[WXC/VFX] Trixbox with VFX trunk registered

Forums › One New Zealand (including Vodafone, WxC, Farmside) › [WXC/VFX] Trixbox with VFX trunk registered

zimbonz

96 posts

Master Geek

#19834 3-Mar-2008 12:07

Hi All,
I got my vfx trunk setup today - which I have configured using Tony Hughes' settings http://www.geekzone.co.nz/tonyhughes/4186 (thanks!)

My trunk is now showing as registered, so I assume all is well with the config. I have a catchall incoming route - but when I try to call my vfx number (from mobile), I get a male, NZ voice telling my that "The party is unavailable etc." I am unable to dial out either, despite correct routes.

Is there something else I need to look at, to get this working? Sorry for posting here 0 but I am on hold with xnet support as we speak- still not getting through. Just checking that it is not something simple that I can check.

Cheers

maverick

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#114316 3-Mar-2008 12:34

2 Things,

First you are registered no problem there and calls are going to your box, but your Box is rejecting all call with a 488 not accepatable, generally relates to codecs not being supported so I would check there first.

Second we see no call attempts whatsoever from your box so i'm afraid your routing rules are probably not correct.

Yes I am a employee of WxC (My Profile) ... but I do have my own opinions as well

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tonyhughes

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#114317 3-Mar-2008 12:44

I just set mine up again after a long break with no voip connected. I was having random issues that seem to have disappeared.

If anyone gets VFX running on AsteriskNOW I would like to hear from them....

99% of problems I have had with VFX have been routing issues on my end (and the other 1% other stupid stuff on my end too).

zimbonz

96 posts

Master Geek

#114380 3-Mar-2008 18:49

Thank you both for your prompt reply's. With regards to codec's - do I need to purchase the G729 codec? I have configured the peer settings of the trunk to:
allow=ulaw&alaw

as per instructions, should that be sufficent?

Also, with outbound routes, I have even created a separate route for my mobile number, with the number being the only one listed in the dial pattern. The log shows that the call to my mobile - from my softphone - is going through the VFX channel - but I get a error 603 declined on my xLite softphone..?

I appreciate your assistance with this.

maverick

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#114399 3-Mar-2008 20:13

The G711 should be okay but G729 is also used and may be well worth the price,

we still see a 488 unaccepatable when sending calls to your box 7 calls since 18:10 so far tonight,

the 603 declined is also coming from your Box and not us as we still see no call attempts from your device the calls is not actaully being sent to us. sorry but it will be in your config somewhere, if you post your details minus all the Phone numbers and passwords someone from the asterisk community may be able to assist you.

Yes I am a employee of WxC (My Profile) ... but I do have my own opinions as well

https://www.facebook.com/wxccommunications

zimbonz

96 posts

Master Geek

#114401 3-Mar-2008 20:18

Here is a section of the debug logs which looks interesting?

[Mar 3 20:07:25] WARNING[2808] rtp.c: Unable to set TOS to 184
[Mar 3 20:07:25] WARNING[2808] chan_sip.c: No audio format found to offer. Cancelling call to 0272xxxxxx
[Mar 3 20:07:25] VERBOSE[2808] logger.c: -- Couldn't call VFX/0272xxxxxx
[Mar 3 20:07:25] VERBOSE[2808] logger.c: == Everyone is busy/congested at this time (0:0/0/0)
[Mar 3 20:07:25] DEBUG[2808] app_macro.c: Executed application: Dial
[Mar 3 20:07:25] VERBOSE[2808] logger.c: -- Executing [s@macro-dialout-trunk:21] Goto("SIP/100-09c42938", "s-CHANUNAVAIL|1") in new stack
[Mar 3 20:07:25] VERBOSE[2808] logger.c: -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
[Mar 3 20:07:25] DEBUG[2808] app_macro.c: Executed application: Goto
[Mar 3 20:07:25] VERBOSE[2808] logger.c: -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] GotoIf("SIP/100-09c42938", "1?noreport") in new stack
[Mar 3 20:07:25] VERBOSE[2808] logger.c: -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,3)
[Mar 3 20:07:25] DEBUG[2808] app_macro.c: Executed application: GotoIf
[Mar 3 20:07:25] VERBOSE[2808] logger.c: -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:3] NoOp("SIP/100-09c42938", "TRUNK Dial failed due to CHANUNAVAIL - failing through to other trunks") in new stack
[Mar 3 20:07:25] DEBUG[2808] app_macro.c: Executed application: Noop
[Mar 3 20:07:25] VERBOSE[2808] logger.c: -- Executing [0272xxxxxx@from-internal:6] Macro("SIP/100-09c42938", "outisbusy|") in new stack

I appreciate this is not your area of responsibility Maverick, does this thread need to be moved to a more appropriate channel? I am not sure how to do that.

zimbonz

96 posts

Master Geek

#114405 3-Mar-2008 20:23

I should clarify that the logs above were created by attemting an outgoing call, and not incoming. I thought that it would be easier to tackle first..!

Thanks for your vfx assistance maverick, I guess I better look to the Trixbox/asterix community for assitance now..!

zimbonz

96 posts

Master Geek

#114409 3-Mar-2008 20:26

I get this on an incoming call...

Mar 3 20:24:08] WARNING[2565] rtp.c: Unable to set TOS to 184
[Mar 3 20:24:08] NOTICE[2565] chan_sip.c: No compatible codecs, not accepting this offer!
[Mar 3 20:24:18] VERBOSE[2856] logger.c: == Parsing '/etc/asterisk/manager.conf': [Mar 3 20:24:18] VERBOSE[2856] logger.c: Found
[Mar 3 20:24:18] VERBOSE[2856] logger.c: == Parsing '/etc/asterisk/manager_additional.conf': [Mar 3 20:24:18] VERBOSE[2856] logger.c: Found
[Mar 3 20:24:18] VERBOSE[2856] logger.c: == Parsing '/etc/asterisk/manager_custom.conf': [Mar 3 20:24:18] VERBOSE[2856] logger.c: Found

Lenovo

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zimbonz

96 posts

Master Geek

#114463 3-Mar-2008 22:26

Ok, I have managed to resolve this.

I had initially followed Tony's instructions here, to setup vfx, from his blog page. I had not followed Josh's instructions here, as it looked like Tony's had superceeded them, or at least offered as an alternative.

Tony's blog does not mention editing the sip.conf files, so I did not. However, the errors above showed up. So tonight, I edited both files - the sip.conf and sip_additional.conf - with exactly what Josh's instructions said to do. Guess what - It Works!!

However, I then hit my next speedbump. All my routes were connecting to my existing sip trunk, made using the web interface. So every time I applied the config for something else - I overwrote the sip_additional.conf with the incorrect settings.

So I deleted the trunk completely, and just editted the the sip_additional.conf as instructed by Josh. BUT then no trunk shows up in freepbx to assign routes etc to - so I am STILL stuck.

Eventually - I setup a trunk as per Tony Hughes blog, but using the data from Josh's blog for the peer details section. I removed the User Context:authID setting, and kept the available channels to 1 instead of 2. This allowed me to configure freepbx as normal.

I can safely say that this now works. The sip.conf has the available codecs to use, instead of the peer details of the trunk. I think this is what fixed it partly for me.

I hope this is not taken as any unconstructive critisism - rather it would be nice for the two methods to be combined - as they are quite different, and produce different results.
I hope this helps someone else who may be in the same boat.

Cheers

tonyhughes

Hawkes Bay
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#114510 4-Mar-2008 07:51

Good stuff.

My blog post method works for me, but of course, everyones results can vary. :-)

joshp

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#114517 4-Mar-2008 08:42

Hi Zimbo,

These configuration details a fairly high level, but even using freepbx (the web interface) basically the information you are editing will end up in the sip_additional configs, so if you add all of the details as specified into the freepbx interface you should be able to get yourself up and running..

There are plent of guides available on how to use freepbx, so take a look around and see how you get on.. :)

Cheers

Josh

http://wxc.co.nz

zimbonz

96 posts

Master Geek

#114520 4-Mar-2008 08:56

Yes thanks Josh, that is what I ended up doing, entering your config details into the Freepbx web interface, all is ok now.

Tony, perhaps the yum update has caused some "funnies" with my box? Seems that there are a few people who have had a few strange issues recently after the latest update. Thanks for your info on freecall btw - my wife is very happy..!