reco

callcentric only accepts 11digit numbers so my dial plan looks like this: prio trunk pattern replacement 98 callcentric 411 99 Try Loopback xx 200 callcentric 011* 011* 202 callcentric xxxxxxxxxx 1* 203 callcentric xxxxxxxxxxx * version 3.2.0.3143 works as expected dial: 222 333 4444 the trunk gets 1 222 333 4444 after upgrading to 3.3.0.3156 dial: 222 333 4444 the trunk gets 222 333 4444 i know this is already another topic... but... sorry thanx

ATTENTION!!! after upgrade outgoing calls do not work anymore... seems like something with the dialplans did get messup.... i am getting a message form callcentric that the number cannot be dialed as entered. back to 3.2.0.3143 and trunks are working again. 9] 20090311120259: SIP Rx tls:PBXIP:2092: INVITE sip:2223334444@domain.com;user=phone SIP/2.0 Via: SIP/2.0/TLS PBXIP:2092;branch=z9hG4bK-hzyu0mnjw7ur;rport From: "Christof Haemmerle" <sip:33@domain.com>;tag=uz3ub5iepq To: <sip:2223334444@domain.com;user=phone> Call-ID: 3c267f040947-s0n92pf3dhh4 CSeq: 1 INVITE Max-Forwards: 70 Contact: <sip:33@PBXIP:2092;transport=tls;line=3lalk32m>;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 1261499776 1261499776 IN IP4 PBXIP s=call c=IN IP4 PBXIP t=0 0 m=audio 49258 RTP/AVP 9 0 8 2 3 18 4 101 a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:XqGHT9IY82/nH54UAbnMsBdqMEe7u+hlebOorv2d 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] 20090311120259: Packet authenticated by transport layer [9] 20090311120259: UDP: Opening socket on 0.0.0.0:51702 [9] 20090311120259: UDP: Opening socket on 0.0.0.0:51703 [9] 20090311120259: UDP: Opening socket on [::]:51702 [9] 20090311120259: UDP: Opening socket on [::]:51703 [1] 20090311120259: UDP: TOS could not be set [1] 20090311120259: Last message repeated 2 times [8] 20090311120259: Could not find a trunk (3 trunks) [9] 20090311120259: Using outbound proxy sip:PBXIP:2092;transport=tls because of flow-label [9] 20090311120259: Last message repeated 2 times [9] 20090311120259: SIP Tx tls:PBXIP:2092: SIP/2.0 100 Trying Via: SIP/2.0/TLS PBXIP:2092;branch=z9hG4bK-hzyu0mnjw7ur;rport=2092 From: "Christof Haemmerle" <sip:33@domain.com>;tag=uz3ub5iepq To: <sip:2223334444@domain.com;user=phone>;tag=1aa0ae9577 Call-ID: 3c267f040947-s0n92pf3dhh4 CSeq: 1 INVITE Content-Length: 0 [7] 20090311120259: Set packet length to 20 [6] 20090311120259: Sending RTP for 3c267f040947-s0n92pf3dhh4#1aa0ae9577 to PBXIP:49258 [9] 20090311120259: Dialplan: Evaluating !^(411)@.*!sip:\1@\r;user=phone!i against 2223334444@domain.com [9] 20090311120259: Dialplan: Evaluating !^([0-9]{10})@.*!sip:1\1@\r;user=phone!i against 2223334444@domain.com [5] 20090311120259: Dialplan voip: Match 2223334444@domain.com to <sip:12223334444@callcentric.com;user=phone> on trunk callcentric-domain [9] 20090311120259: UDP: Opening socket on 0.0.0.0:54758 [9] 20090311120259: UDP: Opening socket on 0.0.0.0:54759 [8] 20090311120259: Play audio_moh/noise.wav [9] 20090311120259: UDP: Opening socket on [::]:54758 [9] 20090311120259: UDP: Opening socket on [::]:54759 [1] 2009

i want to set the following parameters which are not in the PnP settings in the admin interface. holding_reminder!: off user_ringer1!: Ringer7 ring_sound!: Ringer7 alert_internal_ring_sound!: Ringer7 alert_external_ring_sound!: Ringer7 i looked at this post but cannot figure out how to get this work http://forum.pbxnsip.com/index.php?showtop...8&hl=ringer any help? thanx

Updated to 3156. And just works again. with7.1.x and 7.3.x firmware thanx! Reco

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?

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

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

glad you guys see the same problems. looking forward for a fix reco

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...

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

i have the same problem here. i was never worried about the co lines but also intercom does not work anymore will write a new post since its not related to this. thanx

this is wired, is there a way to tell the pbx to flush that cache. this just started to happen since callcentric did some work on their network. no changes on the pbx. after a restart it works again for some time. christof

i have the crash report attached. any idea what happened? i had most of the log stuff turned off no real info in the log. attached there is a crash report. pbxctrl_darwin9.0_2009_01_20_170634_voice.crash.zip Pbxnsip_error.log.zip

i am on version 3.0.1 and this does not work. unfortunately. any idea how this needs to be done?

hi hki zip is the standard compression format build into the mac os x server. its pretty standard since this feature was introduced in the os 2-3 years back. christof

cool thanx looking forward for the a new release

hi there i am using google apps mail service to send all mails with pbxnsip. most of the part this works pretty well. but sometimes its stuck in a loop where it repots in the logs that the connection got refused but the mail was sent. so i end up getting hundres of messages ;( platform: mac os x 10.5.5 Version: 3.0.0.2998 (Darwin) smtp server config in domain: smtp.gmail.com:587 log: [8] 20080929140103: SMTP: Connect to 74.125.93.109:587 [8] 20080929140103: SMTP: Received 220 mx.google.com ESMTP 6sm218032qwk.1 [8] 20080929140103: SMTP: Received 250-mx.google.com at your service, [pbxip] 250-SIZE 35651584 250-8BITMIME 250-STARTTLS 250 ENHANCEDSTATUSCODES [8] 20080929140103: SMTP: Received 220 2.0.0 Ready to start TLS [8] 20080929140103: SMTP: Received [8] 20080929140103: Last message repeated 2 times [8] 20080929140103: SMTP: Received 334 VXNlcm5hbWU6 [8] 20080929140103: SMTP: Received 334 UGFzc3dvcmQ6 [8] 20080929140104: SMTP: Received 235 2.7.0 Accepted [8] 20080929140104: SMTP: Received 250 2.1.0 OK 6sm218032qwk.1 [8] 20080929140104: SMTP: Received 250 2.1.5 OK 6sm218032qwk.1 [8] 20080929140104: SMTP: Received 354 Go ahead 6sm218032qwk.1 [8] 20080929140105: SMTP: Received 250 2.0.0 OK 1222711265 6sm218032qwk.1 [5] 20080929140105: SMTP: Connection refused on 74.125.93.109:587

thanx for the xml file but i still could not figure out how to change this for snom phones. <tone name="custom1"> <vendor ua="Snom.*" type="alert-info">Custom 1</vendor> <vendor type="alert-info">????</vendor> </tone> could the value of the alert-info also be an url to a wav or mp3 file? thanx christof

yes i second that plus and option to record more than one greeting and choose one ad active greeting through web interface and the voice menu. i actually thought this was included in version 3 but i cannot find it. christof

did somebody try this with the snom 320? could not find a template and does not seem so easy to me to configure. i would love just to choose form the ringers in the phone "ringer 7" thanx

hi pradeep, just installed pbxnsip on mac. works good so far. any idea when you will be using my installer scheme with launchd and install path /Library/Pbsnsip? thanx christof

if my pbxnsip looses the internet the trunks do not reregister automatically. trunk page displays: 500 Address Resolution Failed (Registration failed, retry after 60 seconds) but it does not reregister. also clicking `reregister` does not do the trick. a reboot of pbxnsip does solve this. the internet connection is up while clicking `reregister` Version: 3.0.0.2993 (Linux) christof

nice, if this is kind of stable i would switch from 2 mac mini's running debian to maosx right away. christof

i am afraid i need a link for the debian version

he guys amazing.... amazing guys, finally... requests: - lets include the address book dialer posted on this forum. creating install package: 1. download apple developer tools 2. in /Developer/Applications/Utilities/Packagemaker.app is your friend. no clue about the time issue. christof