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Trixbox 2.6.18 and Bandwidth.com dropped audio on outbound calls only.

yakbone
Posts: 4
 Member Since:
2010-01-28
Submitted by yakbone on Mon, 03/01/2010 - 6:46pm.

I am using Trixbox CE 2.6.18 with the defaults (asterisk version 1.4.22). I have purchased sip service through bandwidth.com. We are connecting to bandwidth.com using an Edgewater 4500. The ping times are less than 100ms.

In my current configuration, I can receive calls with no problem. The problem I am running into, is that all outbound calls will lose audio between 30 seconds and 17 1/2 minutes. The timing seems to be random. I have no problem connecting the calls, but only with a total loss of audio in both directions at some point in that time range. Once audio fails it does not return. I have even gone so far as to try with a virtualized Trixbox 2.8 with vanilla trunk information (meaning cut and pasted from their setup guide in their forum)

I have been working with their tech support (good response times). When they changed us to using the "premium routes", the issue disappears. My support representative says that there must be something in my configuration that causes this. As I am still wet behind the ears (but learning), this is quite possible.

My sip trunk configuration is as follows:

canreinvite=yes
context=from-pstn
dtmfmode=rfc2833
host=192.168.0.200
nat=yes
externip=173.160.xxx.xxx
localnet=192.168.0.0/255.255.255.0
outboundproxy=192.168.0.200
progressinbound=yes
qualify=yes
type=peer
stunaddr=stun.zoiper.com:3478
session-timer=refuse

I ran a tcpdump on the Trixbox today during a failed call. It seems that the SIP phone (Zoiper, if it matters) sends several invite requests to Trixbox at the point where audio drops. My technician theorized that the client sending invites may be causing Trixbox to drop the audio.

I could find nothing in /var/log/messages that indicated distress.

During our traces, it seems that Trixbox would send a malformed BYE after the time the audio drops.

BYE sip:+16788170654@ot.bandwidth.com:5060 SIP/2.0
Via: SIP/2.0/UDP 173.160.xxx.xx:5060;branch=z9hG4bK489251fb;rport
Record-Route:<sip:EWGW_0@173.160.173.26;lr>
Route:<sip:216.82.224.202:5060;lr;ftag=as194351eb>
From:"4258181587"<sip:4258181587@ot.bandwidth.com:5060>;tag=as194351eb
To:<sip:+16788170654@ot.bandwidth.com:5060>;tag=gK0695b992
Call-ID: 3aa686bc601a2c8176f5a14651f104a6@192.168.0.101
CSeq: 103 BYE
User-agent: Asterisk PBX
Max-forwards: 70
Content-Length: 0

If I have omitted anything major that would help to point out what I have done wrong, please let me know.

Thanks for considering this.

Joshua

‹ Problem with SIP trunk Messagenet - build 2.8.0 Outgoing inputs not working ›


415eric
Posts: 416
 Member Since:
2009-10-29
Remove externip=173.160.xxx.
Mon, 03/01/2010 - 7:12pm

Remove

externip=173.160.xxx.xxx
localnet=192.168.0.0/255.255.255.0

That should already be in your sip_nat.conf. If it isn't you should configure your sip_nat.conf file.

--


yakbone
Posts: 4
 Member Since:
2010-01-28
Thank you very much for
Mon, 03/01/2010 - 7:39pm

Thank you very much for pointing that out. I made the changes suggested. It still fails, but I do thank you for the lesson. I think I got overzealous when setting up my remote extensions.

Joshua



415eric
Posts: 416
 Member Since:
2009-10-29
Your host is wrong also.
Mon, 03/01/2010 - 9:47pm

Your host is wrong also. Where did you get the config template for your trunk?

The host should be the FQDN of the sip provider like in your case I think it should be sip.broadvoice.com. You need to verify this with broadvoice.

--


yakbone
Posts: 4
 Member Since:
2010-01-28
Hello @415eric, Thanks for
Tue, 03/02/2010 - 10:49am

Hello @415eric,

Thanks for working with me here.

I got the basic information for the trunk from the forum at bandwidth.com. The IP address 192.168.0.200 is the Edgewater provided by Bandwidth. The instructions say to point the host to the Edgewater IP address. I will not ask you to read the document, but I post the link here, just to show where I got the information.

http://forum.bandwidth.com/showthread.php?tid=40

The basic trunk config looked like this:

canreinvite=yes
context=from-pstn
dtmfmode=rfc2833
host=192.168.0.200
nat=no
outboundproxy=192.168.0.200
progressinbound=yes
qualify=no
type=peer

The rest of the fluff in there pertained to my attempts to incorporate remote extensions. This was the trunk information used when I attempted TB 2.8 in a virtual machine.

Best regards,
Joshua



415eric
Posts: 416
 Member Since:
2009-10-29
I would remove the the extra
Tue, 03/02/2010 - 11:31am

I would remove the the extra stuff from your trunk and just get basic functionality going first. Once you have that configured then progress towards getting remote phones working. I can't say for certain but in my experience I have not had to make changes to my trunk settings to get my remote phones to work.

--


yakbone
Posts: 4
 Member Since:
2010-01-28
I set up the virtual machine
Wed, 03/03/2010 - 8:42am

I set up the virtual machine for testing using just the trunk information from my previous post (the basic default settings). This is TB 2.8 with asterisk version 1.6. My test call loses audio at 1.5 minutes. The packet capture again shows invites sent from my SIP phone at that time. It looks as though the invites are sent immediately when audio drops.

Please let me know if I have omitted relevant information.

Thank you for your attention.



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