Vodia PBX

We have the idea on the table to organize the file system according to the domains, so that a backup of a domain would also be a backup of the directory. But so far the only resutlt is that it will not be "trivial". Of course, you can also just restart the PBX process if you want it to re-read everything. Brute force. The other idea that we had was using email to push content into the PBX. Again, just idea stage. This should be very simple for many applications.

Well, try to set a country code (55) and set the alias for the user to 01118887771476. Then the PBX should be able to identify the user.

No matter what, you will always have to expose the password this way or another. If you want to keep it safe, you can still use https transport layer. Actually, the point here is to have a simple way to automatically set something through the web interface of the PBX. I believe the following will do the job: curl 'http://10.10.10.3/post.htm?user=admin&pass=password&file=reg_settings.htm&save=save&inband_decoding=true' In this example, you go to the web page reg_settings.htm and set the setting "inband_decoding" to "true". In order to set something in a domain, you muse include the domain in the style "domain=abc.com". If you want to set something for an account, then you should include the parameter in the style "account=40". For example: curl 'http://10.10.10.3/post.htm?user=admin&pass=password&file=dom_ext.htm&editaccount=save&domain=localhost&account=40&call_redial=123456' This feature will be available in the next head build.

You mean like SIGHUP? I am not a big fan of that... And if you can send a SOAP message then you can also set the table entries with a SOAP message... Plus the format of the file in the file system may change during updates... What is the use case here? Do you want to change the settings of an account?

Thanks for sharing this with us!

Yes, that is a tricky issue. By default, the PBX will just look up the OS routing table and then find out what local identity the OS would use for sending out the packet. If you choose to use the private gateway address as the default route it will advertize the private IP address. In TCP that is no problem because everything is connection oriented anyway. But for media, the PBX must offer something with a public IP. I think this is explained well in the following article: http://kiwi.pbxnsip.com/index.php/Office_w...ic_IP_addresses

Yes, this is a known problem. We are thinking about a way so that you can automatically authenticate without the need to go through a login form. I am thinking about something like http://pbx/auto.htm?page=reg_settings.htm&...e&form2=123

I don't understand the question... Logging in by redirecting the extension???

The next major release will have total customizability (is that a existing word?). Until then, if you want to change the HTML you have to send us an email and then we can give you a snapshot of the page.

I guess this is a violation of their license agreement. Not sure if they can do anything against it.

We'll check.

Yea, that input field got forgotten! Next version will have it. You can also charge the domain with a SOAP call. I guess in the real world, thats the way to go, because it does not set the amount, it adds it (even if there are calls going on). You can use AddCredit as the SOAP Method for this. The arguments are Domain (e.g. "company.com") and Amount (e.g. "123.50"). If you use Extension (e.g. "40"), then only the extension will be charged.

We added a lot more now. For example, you can transfer a call or accept a call (if the connected device supports the "talk" event). The Wiki is pretty old on CSTA. We need to update the documentation there.

Agreed. This was a bug in the PBX. A WAC state update should also update the session duration.

There was a firmware which had problems with G.729A, where you would hear the audio only after 8 seconds. This was last year, and there was a firmware update available that fixed the problem. Not sure if the latest firmware introduced a new problem. In any case, make sure that you don't use STUN on the phone. STUN is usually an excellent source for such problems. If the problem also exists with other phone, make sure you go through the checklist at http://kiwi.pbxnsip.com/index.php/One-way_Audio.

Well, that would be extremly buggy. In SIP you can essentially write "hello_have_a_nice_day" in the user part of the contant and it should still work. What about the Alert-Info? Maybe the device gets confused with the Alert-Info because it cannot render the requested melody. Try to select a different tone (or none) in the hunt group. Anyway, maybe you should open a ticket with Panasonic and ask them if they support Alert-Info or they at least ignore it (as they should by default).

Well usually when the PBX sends a 2xx code, it should reset the blacklist counter. Seems sometimes you have too many unanswered calls open. White listing definitevely solves the problem.

4.0.1.3499 is the current stable release, see http://www.pbxnsip.com/download-software/software.php. I believe the version that you are running is not the latest with the certificates.

Maybe try this one: http://provisioning.snom.com/download/fw/snom870-8.4.16-SIP-r.bin

In Windows XP, you could create an internal loop so that the mic input was the WAV output. That made it possible to use a standard MP3 player to create the music you want and loop it back to the PBX, or the pagmoh. Seems that option was removed in Windows 7 (probably to fight illegal copying of music). The workaround now is to use a physical analog cable to create such a loop. The pagmoh should work independently from the PBX service. It is just using standard RTP on the network side.

Sure. Not sure if Exchange 2007 differes from Exchange 2010 here. Anyway, we can run your config and see if we can reproduce the problem.

In version 4, things have changed a little bit. The certificate can now be linked to a specific domain (see RFC3546), therefore you can import the certificate as a "global" certificate or a "domain" certificate. The domain name in the certificate must match the name in the PBX; otherwise clients that chech the cert will reject it.

Can you zip/tar the working directory and provide it to us? Then we can run it here with the Exchange 2010.

Yes, this must be limited because otherwise the PBX could run out of virtual memory.

Okay, I tried this. "It works"... How do you get the 487 responses? I think that must be the difference in the setup. Do you have a long delay between the handset and the PBX when the call rolls over to the mailbox (delay to phone a lot longer than delay to the Exchange)? [6] 2010/06/28 21:55:39: Redirecting to external voicemail account 401 destination sip:8401@localhost [5] 2010/06/28 21:55:39: Dialplan "Default Dialplan": Match [email="8401@localhost"]8401@localhost[/email] to <sip:401@vmail.pbxnsip.com;user=phone> on trunk Exchange [5] 2010/06/28 21:55:39: Using <sip:420@localhost> as redirect source address [5] 2010/06/28 21:55:39: Charge user 401 for redirecting calls [5] 2010/06/28 21:55:39: SIP Tx tcp:10.10.10.212:5060: INVITE sip:401@vmail.pbxnsip.com;user=phone SIP/2.0 Via: SIP/2.0/TCP 10.10.10.3:1549;branch=z9hG4bK-15870d25271e8549b980287692949bca;rport From: <sip:401@vmail.pbxnsip.com;user=phone>;tag=38448 To: <sip:401@vmail.pbxnsip.com;user=phone> Call-ID: [email="bda2fdb5@pbx"]bda2fdb5@pbx[/email] CSeq: 16811 INVITE Max-Forwards: 70 Contact: <sip:401@10.10.10.3:1549;transport=tcp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/4.0.1.3499 Diversion: <tel:401>;reason=no-answer;screen=no;privacy=off P-Asserted-Identity: "401" <sip:401@vmail.pbxnsip.com;user=phone> Content-Type: application/sdp Content-Length: 261 v=0 o=- 50319 50319 IN IP4 10.10.10.3 s=- c=IN IP4 10.10.10.3 t=0 0 a=oa:offer m=audio 50552 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=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv [5] 2010/06/28 21:55:39: SIP Rx tcp:10.10.10.212:5060: SIP/2.0 100 Trying FROM: <sip:401@vmail.pbxnsip.com;user=phone>;tag=38448 TO: <sip:401@vmail.pbxnsip.com;user=phone> CSEQ: 16811 INVITE CALL-ID: [email="bda2fdb5@pbx"]bda2fdb5@pbx[/email] VIA: SIP/2.0/TCP 10.10.10.3:1549;branch=z9hG4bK-15870d25271e8549b980287692949bca;rport CONTENT-LENGTH: 0 [5] 2010/06/28 21:55:40: SIP Rx tcp:10.10.10.212:5060: SIP/2.0 302 Moved Temporarily FROM: <sip:401@vmail.pbxnsip.com;user=phone>;tag=38448 TO: <sip:401@vmail.pbxnsip.com;user=phone>;tag=f8d9f25460 CSEQ: 16811 INVITE CALL-ID: [email="bda2fdb5@pbx"]bda2fdb5@pbx[/email] VIA: SIP/2.0/TCP 10.10.10.3:1549;branch=z9hG4bK-15870d25271e8549b980287692949bca;rport CONTACT: <sip:401@vmail.pbxnsip.com:5065;user=phone;transport=Tcp> CONTENT-LENGTH: 0 SERVER: RTCC/3.1.0.0 Diversion: <tel:401>;reason=no-answer;screen=no;privacy=off [7] 2010/06/28 21:55:40: Call [email="bda2fdb5@pbx#38448"]bda2fdb5@pbx#38448[/email]: Clear last INVITE [5] 2010/06/28 21:55:40: SIP Tx tcp:10.10.10.212:5060: ACK sip:401@vmail.pbxnsip.com;user=phone SIP/2.0 Via: SIP/2.0/TCP 10.10.10.3:1549;branch=z9hG4bK-15870d25271e8549b980287692949bca;rport From: <sip:401@vmail.pbxnsip.com;user=phone>;tag=38448 To: <sip:401@vmail.pbxnsip.com;user=phone>;tag=f8d9f25460 Call-ID: [email="bda2fdb5@pbx"]bda2fdb5@pbx[/email] CSeq: 16811 ACK Max-Forwards: 70 Contact: <sip:401@10.10.10.3:1549;transport=tcp> P-Asserted-Identity: "401" <sip:401@vmail.pbxnsip.com;user=phone> Content-Length: 0 [5] 2010/06/28 21:55:40: Redirecting call [5] 2010/06/28 21:55:40: SIP Tx tcp:10.10.10.212:5065: INVITE sip:401@vmail.pbxnsip.com:5065;user=phone;transport=Tcp SIP/2.0 Via: SIP/2.0/TCP 10.10.10.3:1550;branch=z9hG4bK-c83658d05aac62fb15208b870f80fcfb;rport From: <sip:401@vmail.pbxnsip.com;user=phone>;tag=38448 To: <sip:401@vmail.pbxnsip.com;user=phone> Call-ID: [email="bda2fdb5@pbx"]bda2fdb5@pbx[/email] CSeq: 16812 INVITE Max-Forwards: 70 Contact: <sip:401@10.10.10.3:1550;transport=tcp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: pbxnsip-PBX/4.0.1.3499 Diversion: <tel:401>;reason=no-answer;screen=no;privacy=off P-Asserted-Identity: "401" <sip:401@vmail.pbxnsip.com;user=phone> Content-Type: application/sdp Content-Length: 261 v=0 o=- 50319 50319 IN IP4 10.10.10.3 s=- c=IN IP4 10.10.10.3 t=0 0 a=oa:offer m=audio 50552 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=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv [5] 2010/06/28 21:55:40: SIP Rx tcp:10.10.10.212:5065: SIP/2.0 100 Trying FROM: <sip:401@vmail.pbxnsip.com;user=phone>;tag=38448 TO: <sip:401@vmail.pbxnsip.com;user=phone> CSEQ: 16812 INVITE CALL-ID: [email="bda2fdb5@pbx"]bda2fdb5@pbx[/email] VIA: SIP/2.0/TCP 10.10.10.3:1550;branch=z9hG4bK-c83658d05aac62fb15208b870f80fcfb;rport CONTENT-LENGTH: 0 [6] 2010/06/28 21:55:40: Sending RTP for [email="bda2fdb5@pbx#38448"]bda2fdb5@pbx#38448[/email] to 10.10.10.212:38400 [5] 2010/06/28 21:55:40: SIP Rx tcp:10.10.10.212:5065: SIP/2.0 180 Ringing FROM: <sip:401@vmail.pbxnsip.com;user=phone>;tag=38448 TO: <sip:401@vmail.pbxnsip.com;user=phone>;epid=9D18C5EB27;tag=20e15cb42f CSEQ: 16812 INVITE CALL-ID: [email="bda2fdb5@pbx"]bda2fdb5@pbx[/email] VIA: SIP/2.0/TCP 10.10.10.3:1550;branch=z9hG4bK-c83658d05aac62fb15208b870f80fcfb;rport CONTENT-LENGTH: 0 SERVER: RTCC/3.1.0.0 [5] 2010/06/28 21:55:40: SIP Rx tcp:10.10.10.212:5065: SIP/2.0 200 OK FROM: <sip:401@vmail.pbxnsip.com;user=phone>;tag=38448 TO: <sip:401@vmail.pbxnsip.com;user=phone>;epid=9D18C5EB27;tag=20e15cb42f CSEQ: 16812 INVITE CALL-ID: [email="bda2fdb5@pbx"]bda2fdb5@pbx[/email] VIA: SIP/2.0/TCP 10.10.10.3:1550;branch=z9hG4bK-c83658d05aac62fb15208b870f80fcfb;rport CONTACT: <sip:vmail.pbxnsip.com:5065;transport=Tcp;maddr=10.10.10.212>;automata CONTENT-LENGTH: 195 CONTENT-TYPE: application/sdp ALLOW: UPDATE SERVER: RTCC/3.1.0.0 ALLOW: Ack, Cancel, Bye,Invite,Message,Info,Service,Options,BeNotify v=0 o=- 0 0 IN IP4 10.10.10.212 s=Microsoft Exchange Speech Engine c=IN IP4 10.10.10.212 t=0 0 m=audio 38400 RTP/AVP 0 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 [7] 2010/06/28 21:55:40: Call [email="bda2fdb5@pbx#38448"]bda2fdb5@pbx#38448[/email]: Clear last INVITE [7] 2010/06/28 21:55:40: Set packet length to 20 [5] 2010/06/28 21:55:40: SIP Tx tcp:10.10.10.212:5065: ACK sip:vmail.pbxnsip.com:5065;transport=Tcp;maddr=10.10.10.212 SIP/2.0 Via: SIP/2.0/TCP 10.10.10.3:1550;branch=z9hG4bK-ccd4e640010754363cde45f98ef1fa20;rport From: <sip:401@vmail.pbxnsip.com;user=phone>;tag=38448 To: <sip:401@vmail.pbxnsip.com;user=phone>;tag=20e15cb42f Call-ID: [email="bda2fdb5@pbx"]bda2fdb5@pbx[/email] CSeq: 16812 ACK Max-Forwards: 70 Contact: <sip:401@10.10.10.3:1550;transport=tcp> P-Asserted-Identity: "401" <sip:401@vmail.pbxnsip.com;user=phone> Content-Length: 0 [7] 2010/06/28 21:55:40: Determine pass-through mode after receiving response [7] 2010/06/28 21:55:41: [email="bda2fdb5@pbx#38448"]bda2fdb5@pbx#38448[/email]: RTP pass-through mode [7] 2010/06/28 21:55:41: N2U0NmYyNzYzZDhmY2VhMDNhNDllNTNjZWY3MGU2ZWY.#fe4445a310: RTP pass-through mode