T1 AMI, D4 one-way audio problem

mebv

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#1
Hello,

I'm having problem with a T1 line from telco. Is not PRI.
I configured the T1 as ami,d4.

zaptel.conf

span=1,0,0,d4,ami
e&m=1-24

zapata-channels.conf o zapata.conf

signalling = em_w
channel=>1-24

I can receive calls and DID. Works OK.
I can make calls but nobody can hear me on other side. I can hear good.
I test from SIP softphone and from Grandstream and Linksys phones. The same.
When I connect and IAX extension via Zoiper the audio works OK both way when I call.
I'm using ulaw.
I tested on different T1 line from same telco and works OK both way.
The telco tell me everything is OK but I know that there's something wrong about because nobody can hear me when I make a call.

Any help about?

Thanks in advance.
 

dicko

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#2
I wish I could help, I have more than a little experience with ami,D4 trunks working with asterisk boxes and have never seen anything like this,
a) IAX2 works but not SIP
b) Same Telco T1's work but not this one

Something's obviously wrong! probably asterisk doesn't fully recognise the "connected" state, but why?

variables to eliminate I would include in the trouble shooting:-

The CSU/DSU involved, (can you trade it out with a working T1's CSU/DSU? )
Signalling from Telco, they say it is fine but there are MANY variables that can be mismatched, but is it provisioned "exactly" the same as the workering ones?, (wink timing, feat-d CallerID, DID's, dialtone presentation ad nauseum . . ?) Only they can be authoritative as to two trunks being exactly the same but they all "lie like dogs"!

I would be tempted to add immediate=yes,and possibly callprogress=yes to zapata.conf (restart asterisk, to make it take) (I know that doesn't make sense but YNK), also I'd watch zttool to ensure the call flags shows "answered", maybe ztmonitor and scream into the mouthpiece to see ANY change in the TX graph.

Maybe more clues here:
http://www.voip-info.org/wiki/index.php ... apata.conf


... I'm intrigued, please let me know and good luck

Dick


p.s.
as an afterthought do you have allow=ulaw AFTER any "deny=" statements in your extension configs?
 

mebv

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#3
Hi Dicko,

I tried inmediate=yes and ulaw as you told me already and nothing.
There's something very strange.
The worst thing for me is that the customer has the previous PBX (Samsung) working OK with this T1 line.
The problem is just with Asterisk.

I spent few hours with telco trying to fix this and they told me everything is OK in their side (of course, I think not) but the actual PBX works fine.

I'll keep you in informed.

Thanks.
 

ramoncio

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#4
I have no experience with T1, but maybe if you enable full debugs you can see the problem.

The following commands are available if the channel is built with support for libpri:

* pri debug span: Enables PRI debugging on a span
* pri intense debug span: Enables REALLY INTENSE PRI debugging
* pri no debug span: Disables PRI debugging on a span
* pri show spans: List spans and their status.
* pri show span: Information about a span.
* pri show debug: show where debug is enabled.

Ask the Samsung PBX to do a pri intense debug!! XD
This is the best about open source.

Anyway, I'm not too worried if Dicko is here to help you. He is a crack. :)
 

ramoncio

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#5
I have just noticed in your first post that your T1 is not PRI, so maybe this commands won't work for you, I don't know.
But you should have some alternative debug commands for ami,d4..
 

dicko

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#6
hey ramoncio:

I wish!

Unfortunately, traditional T1's have no debugging in asterisk except for the ztmonitor/zttool and the good old CLI zap show channel(s) zap show span(s)

mebv:

can you post a full debug (set debug 99, set verbose 99 from CLI) output of /var/log/asterisk/full of an outgoing phone call?
 

ramoncio

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#7
I've been testing with Odicha his patches to support HFC cards in dahdi for BRI isdn connections. And you can use all the PRI debug commands. This is cool. BRI is really widely used here and mISDN had lots of problems.
 

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