-Forwards: 70 Contact: <sip:33@10.0.24.24:2056;transport=tls;line=4zhgbysz>;reg-id=1 P-Key-Flags: keys="3" User-Agent: snom320/7.3.14 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Session-Expires: 3600;refresher=uas Min-SE: 90 Proxy-Require: buttons Content-Type: application/sdp Content-Length: 448 v=0 o=root 1240381312 1240381312 IN IP4 10.0.24.24 s=call c=IN IP4 10.0.24.24 t=0 0 m=audio 63900 RTP/AVP 9 0 8 2 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:9QkeySQSHNoHQxwWRkkQO3F9u1Zs4/lIEMgvFsFF a=rtpmap:9 g722/8000 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv [8] 20090303225858: Packet authenticated by transport layer [9] 20090303225858: UDP: Opening socket on port 50530 [9] 20090303225858: UDP: Opening socket on port 50531 [9] 20090303225858: UDPv6: Opening socket on port 50530 [9] 20090303225858: UDPv6: Opening socket on port 50531 [1] 20090303225858: UDP: TOS could not be set [1] 20090303225858: Last message repeated 2 times [8] 20090303225858: Could not find a trunk (3 trunks) [9] 20090303225858: Using outbound proxy sip:10.0.24.24:2056;transport=tls because of flow-label [9] 20090303225858: Last message repeated 2 times [9] 20090303225858: SIP Tx tls:10.0.24.24:2056: SIP/2.0 100 Trying Via: SIP/2.0/TLS 10.0.24.24:2056;branch=z9hG4bK-vwhdetk7intn;rport=2056 From: "Christof Haemmerle" <sip:33@nex9.com>;tag=f6yifu0fcj To: <sip:*9034@nex9.com;user=phone>;tag=d6ea99ba48 Call-ID: 3c2712760257-tf8ewlridkyh CSeq: 1 INVITE Content-Length: 0 [7] 20090303225858: Set packet length to 20 [6] 20090303225858: Sending RTP for 3c2712760257-tf8ewlridkyh#d6ea99ba48 to 10.0.24.24:63900 [8] 20090303225858: Play audio_moh/noise.wav [9] 20090303225858: UDP: Opening socket on port 54934 [9] 20090303225858: UDP: Opening socket on port 54935 [9] 20090303225858: UDPv6: Opening socket on port 54934 [9] 20090303225858: UDPv6: Opening socket on port 54935 [1] 20090303225858: UDP: TOS could not be set [1] 20090303225858: Last message repeated 2 times [9] 20090303225858: Using outbound proxy sip:10.0.24.23:2056;transport=tls because of flow-label [9] 20090303225858: SIP Tx tls:10.0.24.23:2056: INVITE sip:34@10.0.24.23:2056;transport=tls;line=1hjzdmsd SIP/2.0 Via: SIP/2.0/TLS 10.0.24.2:5061;branch=z9hG4bK-d7be59d02d9fb308cef2e19b334d015f;rport From: "Christof Haemmerle" <sip:33@nex9.com>;tag=311957039 To: "Tina Preschitz" <sip:34@nex9.com> Call-ID: cb1fa236@pbx CSeq: 28978 INVITE Max-Forwards: 70 Contact: <sip:34@10.0.24.2:5061;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/3.2.0.3143 Content-Type: application/sdp Content-Length: 352 v=0 o=- 376158563 376158563 IN IP4 10.0.24.2 s=- c=IN IP4 10.0.24.2 t=0 0 m=audio 54934 RTP/AVP 0 8 3 2 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:FMw/tCREc0JA5/5jLpkctEANAGxJX6bVwcLGQ1Cz a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv [7] 20090303225858: Set packet length to 20 [9] 20090303225858: SIP Rx tls:10.0.24.23:2056: SIP/2.0 180 Ringing Via: SIP/2.0/TLS 10.0.24.2:5061;branch=z9hG4bK-d7be59d02d9fb308cef2e19b334d015f;rport=5061 From: "Christof Haemmerle" <sip:33@nex9.com>;tag=311957039 To: "Tina Preschitz" <sip:34@nex9.com>;tag=3tezmb6ixm Call-ID: cb1fa236@pbx CSeq: 28978 INVITE Contact: <sip:34@10.0.24.23:2056;transport=tls;line=1hjzdmsd>;reg-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 [9] 20090303225858: SIP Tx tls:10.0.24.23:2056: PRACK sip:34@10.0.24.23:2056;transport=tls;line=1hjzdmsd SIP/2.0 Via: SIP/2.0/TLS 10.0.24.2:5061;branch=z9hG4bK-ae98c15052991447af66ffbb02e6b539;rport From: "Christof Haemmerle" <sip:33@nex9.com>;tag=311957039 To: "Tina Preschitz" <sip:34@nex9.com>;tag=3tezmb6ixm Call-ID: cb1fa236@pbx CSeq: 28979 PRACK Max-Forwards: 70 Contact: <sip:34@10.0.24.2:5061;transport=tls> RAck: 1 28978 INVITE Content-Length: 0 [8] 20090303225858: Play audio_en/ringback.wav [9] 20090303225858: SIP Tx tls:10.0.24.24:2056: SIP/2.0 183 Ringing Via: SIP/2.0/TLS 10.0.24.24:2056;branch=z9hG4bK-vwhdetk7intn;rport=2056 From: "Christof Haemmerle" <sip:33@nex9.com>;tag=f6yifu0fcj To: <sip:*9034@nex9.com;user=phone>;tag=d6ea99ba48 Call-ID: 3c2712760257-tf8ewlridkyh CSeq: 1 INVITE Contact: <sip:33@10.0.24.2:5061;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/3.2.0.3143 Require: 100rel RSeq: 1 Content-Type: application/sdp Content-Length: 366 v=0 o=- 1421313999 1421313999 IN IP4 10.0.24.2 s=- c=IN IP4 10.0.24.2 t=0 0 m=audio 50530 RTP/AVP 0 8 3 2 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:fLIdE/rUu5cghG97S14nBTgil4tkNU66VstPOaVQ a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv [9] 20090303225858: SIP Rx tls:10.0.24.24:2056: PRACK sip:33@10.0.24.2:5061;transport=tls SIP/2.0 Via: SIP/2.0/TLS 10.0.24.24:2056;branch=z9hG4bK-ieriregozcix;rport From: "Christof Haemmerle" <sip:33@nex9.com>;tag=f6yifu0fcj To: <sip:*9034@nex9.com;user=phone>;tag=d6ea99ba48 Call-ID: 3c2712760257-tf8ewlridkyh CSeq: 2 PRACK Max-Forwards: 70 Contact: <sip:33@10.0.24.24:2056;transport=tls;line=4zhgbysz>;reg-id=1 RAck: 1 1 INVITE Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer, call-info Proxy-Require: buttons Content-Length: 0 [8] 20090303225858: Packet authenticated by transport layer [9] 20090303225858: SIP Tx tls:10.0.24.24:2056: SIP/2.0 200 Ok Via: SIP/2.0/TLS 10.0.24.24:2056;branch=z9hG4bK-ieriregozcix;rport=2056 From: "Christof Haemmerle" <sip:33@nex9.com>;tag=f6yifu0fcj To: <sip:*9034@nex9.com;user=phone>;tag=d6ea99ba48 Call-ID: 3c2712760257-tf8ewlridkyh CSeq: 2 PRACK Contact: <sip:33@10.0.24.2:5061;transport=tls> User-Agent: pbxnsip-PBX/3.2.0.3143 Content-Length: 0 [7] 20090303225859: Receiving DTMF on codec 101 [9] 20090303225859: SIP Rx tls:10.0.24.23:2056: SIP/2.0 200 Ok Via: SIP/2.0/TLS 10.0.24.2:5061;branch=z9hG4bK-ae98c15052991447af66ffbb02e6b539;rport=5061 From: "Christof Haemmerle" <sip:33@nex9.com>;tag=311957039 To: "Tina Preschitz" <sip:34@nex9.com>;tag=3tezmb6ixm Call-ID: cb1fa236@pbx CSeq: 28979 PRACK Contact: <sip:34@10.0.24.23:2056;transport=tls;line=1hjzdmsd>;reg-id=1 Content-Length: 0 [7] 20090303225859: Call cb1fa236@pbx#311957039: Clear last request [9] 20090303225900: SIP Rx udp:10.0.24.20:5060: SUBSCRIBE sip:10.0.24.2 SIP/2.0 Via: SIP/2.0/UDP 10.0.24.20;branch=z9hG4bK33hx4q971oe1oe Max-Forwards: 70 From: <sip:32@buero-newyork.com>;tag=niwan To: <sip:32@buero-newyork.com>;tag=79e2a83e4f Call-ID: 3f6gv27wo85af@buero-newyork.com CSeq: 8616 SUBSCRIBE Contact: <sip:32@10.0.24.20> Accept: application/simple-message-summary Allow: INVITE, CANCEL, BYE, ACK, REGISTER, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE, INFO, PRACK Allow-Events: dialog,message-summary Authorization: Digest username="32", realm="buero-newyork.com", nonce="bcb648cf721f1c55464482e81d2fabce", uri="sip:10.0.24.2", response="1e01d5b0a7a737753a7555c74fd01bb2", algorithm=MD5 Event: message-summary Expires: 28 Supported: replaces User-Agent: snom-m3-SIP/01.22 (MAC=0004132A0F44; HW=1) Content-Type: text/plain Content-Length: 0 [9] 20090303225900: Resolve 1217647: aaaa udp 10.0.24.20 5060 [9] 20090303225900: Resolve 1217647: a udp 10.0.24.20 5060 [9] 20090303225900: Resolve 1217647: udp 10.0.24.20 5060 [9] 20090303225900: SIP Tx udp:10.0.24.20:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 10.0.24.20;branch=z9hG4bK33hx4q971oe1oe From: <sip:32@buero-newyork.com>;tag=niwan To: <sip:32@buero-newyork.com>;tag=79e2a83e4f Call-ID: 3f6gv27wo85af@buero-newyork.com CSeq: 8616 SUBSCRIBE Contact: <sip:10.0.24.2:5060;transport=udp> Expires: 32 Content-Length: 0 Quote Link to comment Share on other sites More sharing options...
pbx support Posted March 4, 2009 Report Share Posted March 4, 2009 here the phone log: Received from tls:10.0.24.2:5061 at 3/3/2009 23:05:02:428 (956 bytes): Interesting.. the invite sent by the PBX to the called party should have either "Call-Info" header or "Alert-Info" header. Call-Info: <sip:41@pbx.company.com>;answer-after=0OrAlert-Info: Auto Answer I just tested this on pbxnsip-PBX/3.2.0.3143 (Mac OS) and it worked fine. PBX and both phones are on the same network, snom-320 using version 7.1.35 (and 7.3.14) Quote Link to comment Share on other sites More sharing options...
reco Posted March 4, 2009 Author Report Share Posted March 4, 2009 hi there, pbx running on mac os x 10.5.6 phone and pbx on the same subnet phone firmware version 7.3.14 is this a bug? Quote Link to comment Share on other sites More sharing options...
pbx support Posted March 4, 2009 Report Share Posted March 4, 2009 hi there, pbx running on mac os x 10.5.6 phone and pbx on the same subnet phone firmware version 7.3.14 is this a bug? Since it is working in our lab with the same versions, it is hard to declare it as a bug yet. We are looking into other aspect that could impact the intercom behavior now. BTW, there is a new version for both windows and mac - http://www.pbxnsip.com/protect/pbxctrl-3.3.0.3156.exe, http://www.pbxnsip.com/protect/pbx-darwin9.0-3.3.0.3156.zip If you want to give the Mac version a spin, feel free(make a back up of the old data). Intercom works ok on both the versions here. Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted March 5, 2009 Report Share Posted March 5, 2009 Check the status of the setting called "answer_after_policy" on the phone. It should be "idle", not "off". Quote Link to comment Share on other sites More sharing options...
nate Posted March 9, 2009 Report Share Posted March 9, 2009 Check the status of the setting called "answer_after_policy" on the phone. It should be "idle", not "off". Could someone post a ini file of a working snom phone. i don't think it matters what phone. i'm thinking this happened after latest firmware update. Quote Link to comment Share on other sites More sharing options...
nate Posted March 9, 2009 Report Share Posted March 9, 2009 Could someone post a ini file of a working snom phone. i don't think it matters what phone. i'm thinking this happened after latest firmware update. I just tried, *90 it asks "please enter the extension number" i dial "ext" it says, you are not allowed to place this call. Quote Link to comment Share on other sites More sharing options...
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