Interconnecting two PBXnSIP

[Detlef]

I am impressed how easy it is to interconnect two PBXnSIP servers via two SIP Gateway trunks and a simple dialplan entry which forwards extensions 1xx to the first system and 2xx to the 2nd system. Everything seems to work inbetween those two systems.

 

Only problem I have is when you dial in with your cell phone over a PSTN gateway (Grandstream GXW410x) and select the outbound call option to dial an extension in the 2nd system it only rings but no audio is transfered either way. Those systems are connected via a simple VPN, no port blocking or anything - both systems can reach each other completely freely.

 

So I am stuck on how I would troubleshoot this problem and what do I need to look for?

[Vodia PBX]

It should be possible to trouble shoot the one way audio independently from the call redirection. You say that making a call through the VPN is no problem?

 

Does the VPN go up and down? The PBX has problems when the IP configuration changes. Going to the status web screen of the PBX re-reads the configuration, so if you want to refresh the internal tables for the IP config you can just go to that web page.

[Detlef]

Normally our VPN is pretty stable but still, it goes to Mexico so we have days where it goes up and down alot.

 

Internal calls from one PBXnSIP to the other works fine back and forth. Never had a problem with Audio.

 

The problem comes when I try to call in with my cell phone over a Grandstream GWX4108 gateway. I get the system menu which asks me if I want to place an outbound call - which I select. Entering the 3 digit extension of the PBXnSIP system in Mexico rings the correct phone over there but if someone picks up there is no audio in either way.

 

I dont believe it is the PBXnSIP, may something with redirection that the Grandstream doesnt like? The PBXnSIP internally over the VPN always works.

 

Detlef

[Vodia PBX]

Well, whenever an IP interface becomes unavailable the PBX has a problem because the internal routing tables don't fit any more. Is there any way you can stabilize that VPN connection on the PBX? For example, have another server doing the VPN and just send the traffic there (in a different IP segment)? Then the PBX IP config never changes and the PBX does not get confused.

[Detlef]

The VPN is not setup on the PBX itself. Its done with two identical DLink VPN router. The 2 PBXs computer have those DLink routers set as default gateway. So whatever goes from our IP space 192.168.104.x to 192.168.108.x or vise versa is send to the router. Same with the Grandstream Gateway, that one has also the DLink router set as default gateway. So I assume whatever happens to the VPN status does not affect the PBX or the GWX4108 gateway - those should just forward everything what goes out of our local IP space to the router.

 

Detlef

[Detlef]

I did some more try and error. Trying to dial in with my regular cell phone over the PSTN and the Grandstream gateway the audio fails to transmitted to the second pbx system. If I try to call in over a callcentric account the audio works. Comparing the logs the only difference I see is this:

 

Fails - Calling in over Grandstream PSTN Gateway:

[5] 2007/10/08 14:16:22: Using "User Name (Cell)" <sip:757xxxxxxx@localhost> as redirect from

 

Works - Calling in over Callcentric:

[5] 2007/10/08 13:52:42: Using "User Name" <sip:1777xxxxxxx@callcentric.com>;tag=781da70e as redirect from

 

It puts on the grandstream call a @localhost as redirect from rather than the real hostname or IP. Could that be the reason why I dont hear any audio?

 

As attachment both complete logs...

 

Fails - Call over PSTN Gateway:

===============================

[7] 2007/10/08 14:16:09: UDP: Opening socket on port 62650

[7] 2007/10/08 14:16:09: UDP: Opening socket on port 62651

[5] 2007/10/08 14:16:09: Identify trunk (IP address and domain match) 6

[7] 2007/10/08 14:16:09: Set packet length to 20

[6] 2007/10/08 14:16:09: Sending RTP for 74624ba1f9843062@192.168.104.26#59b97d943e to 192.168.104.26:5008

[5] 2007/10/08 14:16:09: Trunk PSTN Cox sends call to 100

[7] 2007/10/08 14:16:09: Received call from cell phone 757xxxxxxx

[8] 2007/10/08 14:16:09: Play audio_en/aa_outbound.wav audio_en/bi_press_1.wav audio_en/aa_goto_mailbox.wav audio_en/bi_press_2.wav audio_en/aa_goto_attendant.wav audio_en/bi_press_3.wav space50

[7] 2007/10/08 14:16:09: Set packet length to 20

[5] 2007/10/08 14:16:10: Registration on trunk 5 (VoIP GER) failed. Retry in 60 seconds

[6] 2007/10/08 14:16:12: Received DTMF 1

[8] 2007/10/08 14:16:12: Play audio_en/ex_enter_access_code.wav

[8] 2007/10/08 14:16:14: Play space20

[6] 2007/10/08 14:16:15: Received DTMF 9

[6] 2007/10/08 14:16:16: Received DTMF 9

[6] 2007/10/08 14:16:17: Received DTMF 9

[8] 2007/10/08 14:16:17: Play audio_en/ex_enter_number.wav

[6] 2007/10/08 14:16:20: Received DTMF 3

[6] 2007/10/08 14:16:21: Received DTMF 8

[6] 2007/10/08 14:16:22: Received DTMF 0

[6] 2007/10/08 14:16:22: Received DTMF #

[5] 2007/10/08 14:16:22: Dialplan: Match 380@localhost to <sip:380@192.168.108.220;user=phone> on trunk VoIP MX

[5] 2007/10/08 14:16:22: Using "User Name (Cell)" <sip:757xxxxxxx@localhost> as redirect from

[8] 2007/10/08 14:16:22: Play audio_moh/noise.wav

[7] 2007/10/08 14:16:22: UDP: Opening socket on port 51824

[7] 2007/10/08 14:16:22: UDP: Opening socket on port 51825

[7] 2007/10/08 14:16:23: Call 63ce148a@pbx#38760: Clear last INVITE

[6] 2007/10/08 14:16:23: Sending RTP for 63ce148a@pbx#38760 to 192.168.108.220:56130

[7] 2007/10/08 14:16:23: 63ce148a@pbx#38760: RTP pass-through mode

[7] 2007/10/08 14:16:23: 74624ba1f9843062@192.168.104.26#59b97d943e: RTP pass-through mode

[7] 2007/10/08 14:16:33: Other Ports: 3

[7] 2007/10/08 14:16:33: Call Port: 09a07909e67a241414771b275404cb9f@192.168.104.235#186da2d5ab

[7] 2007/10/08 14:16:33: Call Port: 63ce148a@pbx#38760

[7] 2007/10/08 14:16:33: Call Port: 9351a6ed@pbx#34231

[7] 2007/10/08 14:16:33: Call 63ce148a@pbx#38760: Clear last request

[5] 2007/10/08 14:16:33: BYE Response: Terminate 63ce148a@pbx

[7] 2007/10/08 14:16:33: Other Ports: 2

[7] 2007/10/08 14:16:33: Call Port: 09a07909e67a241414771b275404cb9f@192.168.104.235#186da2d5ab

[7] 2007/10/08 14:16:33: Call Port: 9351a6ed@pbx#34231

 

Works - Call over Callcentric:

==============================

[7] 2007/10/08 13:52:32: UDP: Opening socket on port 61862

[7] 2007/10/08 13:52:32: UDP: Opening socket on port 61863

[5] 2007/10/08 13:52:32: Identify trunk (line match) 4

[6] 2007/10/08 13:52:32: Sending RTP for NjUxZTFhZWQxZjFkYWE4ZDE1OWUyNDgxOGU3M2Y0ZGQ.#c8a9483aee to 204.11.192.22:63674

[5] 2007/10/08 13:52:32: Trunk VoIP CC sends call to 100

[7] 2007/10/08 13:52:32: Received call from cell phone 1777xxxxxxx

[8] 2007/10/08 13:52:32: Play audio_en/aa_outbound.wav audio_en/bi_press_1.wav audio_en/aa_goto_mailbox.wav audio_en/bi_press_2.wav audio_en/aa_goto_attendant.wav audio_en/bi_press_3.wav space50

[6] 2007/10/08 13:52:34: Received DTMF 1

[8] 2007/10/08 13:52:34: Play audio_en/ex_enter_access_code.wav

[6] 2007/10/08 13:52:36: Received DTMF 9

[6] 2007/10/08 13:52:36: Received DTMF 9

[6] 2007/10/08 13:52:37: Received DTMF 9

[8] 2007/10/08 13:52:37: Play audio_en/ex_enter_number.wav

[6] 2007/10/08 13:52:41: Received DTMF 3

[6] 2007/10/08 13:52:41: Received DTMF 8

[6] 2007/10/08 13:52:42: Received DTMF 0

[6] 2007/10/08 13:52:42: Received DTMF #

[5] 2007/10/08 13:52:42: Dialplan: Match 380@localhost to <sip:380@192.168.108.220;user=phone> on trunk VoIP MX

[5] 2007/10/08 13:52:42: Using "User Name" <sip:1777xxxxxxx@callcentric.com>;tag=781da70e as redirect from

[8] 2007/10/08 13:52:42: Play audio_moh/noise.wav

[7] 2007/10/08 13:52:42: UDP: Opening socket on port 62364

[7] 2007/10/08 13:52:42: UDP: Opening socket on port 62365

[7] 2007/10/08 13:52:42: Call 4408d49c@pbx#5176: Clear last INVITE

[6] 2007/10/08 13:52:42: Sending RTP for 4408d49c@pbx#5176 to 192.168.108.220:54286

[7] 2007/10/08 13:52:42: 4408d49c@pbx#5176: RTP pass-through mode

[7] 2007/10/08 13:52:42: NjUxZTFhZWQxZjFkYWE4ZDE1OWUyNDgxOGU3M2Y0ZGQ.#c8a9483aee: RTP pass-through mode

[7] 2007/10/08 13:52:47: Other Ports: 1

[7] 2007/10/08 13:52:47: Call Port: 4408d49c@pbx#5176

[7] 2007/10/08 13:52:47: Call 4408d49c@pbx#5176: Clear last request

[5] 2007/10/08 13:52:47: BYE Response: Terminate 4408d49c@pbx

[Vodia PBX]

Hmm. The only differernce that I can see is the missing tag on the GS. In SIP 2.0, requests must be tagged on the From header. Maybe they are using the old SIP RFC?

[Detlef]

Hmm. The only differernce that I can see is the missing tag on the GS. In SIP 2.0, requests must be tagged on the From header. Maybe they are using the old SIP RFC?

 

This is what the GS GXW4108 logs in DEBUG mode if I call in:

 

10-08-2007 17:14:19 Sess: 1 INVITE sip:100@192.168.104.220 SIP/2.0 Via: SIP/2.0/UDP 192.168.104.26:5064;branch=z9hG4bK140724266952cdb2 From: "Cell Phone VA"<sip:757xxxxxxx@192.168.104.220>;tag=5120b893f992da17 To: <sip:100@192.168.104.220> Contact: <sip:192.168.104.26:5064> Supported: replaces, timer, path Call-ID: cf40f4173c6140e0@192.168.104.26 CSeq: 31868 INVITE User-Agent: Grandstream GXW4108 (HW 0001, Ch:1) 1.0.1.2 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Type: application/sdp Content-Length: 290 v=0 o=system 8001 8000 IN IP4 192.168.104.26 s=SIP Call c=IN IP4 192.168.104.26 t=0 0 m=audio 5008 RTP/AVP 0 8 18 3 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:3 GSM/8000 a=ptime:20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11

 

10-08-2007 17:14:19 Sess: 1 ACK sip:100@192.168.104.220:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.104.26:5064;branch=z9hG4bKeb37fb8018c5b932 From: "Cell Phone VA"<sip:757xxxxxxx@192.168.104.220>;tag=5120b893f992da17 To: <sip:100@192.168.104.220>;tag=39c9da9605 Contact: <sip:192.168.104.26:5064;user=phone> Call-ID: cf40f4173c6140e0@192.168.104.26 CSeq: 31868 ACK User-Agent: Grandstream GXW4108 (HW 0001, Ch:1) 1.0.1.2 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Length: 0

 

10-08-2007 17:14:28 Sess: 1 BYE sip:100@192.168.104.220:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.104.26:5064;branch=z9hG4bK8b2096112a327c03 From: "Cell Phone VA"<sip:757xxxxxxx@192.168.104.220>;tag=5120b893f992da17 To: <sip:100@192.168.104.220>;tag=39c9da9605 Call-ID: cf40f4173c6140e0@192.168.104.26 CSeq: 31869 BYE User-Agent: Grandstream GXW4108 (HW 0001, Ch:1) 1.0.1.2 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK Content-Length: 0

[Detlef]

Now beats me, I changed some settings around in the GS gateway. Some wait for dialtone timing which should affect my connection problems between the two PBXnSIP but I also added in the channels page for each channel a number in the field as suggested here in some other post about GXW410x... and wonder I all a sudden get audio to my PBXnSIP in Mexico.

 

I dont see any difference in the logfile either... hmmm

 

Working incoming PSTN Call forwarded to second PBXnSIP:

========================================

[7] 2007/10/08 17:57:27: UDP: Opening socket on port 63454

[7] 2007/10/08 17:57:27: UDP: Opening socket on port 63455

[5] 2007/10/08 17:57:27: Identify trunk (IP address and domain match) 6

[7] 2007/10/08 17:57:27: Set packet length to 20

[6] 2007/10/08 17:57:27: Sending RTP for 58806270b5d787c5@192.168.104.26#3cf4ad98e7 to 192.168.104.26:5020

[5] 2007/10/08 17:57:27: Trunk PSTN Cox sends call to 100

[7] 2007/10/08 17:57:27: Received call from cell phone 757xxxxxxx

[8] 2007/10/08 17:57:27: Play audio_en/aa_outbound.wav audio_en/bi_press_1.wav audio_en/aa_goto_mailbox.wav audio_en/bi_press_2.wav audio_en/aa_goto_attendant.wav audio_en/bi_press_3.wav space50

[7] 2007/10/08 17:57:27: Set packet length to 20

[6] 2007/10/08 17:57:32: Received DTMF 1

[8] 2007/10/08 17:57:32: Play audio_en/ex_enter_access_code.wav

[8] 2007/10/08 17:57:34: Play space20

[6] 2007/10/08 17:57:34: Received DTMF 9

[6] 2007/10/08 17:57:35: Received DTMF 9

[6] 2007/10/08 17:57:36: Received DTMF 9

[8] 2007/10/08 17:57:36: Play audio_en/ex_enter_number.wav

[6] 2007/10/08 17:57:38: Received DTMF 3

[6] 2007/10/08 17:57:39: Received DTMF 8

[6] 2007/10/08 17:57:40: Received DTMF 0

[6] 2007/10/08 17:57:40: Received DTMF #

[5] 2007/10/08 17:57:41: Dialplan: Match 380@localhost to <sip:380@192.168.108.220;user=phone> on trunk VoIP MX

[5] 2007/10/08 17:57:41: Using "User Name" <sip:757xxxxxxx@localhost> as redirect from

[8] 2007/10/08 17:57:41: Play audio_moh/noise.wav

[7] 2007/10/08 17:57:41: UDP: Opening socket on port 55478

[7] 2007/10/08 17:57:41: UDP: Opening socket on port 55479

[7] 2007/10/08 17:57:41: Call 93a2ec14@pbx#7094: Clear last INVITE

[6] 2007/10/08 17:57:41: Sending RTP for 93a2ec14@pbx#7094 to 192.168.108.220:59096

[7] 2007/10/08 17:57:41: 93a2ec14@pbx#7094: RTP pass-through mode

[7] 2007/10/08 17:57:41: 58806270b5d787c5@192.168.104.26#3cf4ad98e7: RTP pass-through mode

[7] 2007/10/08 17:57:51: Other Ports: 2

[7] 2007/10/08 17:57:51: Call Port: 93a2ec14@pbx#7094

[7] 2007/10/08 17:57:51: Call Port: 97829f62e3d24894@192.168.104.26#9fb3705bc9

[7] 2007/10/08 17:57:51: Call 93a2ec14@pbx#7094: Clear last request

[5] 2007/10/08 17:57:51: BYE Response: Terminate 93a2ec14@pbx

[Vodia PBX]

Now beats me, I changed some settings around in the GS gateway.

 

So it works now?

[Detlef]

So it works now?

 

Yes it actually does work now. The only other thing I changed yesterday was the codec preference under the PBXnSIP settings in the ports screen. There I had one system configured to the default 0 8 18 2 3 and the other one to 18 0 8 2 3. I have both systems licenses with 10 G.729 codecs. So both should have been able to handle the preference 18 from the one system.

 

Anyway, I changed both back to the default 0 8 18 2 3. Dont ask me if that made the difference.

 

Detlef