mabbott

Can I get a link to where I can get a pbxnsip build of 4.2.0.3966, I tried the other day to used the link on the website, changing the build number to 3966, downloaded the file, but it has the executable for build for 4.2.0.3981. Thanks.

I was just playing with this yesterday. You just need to add 127.0.0.1 mask:255.255.255.255 to the Allow list.

Version:3.4.0.3201 The MWI/message count seems to not be getting updated when a moving a message is completed in a call to a mailbox. I am not seeing any notifies being sent from the system to update the message counts. I have done a few tests using an auto-provisioned Snom 370 as the only phone on an extension. Test 1: 1new/0saved indicated on phone and pbx, log into mailbox using phone and move message to another extension, message appears on other extension, however pbx says 0/0 but phone still says 1/0. Test 2: 2new/0saved indicated on phone and pbx, log into mailbox using phone and delete one message then move 2nd message to another extension, message appears on other extension, however pbx says 0/0 but phone still says 2/0. If deleting or saving all messages or using the * code to clear the MWI, the message counts will be updated/cleared on the phone.

I thought this worked for other feature codes beyond DND? I changed enable/disable forward all to the same code and it always says forwarding is disabled.

Yes, you are right, it is just a Javascript error. I changed it in the config file and it does toggle the features.

From Wiki "If the codes for DND on and off are the same, the PBX toggles between the two states. This is useful if you have a phone that just has DND programmed as a speed dial button." Has this feature been removed in 3.4? I cannot enter the same * code in DND on and off. This is useful for other codes beyond DND as well.

hmm, I just noticed something. If I restart pbxnsip once the computer is up an running, it will then work correctly, but if i reboot the computer, it doesn't come up correctly. Is pbxnsip maybe starting before the networking is totally up? Here are 2 logs, 1st is after a reboot 2nd after restarting pbxnsip. 1]Â 2009/09/25Â 09:28:37: Starting up version 3.4.0.3201 [7]Â 2009/09/25Â 09:28:37: Found time zones HST AKDT AKST PDT PST MDT MST CDT CST2 EDT EST ADT AST NDT NST BST CET GMT+2 GMT+3 GMT+4 GMT+5 GMT+6 GMT+7 GMT+8 GMT+9 CST CAT IST AUS1 AUS2 AUS3 AUS4 AUS5 AUS6 GMT [8]Â 2009/09/25Â 09:28:38: Scheduler precision is 0 us [1]Â 2009/09/25Â 09:28:38: Working Directory is /Library/pbxnsip [5]Â 2009/09/25Â 09:28:38: Starting threads [7]Â 2009/09/25Â 09:28:38: UDP: Opening socket on 0.0.0.0 [7]Â 2009/09/25Â 09:28:38: UDP: Opening socket on [::] [7]Â 2009/09/25Â 09:28:38: UDP: Opening socket on 0.0.0.0:5060 [8]Â 2009/09/25Â 09:28:38: Joined multicast group 224.0.1.75 [7]Â 2009/09/25Â 09:28:38: UDP: Opening socket on [::]:5060 [1]Â 2009/09/25Â 09:28:38: UDP: TOS could not be set [7]Â 2009/09/25Â 09:28:38: TCP: Opening socket on 0.0.0.0:5060 [7]Â 2009/09/25Â 09:28:38: TCP: Opening socket on [::]:5060 [7]Â 2009/09/25Â 09:28:38: TCP: Opening socket on 0.0.0.0:5061 [7]Â 2009/09/25Â 09:28:38: TCP: Opening socket on [::]:5061 [5]Â 2009/09/25Â 09:28:38: Set scheduling priority to 15 [7]Â 2009/09/25Â 09:28:38: TCP: Opening socket on 0.0.0.0:80 [7]Â 2009/09/25Â 09:28:38: TCP: Opening socket on [::]:80 [7]Â 2009/09/25Â 09:28:38: TCP: Opening socket on 0.0.0.0:443 [7]Â 2009/09/25Â 09:28:38: TCP: Opening socket on [::]:443 [7]Â 2009/09/25Â 09:28:38: UDP: Opening socket on 0.0.0.0:161 [7]Â 2009/09/25Â 09:28:38: UDP: Opening socket on [::]:161 [0]Â 2009/09/25Â 09:28:38: Could not open UDP port 69 for TFTP [0]Â 2009/09/25Â 09:29:17: Last message repeated 2 times [3]Â 2009/09/25Â 09:29:17: SMTP: Cannot resolve smtp 1]Â 2009/09/25Â 09:34:05: Starting up version 3.4.0.3201 [7]Â 2009/09/25Â 09:34:05: Found time zones HST AKDT AKST PDT PST MDT MST CDT CST2 EDT EST ADT AST NDT NST BST CET GMT+2 GMT+3 GMT+4 GMT+5 GMT+6 GMT+7 GMT+8 GMT+9 CST CAT IST AUS1 AUS2 AUS3 AUS4 AUS5 AUS6 GMT [8]Â 2009/09/25Â 09:34:05: Scheduler precision is 0 us [1]Â 2009/09/25Â 09:34:05: Working Directory is /Library/pbxnsip [7]Â 2009/09/25Â 09:34:05: UDP: Opening socket on 192.168.1.139 [5]Â 2009/09/25Â 09:34:05: Starting threads [7]Â 2009/09/25Â 09:34:05: UDP: Opening socket on 0.0.0.0 [7]Â 2009/09/25Â 09:34:05: UDP: Opening socket on [::] [7]Â 2009/09/25Â 09:34:05: UDP: Opening socket on 0.0.0.0:5060 [8]Â 2009/09/25Â 09:34:05: Joined multicast group 224.0.1.75 [7]Â 2009/09/25Â 09:34:05: UDP: Opening socket on [::]:5060 [1]Â 2009/09/25Â 09:34:05: UDP: TOS could not be set [7]Â 2009/09/25Â 09:34:05: TCP: Opening socket on 0.0.0.0:5060 [7]Â 2009/09/25Â 09:34:05: TCP: Opening socket on [::]:5060 [7]Â 2009/09/25Â 09:34:05: TCP: Opening socket on 0.0.0.0:5061 [7]Â 2009/09/25Â 09:34:05: TCP: Opening socket on [::]:5061 [7]Â 2009/09/25Â 09:34:05: TCP: Opening socket on 0.0.0.0:80 [7]Â 2009/09/25Â 09:34:05: TCP: Opening socket on [::]:80 [7]Â 2009/09/25Â 09:34:05: TCP: Opening socket on 0.0.0.0:443 [7]Â 2009/09/25Â 09:34:05: TCP: Opening socket on [::]:443 [7]Â 2009/09/25Â 09:34:05: UDP: Opening socket on 0.0.0.0:161 [7]Â 2009/09/25Â 09:34:05: UDP: Opening socket on [::]:161 [0]Â 2009/09/25Â 09:34:05: Could not open UDP port 69 for TFTP

Yes, it does have the name server that I want to use. I have also tried multiple name servers with the same results.

I am having an issue with my install on OSX resolving dns entries(pbx version 3.4.0.3201). I believe it started when I upgraded to Snow Leopard (10.6), but I am not 100% sure. The logs are showing errors trying to resolve both my email server and trunk registration server. If I change either to the servers ip, the email gets sent or registers the trunk. The computers DNS servers are correct and will resolve both hosts properly.

This is an option on the Snom phones. There is a setting called "Dial tone during hold" under advanced-->audio that is used to disable the dial tone or set cw_dialtone to off in one of the snom_xxx_custom.xml config files to change it globally.

The 3.4 build did not solve the problem. It still isn't switching back to inband dtmf when 2833 is removed from the re-invite.

Can I get this in a 3.3 cs410 version to test? The carrier won't give me a number to test the beta with and I can't test this with the customers equipment.

I would need a build for a CS410.

Is there a setting for force to inband? Or would this require a new build for a fix?

They gave me a little more information about what is going on. They say "when the call is re-INVITEd back to G.711 there is no RFC2833 DTMF listed in the SDP (and it should start detecting inband DTMF at that point)" Below are the logs from the call. Any help would be appreciated. INVITE sip:1877658xxxx@209.190.245.xxx:5060 SIP/2.0 v: SIP/2.0/UDP 207.138.151.33:5060;branch=z9hG4bK2b397eb16b1135e8c1be0f05eeae552d-1 f: <sip:1603870xxxx@207.138.151.33:5060;GXcomtechtollgold=GXCOMTECHTOLLGOLD-n3u3u5j4d1eo2>;tag=466343c3394a41487813f5d29e148a71 t: <sip:1877658xxxx@209.190.245.xxx:5060> i: 26739504b2880926d2afff419f370676-49ef1fd9@209.190.245.xxx CSeq: 41511 INVITE Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,REGISTER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed m: <sip:+1603870xxxx-GXcomtechtollgold-n3u3u5j4d1eo2@207.138.151.33:5060;transport=udp> k: timer x: 64800 Min-SE: 64800 l: 292 Content-Disposition: session; handling=required c: application/sdp v=0 o=Sonus_UAC 78310 7831000 IN IP4 207.138.151.38 s=SIP Media Capabilities c=IN IP4 207.138.151.38 t=0 0 m=audio 12714 RTP/AVP 18 0 8 100 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:100 telephone-event/8000 a=fmtp:100 0-15 a=sendrecv a=maxptime:20 [9] 2009/04/22 09:47:37: UDP: Opening socket on 0.0.0.0:50876 [9] 2009/04/22 09:47:37: UDP: Opening socket on 0.0.0.0:50877 [5] 2009/04/22 09:47:37: Identify trunk (IP address/port and domain match) 4 [9] 2009/04/22 09:47:37: Resolve 142564: aaaa udp 207.138.151.33 5060 [9] 2009/04/22 09:47:37: Resolve 142564: a udp 207.138.151.33 5060 [9] 2009/04/22 09:47:37: Resolve 142564: udp 207.138.151.33 5060 [9] 2009/04/22 09:47:37: SIP Tx udp:207.138.151.33:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 207.138.151.33:5060;branch=z9hG4bK2b397eb16b1135e8c1be0f05eeae552d-1 From: <sip:1603870xxxx@207.138.151.33:5060;GXcomtechtollgold=GXCOMTECHTOLLGOLD-n3u3u5j4d1eo2>;tag=466343c3394a41487813f5d29e148a71 To: <sip:1877658xxxx@209.190.245.xxx:5060>;tag=813034ca74 Call-ID: 26739504b2880926d2afff419f370676-49ef1fd9@209.190.245.xxx CSeq: 41511 INVITE Content-Length: 0 [6] 2009/04/22 09:47:37: Sending RTP for 26739504b2880926d2afff419f370676-49ef1fd9@209.190.245.xxx#813034ca74 to 207.138.151.38:12714 [5] 2009/04/22 09:47:37: Trunk ITSP (not global) sends call to account 90 in domain localhost [7] 2009/04/22 09:47:37: Attendant: Set language to first language en [8] 2009/04/22 09:47:37: Play recordings/att11.wav space20 [9] 2009/04/22 09:47:37: Resolve 142565: udp 209.190.198.110 34766 [6] 2009/04/22 09:47:37: send codec=pcmu/8000 [9] 2009/04/22 09:47:37: Resolve 142566: aaaa udp 207.138.151.33 5060 [9] 2009/04/22 09:47:37: Resolve 142566: a udp 207.138.151.33 5060 [9] 2009/04/22 09:47:37: Resolve 142566: udp 207.138.151.33 5060 [9] 2009/04/22 09:47:37: SIP Tx udp:207.138.151.33:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 207.138.151.33:5060;branch=z9hG4bK2b397eb16b1135e8c1be0f05eeae552d-1 From: <sip:1603870xxxx@207.138.151.33:5060;GXcomtechtollgold=GXCOMTECHTOLLGOLD-n3u3u5j4d1eo2>;tag=466343c3394a41487813f5d29e148a71 To: <sip:1877658xxxx@209.190.245.xxx:5060>;tag=813034ca74 Call-ID: 26739504b2880926d2afff419f370676-49ef1fd9@209.190.245.xxx CSeq: 41511 INVITE Contact: <sip:1877658xxxx@209.190.245.xxx:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbx/3.3.1.3177 Content-Type: application/sdp Content-Length: 230 v=0 o=- 312390002 312390002 IN IP4 209.190.245.xxx s=- c=IN IP4 209.190.245.xxx t=0 0 m=audio 50876 RTP/AVP 0 8 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv [6] 2009/04/22 09:47:37: send codec=pcmu/8000 [9] 2009/04/22 09:47:37: Resolve 142567: aaaa udp 207.138.151.33 5060 [9] 2009/04/22 09:47:37: Resolve 142567: a udp 207.138.151.33 5060 [9] 2009/04/22 09:47:37: Resolve 142567: udp 207.138.151.33 5060 [9] 2009/04/22 09:47:37: SIP Tx udp:207.138.151.33:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 207.138.151.33:5060;branch=z9hG4bK2b397eb16b1135e8c1be0f05eeae552d-1 From: <sip:1603870xxxx@207.138.151.33:5060;GXcomtechtollgold=GXCOMTECHTOLLGOLD-n3u3u5j4d1eo2>;tag=466343c3394a41487813f5d29e148a71 To: <sip:1877658xxxx@209.190.245.xxx:5060>;tag=813034ca74 Call-ID: 26739504b2880926d2afff419f370676-49ef1fd9@209.190.245.xxx CSeq: 41511 INVITE Contact: <sip:1877658xxxx@209.190.245.xxx:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbx/3.3.1.3177 Content-Type: application/sdp Content-Length: 230 v=0 o=- 312390002 312390002 IN IP4 209.190.245.xxx s=- c=IN IP4 209.190.245.xxx t=0 0 m=audio 50876 RTP/AVP 0 8 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv [9] 2009/04/22 09:47:37: SIP Rx udp:207.138.151.33:5060: ACK sip:1877658xxxx@209.190.245.xxx:5060;transport=udp SIP/2.0 v: SIP/2.0/UDP 207.138.151.33:5060;branch=z9hG4bK7e66eb355ab66842cfe2f0e95d8b02d9-1 f: <sip:1603870xxxx@207.138.151.33:5060;GXcomtechtollgold=GXCOMTECHTOLLGOLD-n3u3u5j4d1eo2>;tag=466343c3394a41487813f5d29e148a71 t: <sip:1877658xxxx@209.190.245.xxx:5060>;tag=813034ca74 i: 26739504b2880926d2afff419f370676-49ef1fd9@209.190.245.xxx CSeq: 41511 ACK Max-Forwards: 70 l: 0 [9] 2009/04/22 09:47:37: SIP Rx udp:207.138.151.33:5060: INVITE sip:1877658xxxx@209.190.245.xxx:5060;transport=udp SIP/2.0 v: SIP/2.0/UDP 207.138.151.33:5060;branch=z9hG4bK8789d802947ade0d00ef54990918c76a-1 f: <sip:1603870xxxx@207.138.151.33:5060;GXcomtechtollgold=GXCOMTECHTOLLGOLD-n3u3u5j4d1eo2>;tag=466343c3394a41487813f5d29e148a71 t: <sip:1877658xxxx@209.190.245.xxx:5060>;tag=813034ca74 i: 26739504b2880926d2afff419f370676-49ef1fd9@209.190.245.xxx CSeq: 41512 INVITE Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,REGISTER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed m: <sip:+1603870xxxx-GXcomtechtollgold-n3u3u5j4d1eo2@207.138.151.33:5060;transport=udp> k: timer x: 64800;refresher=uac Min-SE: 64800 l: 186 Content-Disposition: session; handling=required c: application/sdp v=0 o=Sonus_UAC 78310 7831001 IN IP4 207.138.151.38 s=SIP Media Capabilities c=IN IP4 207.138.151.38 t=0 0 m=audio 12714 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=sendrecv a=maxptime:20 [6] 2009/04/22 09:47:37: send codec=pcmu/8000 [9] 2009/04/22 09:47:37: Resolve 142568: aaaa udp 207.138.151.33 5060 [9] 2009/04/22 09:47:37: Resolve 142568: a udp 207.138.151.33 5060 [9] 2009/04/22 09:47:37: Resolve 142568: udp 207.138.151.33 5060 [9] 2009/04/22 09:47:37: SIP Tx udp:207.138.151.33:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 207.138.151.33:5060;branch=z9hG4bK8789d802947ade0d00ef54990918c76a-1 From: <sip:1603870xxxx@207.138.151.33:5060;GXcomtechtollgold=GXCOMTECHTOLLGOLD-n3u3u5j4d1eo2>;tag=466343c3394a41487813f5d29e148a71 To: <sip:1877658xxxx@209.190.245.xxx:5060>;tag=813034ca74 Call-ID: 26739504b2880926d2afff419f370676-49ef1fd9@209.190.245.xxx CSeq: 41512 INVITE Contact: <sip:1877658xxxx@209.190.245.xxx:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbx/3.3.1.3177 Content-Type: application/sdp Content-Length: 150 v=0 o=- 312390002 312390002 IN IP4 209.190.245.xxx s=- c=IN IP4 209.190.245.xxx t=0 0 m=audio 50876 RTP/AVP 0 a=rtpmap:0 pcmu/8000 a=sendrecv [6] 2009/04/22 09:47:37: Call hold from trunk [9] 2009/04/22 09:47:38: SIP Rx udp:207.138.151.33:5060: ACK sip:1877658xxxx@209.190.245.xxx:5060;transport=udp SIP/2.0 v: SIP/2.0/UDP 207.138.151.33:5060;branch=z9hG4bKc271aed5a61a5ea57eddb2122c8d680b-1 f: <sip:1603870xxxx@207.138.151.33:5060;GXcomtechtollgold=GXCOMTECHTOLLGOLD-n3u3u5j4d1eo2>;tag=466343c3394a41487813f5d29e148a71 t: <sip:1877658xxxx@209.190.245.xxx:5060>;tag=813034ca74 i: 26739504b2880926d2afff419f370676-49ef1fd9@209.190.245.xxx CSeq: 41512 ACK Max-Forwards: 70 l: 0 [9] 2009/04/22 09:47:39: Resolve 142569: udp 200.105.211.92 40930 [9] 2009/04/22 09:47:40: Resolve 142570: udp 209.190.198.110 35920 [9] 2009/04/22 09:47:40: Resolve 142571: udp 200.105.211.92 8254 [9] 2009/04/22 09:47:41: Resolve 142572: udp 209.190.198.110 34766 [9] 2009/04/22 09:47:41: Resolve 142573: udp 209.190.198.110 33742 [9] 2009/04/22 09:47:41: Resolve 142574: udp 209.190.198.110 33742 [9] 2009/04/22 09:47:42: SIP Rx udp:207.138.151.33:5060: BYE sip:1877658xxxx@209.190.245.xxx:5060;transport=udp SIP/2.0 v: SIP/2.0/UDP 207.138.151.33:5060;branch=z9hG4bK12bf9d14dafeee9ef90293720ae76780-1 f: <sip:1603870xxxx@207.138.151.33:5060;GXcomtechtollgold=GXCOMTECHTOLLGOLD-n3u3u5j4d1eo2>;tag=466343c3394a41487813f5d29e148a71 t: <sip:1877658xxxx@209.190.245.xxx:5060>;tag=813034ca74 i: 26739504b2880926d2afff419f370676-49ef1fd9@209.190.245.xxx CSeq: 41513 BYE Max-Forwards: 70 l: 0 [9] 2009/04/22 09:47:42: Resolve 142575: aaaa udp 207.138.151.33 5060 [9] 2009/04/22 09:47:42: Resolve 142575: a udp 207.138.151.33 5060 [9] 2009/04/22 09:47:42: Resolve 142575: udp 207.138.151.33 5060 [9] 2009/04/22 09:47:42: SIP Tx udp:207.138.151.33:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 207.138.151.33:5060;branch=z9hG4bK12bf9d14dafeee9ef90293720ae76780-1 From: <sip:1603870xxxx@207.138.151.33:5060;GXcomtechtollgold=GXCOMTECHTOLLGOLD-n3u3u5j4d1eo2>;tag=466343c3394a41487813f5d29e148a71 To: <sip:1877658xxxx@209.190.245.xxx:5060>;tag=813034ca74 Call-ID: 26739504b2880926d2afff419f370676-49ef1fd9@209.190.245.xxx CSeq: 41513 BYE Contact: <sip:1877658xxxx@209.190.245.xxx:5060;transport=udp> User-Agent: pbx/3.3.1.3177 RTP-RxStat: Dur=4,Pkt=207,Oct=35604,Underun=0 RTP-TxStat: Dur=4,Pkt=210,Oct=36120 Content-Length: 0

That's a very good question. It's a carrier I think you guys work/worked with closely, Prescient Worldwide. Is there a way to force inband on all calls, even if 2833 is offered?

A VOIP provider I am working with requires using inband dtmf if 711 is used as the codec, but it doesn't work because they are offering RFC2833 DTMF. Is there any way around this? They suggested switching to 729, but my license doesn't have 729 enabled. What is the best solution?

There is only one registration on both extensions. I think it is that the destination of the intercom call can only have 1 registration for it to work with auto answer and it doesn't matter if the source has multiple. At least that's how it works with Snom phones.

It does seem to be just with this phone. In band is doing the opposite. Calls on the FXO ports transmit DTMF, calls to other accounts on the system DTMF doesn't work.

I will try out 3.2 as well. INVITE sip:4797@demo.bizfon.com:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.0.0.76;branch=z9hG4bK5138e8d93A80051A From: "Demo 4796" <sip:4796@demo.bizfon.com>;tag=CAE127CF-406D3BBE To: <sip:4797@demo.bizfon.com;user=phone> CSeq: 1 INVITE Call-ID: 3cd348eb-71cc9b3d-9137a54c@10.0.0.76 Contact: <sip:4796@10.0.0.76> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.2.2.0084 Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 141 v=0 o=- 1167610069 1167610069 IN IP4 10.0.0.76 s=Polycom IP Phone c=IN IP4 10.0.0.76 t=0 0 m=message 5060 sip sip:4796@demo.bizfon.com [9] 2009/03/16 12:15:24: UDP: Opening socket on port 61032 [9] 2009/03/16 12:15:24: UDP: Opening socket on port 61033 [8] 2009/03/16 12:15:24: Using outbound proxy sip:209.190.198.110:59342;transport=udp because UDP packet source did not match the via header [9] 2009/03/16 12:15:24: Resolve 385703: udp 209.190.198.110 59342 [9] 2009/03/16 12:15:24: SIP Tx udp:209.190.198.110:59342: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.0.76;branch=z9hG4bK5138e8d93A80051A;rport=59342;received=209.190.198.110 From: "Demo 4796" <sip:4796@demo.bizfon.com>;tag=CAE127CF-406D3BBE To: <sip:4797@demo.bizfon.com;user=phone>;tag=f35dc805b0 Call-ID: 3cd348eb-71cc9b3d-9137a54c@10.0.0.76 CSeq: 1 INVITE Content-Length: 0 [9] 2009/03/16 12:15:24: Resolve 385704: udp 209.190.198.110 59342 [9] 2009/03/16 12:15:24: SIP Tx udp:209.190.198.110:59342: SIP/2.0 401 Authentication Required Via: SIP/2.0/UDP 10.0.0.76;branch=z9hG4bK5138e8d93A80051A;rport=59342;received=209.190.198.110 From: "Demo 4796" <sip:4796@demo.bizfon.com>;tag=CAE127CF-406D3BBE To: <sip:4797@demo.bizfon.com;user=phone>;tag=f35dc805b0 Call-ID: 3cd348eb-71cc9b3d-9137a54c@10.0.0.76 CSeq: 1 INVITE User-Agent: Bizfon-PBX/3.2.0.3144 WWW-Authenticate: Digest realm="demo.bizfon.com",nonce="4820c463cc377d4937e0ef2956a6cff2",domain="sip:4797@demo.bizfon.com:5060;user=phone",algorithm=MD5 Content-Length: 0 [9] 2009/03/16 12:15:24: SIP Rx udp:209.190.198.110:59342: ACK sip:4797@demo.bizfon.com:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.76;branch=z9hG4bK5138e8d93A80051A From: "Demo 4796" <sip:4796@demo.bizfon.com>;tag=CAE127CF-406D3BBE To: <sip:4797@demo.bizfon.com;user=phone>;tag=f35dc805b0 CSeq: 1 ACK Call-ID: 3cd348eb-71cc9b3d-9137a54c@10.0.0.76 Contact: <sip:4796@10.0.0.76> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.2.2.0084 Max-Forwards: 70 Content-Length: 0 [9] 2009/03/16 12:15:24: SIP Rx udp:209.190.198.110:59342: INVITE sip:4797@demo.bizfon.com:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.0.0.76;branch=z9hG4bKe61636a1759E6B62 From: "Demo 4796" <sip:4796@demo.bizfon.com>;tag=CAE127CF-406D3BBE To: <sip:4797@demo.bizfon.com;user=phone> CSeq: 2 INVITE Call-ID: 3cd348eb-71cc9b3d-9137a54c@10.0.0.76 Contact: <sip:4796@10.0.0.76> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.2.2.0084 Supported: 100rel,replaces Allow-Events: talk,hold,conference Authorization: Digest username="4796", realm="demo.bizfon.com", nonce="4820c463cc377d4937e0ef2956a6cff2", uri="sip:4797@demo.bizfon.com:5060;user=phone", response="31a5144e9bca2dd69e3556d64e4831f0", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 141 v=0 o=- 1167610069 1167610069 IN IP4 10.0.0.76 s=Polycom IP Phone c=IN IP4 10.0.0.76 t=0 0 m=message 5060 sip sip:4796@demo.bizfon.com [8] 2009/03/16 12:15:24: Tagging request with existing tag [9] 2009/03/16 12:15:24: Resolve 385705: udp 209.190.198.110 59342 [9] 2009/03/16 12:15:24: SIP Tx udp:209.190.198.110:59342: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.0.76;branch=z9hG4bKe61636a1759E6B62;rport=59342;received=209.190.198.110 From: "Demo 4796" <sip:4796@demo.bizfon.com>;tag=CAE127CF-406D3BBE To: <sip:4797@demo.bizfon.com;user=phone>;tag=f35dc805b0 Call-ID: 3cd348eb-71cc9b3d-9137a54c@10.0.0.76 CSeq: 2 INVITE Content-Length: 0 [9] 2009/03/16 12:15:24: Using outbound proxy sip:209.190.198.110:59342;transport=udp because of flow-label [9] 2009/03/16 12:15:24: Resolve 385706: url sip:209.190.198.110:59342;transport=udp [9] 2009/03/16 12:15:24: Resolve 385706: a udp 209.190.198.110 59342 [9] 2009/03/16 12:15:24: Resolve 385706: udp 209.190.198.110 59342 [9] 2009/03/16 12:15:24: UDP: Opening socket on port 53048 [9] 2009/03/16 12:15:24: UDP: Opening socket on port 53049 [9] 2009/03/16 12:15:24: Using outbound proxy sip:209.190.198.110:3022;transport=udp because of flow-label [9] 2009/03/16 12:15:24: Resolve 385707: url sip:209.190.198.110:3022;transport=udp [9] 2009/03/16 12:15:24: Resolve 385707: a udp 209.190.198.110 3022 [9] 2009/03/16 12:15:24: Resolve 385707: udp 209.190.198.110 3022 [9] 2009/03/16 12:15:24: SIP Tx udp:209.190.198.110:3022: INVITE sip:4797@10.0.0.78 SIP/2.0 Via: SIP/2.0/UDP 155.212.72.124:5060;branch=z9hG4bK-d9ca6c8ebf13571188394ea99520b202;rport From: "Demo 4796" <sip:4796@demo.bizfon.com>;tag=64758 To: "Demo 4797" <sip:4797@demo.bizfon.com> Call-ID: fdb902a7@pbx CSeq: 15263 INVITE Max-Forwards: 70 Contact: <sip:4797@155.212.72.124:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: Bizfon-PBX/3.2.0.3144 Alert-Info: Internal Call-Info: <http://155.212.72.124/images/extensions61.bmp>;purpose=icon Content-Type: application/sdp Content-Length: 270 v=0 o=- 63632 63632 IN IP4 155.212.72.124 s=- c=IN IP4 155.212.72.124 t=0 0 m=audio 53048 RTP/AVP 0 8 2 3 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/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 [9] 2009/03/16 12:15:24: Using outbound proxy sip:209.190.198.110:59342;transport=udp because of flow-label [9] 2009/03/16 12:15:24: Resolve 385708: url sip:209.190.198.110:59342;transport=udp [9] 2009/03/16 12:15:24: Resolve 385708: a udp 209.190.198.110 59342 [9] 2009/03/16 12:15:24: Resolve 385708: udp 209.190.198.110 59342 [9] 2009/03/16 12:15:24: Resolve 385709: udp 209.190.198.110 59342 [9] 2009/03/16 12:15:24: SIP Tx udp:209.190.198.110:59342: SIP/2.0 415 Unsupported Media Type Via: SIP/2.0/UDP 10.0.0.76;branch=z9hG4bKe61636a1759E6B62;rport=59342;received=209.190.198.110 From: "Demo 4796" <sip:4796@demo.bizfon.com>;tag=CAE127CF-406D3BBE To: <sip:4797@demo.bizfon.com;user=phone>;tag=f35dc805b0 Call-ID: 3cd348eb-71cc9b3d-9137a54c@10.0.0.76 CSeq: 2 INVITE Contact: <sip:4796@155.212.72.124:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: Bizfon-PBX/3.2.0.3144 Content-Length: 0 [9] 2009/03/16 12:15:24: Resolve 385710: udp 209.190.198.110 3022 [9] 2009/03/16 12:15:24: SIP Tx udp:209.190.198.110:3022: CANCEL sip:4797@10.0.0.78 SIP/2.0 Via: SIP/2.0/UDP 155.212.72.124:5060;branch=z9hG4bK-d9ca6c8ebf13571188394ea99520b202;rport From: "Demo 4796" <sip:4796@demo.bizfon.com>;tag=64758 To: "Demo 4797" <sip:4797@demo.bizfon.com> Call-ID: fdb902a7@pbx CSeq: 15263 CANCEL Max-Forwards: 70 Content-Length: 0 [9] 2009/03/16 12:15:24: Using outbound proxy sip:209.190.198.110:59342;transport=udp because of flow-label [9] 2009/03/16 12:15:24: Resolve 385711: url sip:209.190.198.110:59342;transport=udp [9] 2009/03/16 12:15:24: Resolve 385711: a udp 209.190.198.110 59342 [9] 2009/03/16 12:15:24: Resolve 385711: udp 209.190.198.110 59342 [9] 2009/03/16 12:15:24: Resolve 385712: udp 209.190.198.110 59342 [9] 2009/03/16 12:15:24: SIP Tx udp:209.190.198.110:59342: SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 10.0.0.76;branch=z9hG4bKe61636a1759E6B62;rport=59342;received=209.190.198.110 From: "Demo 4796" <sip:4796@demo.bizfon.com>;tag=CAE127CF-406D3BBE To: <sip:4797@demo.bizfon.com;user=phone>;tag=f35dc805b0 Call-ID: 3cd348eb-71cc9b3d-9137a54c@10.0.0.76 CSeq: 2 INVITE Contact: <sip:4796@155.212.72.124:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: Bizfon-PBX/3.2.0.3144 Content-Length: 0 [9] 2009/03/16 12:15:24: Using outbound proxy sip:209.190.198.110:59342;transport=udp because of flow-label [9] 2009/03/16 12:15:24: Resolve 385713: url sip:209.190.198.110:59342;transport=udp [9] 2009/03/16 12:15:24: Resolve 385713: a udp 209.190.198.110 59342 [9] 2009/03/16 12:15:24: Resolve 385713: udp 209.190.198.110 59342 [9] 2009/03/16 12:15:25: SIP Rx udp:209.190.198.110:59342: ACK sip:4797@demo.bizfon.com:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.76;branch=z9hG4bKe61636a1759E6B62 From: "Demo 4796" <sip:4796@demo.bizfon.com>;tag=CAE127CF-406D3BBE To: <sip:4797@demo.bizfon.com;user=phone>;tag=f35dc805b0 CSeq: 2 ACK Call-ID: 3cd348eb-71cc9b3d-9137a54c@10.0.0.76 Contact: <sip:4796@10.0.0.76> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.2.2.0084 Authorization: Digest username="4796", realm="demo.bizfon.com", nonce="4820c463cc377d4937e0ef2956a6cff2", uri="sip:4797@demo.bizfon.com:5060;user=phone", response="31a5144e9bca2dd69e3556d64e4831f0", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 [7] 2009/03/16 12:15:25: Other Ports: 1 [7] 2009/03/16 12:15:25: Call Port: fdb902a7@pbx#64758 [9] 2009/03/16 12:15:25: SIP Rx udp:209.190.198.110:3022: SIP/2.0 100 Trying Via: SIP/2.0/UDP 155.212.72.124:5060;branch=z9hG4bK-d9ca6c8ebf13571188394ea99520b202;rport From: "Demo 4796" <sip:4796@demo.bizfon.com>;tag=64758 To: "Demo 4797" <sip:4797@demo.bizfon.com>;tag=EC675947-418DD4FA CSeq: 15263 INVITE Call-ID: fdb902a7@pbx Contact: <sip:4797@10.0.0.78> User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.2.2.0084 Content-Length: 0 [9] 2009/03/16 12:15:25: SIP Rx udp:209.190.198.110:59342: ACK sip:4797@demo.bizfon.com:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.0.0.76;branch=z9hG4bKe61636a1759E6B62 From: "Demo 4796" <sip:4796@demo.bizfon.com>;tag=CAE127CF-406D3BBE To: <sip:4797@demo.bizfon.com;user=phone>;tag=f35dc805b0 CSeq: 2 ACK Call-ID: 3cd348eb-71cc9b3d-9137a54c@10.0.0.76 Contact: <sip:4796@10.0.0.76> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.2.2.0084 Authorization: Digest username="4796", realm="demo.bizfon.com", nonce="4820c463cc377d4937e0ef2956a6cff2", uri="sip:4797@demo.bizfon.com:5060;user=phone", response="7a9958a0b0706fe76428a1d18119b39c", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 [9] 2009/03/16 12:15:25: Message repetition, packet dropped [9] 2009/03/16 12:15:25: SIP Rx udp:209.190.198.110:3022: SIP/2.0 200 OK Via: SIP/2.0/UDP 155.212.72.124:5060;branch=z9hG4bK-d9ca6c8ebf13571188394ea99520b202;rport From: "Demo 4796" <sip:4796@demo.bizfon.com>;tag=64758 To: "Demo 4797" <sip:4797@demo.bizfon.com> CSeq: 15263 CANCEL Call-ID: fdb902a7@pbx Contact: <sip:4797@10.0.0.78> User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.2.2.0084 Content-Length: 0 [7] 2009/03/16 12:15:25: Call fdb902a7@pbx#64758: Clear last request [9] 2009/03/16 12:15:25: SIP Rx udp:209.190.198.110:3022: SIP/2.0 487 Request Cancelled Via: SIP/2.0/UDP 155.212.72.124:5060;branch=z9hG4bK-d9ca6c8ebf13571188394ea99520b202;rport From: "Demo 4796" <sip:4796@demo.bizfon.com>;tag=64758 To: "Demo 4797" <sip:4797@demo.bizfon.com>;tag=EC675947-418DD4FA CSeq: 15263 INVITE Call-ID: fdb902a7@pbx Contact: <sip:4797@10.0.0.78> User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.2.2.0084 Content-Length: 0 [7] 2009/03/16 12:15:25: Call fdb902a7@pbx#64758: Clear last INVITE [9] 2009/03/16 12:15:25: Resolve 385714: url sip:209.190.198.110:3022;transport=udp [9] 2009/03/16 12:15:25: Resolve 385714: a udp 209.190.198.110 3022 [9] 2009/03/16 12:15:25: Resolve 385714: udp 209.190.198.110 3022 [9] 2009/03/16 12:15:25: SIP Tx udp:209.190.198.110:3022: ACK sip:4797@10.0.0.78 SIP/2.0 Via: SIP/2.0/UDP 155.212.72.124:5060;branch=z9hG4bK-d9ca6c8ebf13571188394ea99520b202;rport From: "Demo 4796" <sip:4796@demo.bizfon.com>;tag=64758 To: "Demo 4797" <sip:4797@demo.bizfon.com>;tag=EC675947-418DD4FA Call-ID: fdb902a7@pbx CSeq: 15263 ACK Max-Forwards: 70 Contact: <sip:4797@155.212.72.124:5060;transport=udp> Content-Length: 0

I am trying to send an IM from one extension to another. Both are Polycom's running 2.2.2.0084. When I send an IM one of the phones, it rings the phone for a split second and shows up as a missed call not as an IM. If I send the IM from the pbx, it gets delivered properly to the other phone as an IM. In the logs it looks like the sending phone is trying to call the destination with an invite instead of just sending the message. Is this something that is supported through the pbx? Should I be using a different version of software on the phones other than the version that is listed on the wiki?

I noticed this too, I have the mail options on the cs410 image I downloaded but not in the OSX or Windows builds. All were downloaded from the pbxnsip software page. The Win and OSX versions show 3.2.0.3144 and the 410 shows 3.2.0.3143.