pbx support

Anything changed in the network in the last 2 days for it to stop working? BTW, I did not see it switching to G729 from the trace.

Maybe this is what you are looking for. polycom_phone.xml

Since you are using 3.3.0.3165 of the PBX, you need to set 'just' the IP address of the PBX under snom 370's "Settings URL" (no http or the full path). Also, put the MAC address of the phone under "Registration" tab->"Bind to MAC Address" on PBX. Reset the phone.

We are looking into it. We get emails when the ticket is opened

Can you please send the SIP log from the PBX to support@pbxnsip.com

Did you try the lower values? I have seen value of 80 working some of the installations.

Could this be due to the multiple UAs on the extension?

Did you check this wiki page? http://wiki.pbxnsip.com/index.php/Dialog_State_Notifications

If you start the wireshark capturing before you make the call and stop it after you hung up, then it will record the signaling packets too. Then it is easier do some analysis.

You can dial *98+AA (Ex: If the AA is at 70, then dial *9870) from any extension and when prompted, record the whole message that you would like to play as the welcome greeting. PBX will automatically assign this as AA's welcome greeting. That means, if you pickup a phone and dial the AA now, you will hear the recorded message.

The attached pcap trace shows pretty much only 1 way audio packets (192.168.10.200 to .63). Also, since there is the SIP signaling packets (INVITE, BYE etc) are not captured in this trace, wireshark has difficult analyzing the call.

Since it is random and long distance only, I would attribute the issue to the bandwidth/QoS of the network (end-to-end). Maybe it is good to try some QoS tools http://wiki.pbxnsip.com/index.php/Troubles...blems#QoS_Tools.

Looks like you missed this line on the wiki "The above information is gathered on 3.3.0.3158 version" In any case, the latest software (.3165) contains this inforation.

If you are using the latest software 3.3.0.3165 - 1. On the PBX, please set the MAC address of the phone under the registration tab. 2. On the snom phone, set the PBX's IP address (Ex: 192.168.1.100) under "Settings URL". 3. Reboot/Reset the phone when prompted.

Is this one instance or it is happening to all long distance calls? It would be that the PC running x-lite is busy doing something after 2 minutes into the call.

Here is the latest Mac OS build http://pbxnsip.com/protect/pbx-darwin9.0-3.3.0.3163.zip. Please set, on the PBX, the MAC address of the phone under the registration tab. Also, on the snom phone's "Settings URL", set the PBX's IP address(no need to put 'http' in front or the port info or .../provisioning/snom320_macaddr.htm)

While we are looking at this issue in detail (because after the previous software change, we have got the response from other customers that the problem has been fixed), can you send the latet wireshark trace (SIP/RTP) to support@pbxnsip.com?

Ok. PBX is not supposed to do that. It is supposed to do G711 to the phone and g729 to the trunk, if the "lock codec.. " is set. If I understand correctly, you still have 1-way audio. Is that right?

All you have to do is set the SOAP CDR Url to "cdr:192.168.1.2:10000", where 192.168.1.2:10000 is your SOAP server's ip:port. PBX will send the SOAP CDRs in the format explained in http://wiki.pbxnsip.com/index.php/Simple_CDR_Format and http://wiki.pbxnsip.com/index.php/Processi...DR_from_the_PBX

Try http://www.pbxnsip.com/protect/pbxctrl-3.3.0.3160.exe

There is a possibility that one leg of the call is G711 and the other is G729. But incoming and outgoing of each leg will be of the same codec.

CDRs are PUSHed from the PBX and not pulled the way you wanted. What you can do though, set the SOAP address on the PBX(probably already done) and receive all the CDRs and then filter it in your program.

Oh I see! Looks like multiple "Entry" is not supported. We will fix it in the next release.

What do you mean by Unfortunately, PBXNSIP is still trying to change the codecs during the early media?. As long as the UA(phone) and the PBX are using the same codec, you should not have any issue. Wireshark RTP trace will tell you the exact codec that is used on both direction.

Currently, it is designed to create 1 dial plan at a time. What you can do though, in your program, instead of sending the multiple dialplans at the same time, you can send one at a time in a loop.