[Asterisk-Users] * and Cisco Gateways

[Jess Magnaye]

Testing between ATA and Asterisk is working fine. I am getting voicemail
etc.  But when I'm trying to call to the "carrier side" i find it not
working.  I see on my Cisco gateway that it negotiated the g711ulaw codec,
but when the "state" goes into active, I just automatically get busy tone
from my ATA.  It looks like that the ATA-* leg is disconnected while
*-CiscoGateway is still trying to connect.  My ATA is setup as g711ulaw
(using Txcodec:2, Rxcodec:2).

Probably * is trying to negotiate with ATA using GSM codec? Is there a
command in * (or * utility) that I can use to debug the codec negotiation,
and/or the RTP status?

My sip and exten are as follows:


sip.conf
--------
[general]
port = 5060
bindaddr = 0.0.0.0
context = siptest
tos=lowdelay
tos=184
maxexpirey=3600
disallow=all
allow=gsm
allow=ulaw

register => XXXX at mydomain

nat=yes
query=yes
autocreatepeer=yes

[carrier]
type=friend
fromdomain=mydomain
host=<X.X.X.X - carrier IP>


extensions.conf
----------------
;extension
exten => 1234,1,Dial(SIP/1234,10,tr)
exten => 1234,2,Voicemail,u1234
exten => 1234,102,Voicemail,b1234

;oubound to carrier
exten => _011.,1,Dial(SIP/{$EXTEN}@carrier,tr)

;getting voicmails
exten => 73*,1,VoicemailMain





----- Original Message ----- 
From: "Arslan Saeed" <Arslan.Saeed at resgrp.com.pk>
To: <asterisk-users at lists.digium.com>
Sent: Thursday, January 08, 2004 6:55 PM
Subject: RE: [Asterisk-Users] * and Cisco Gateways


Hi, Send me your * configuration u r using for troubleshooting this
problem.

Arslan.

-----Original Message-----
From: Jess Magnaye [mailto:jess at arretni.com]
Sent: Friday, January 09, 2004 1:37 AM
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] * and Cisco Gateways
Importance: High

That didn't work. I ran tcpdump and looks like looks like my Linux has
problems with its on-board Ethernet.  I'm getting the following error,
which
most likely is the reason for no audio.

14:40:57.628080 asterisk > cisco: icmp: asterisk udp port 13696
unreachable
[tos 0xd4]
14:40:58.275665 asterisk > cisco: icmp: asterisk udp port 13697
unreachable
[tos 0xc0]
14:40:58.622308 asterisk > cisco: icmp: asterisk udp port 13696
unreachable
[tos 0xd4]

I am now installing a new linux with (pci-ethernet) to load the *.  I
hope
when I'm done, things will go smoothly.



----- Original Message ----- 
From: "John Todd" <jtodd at loligo.com>
To: <asterisk-users at lists.digium.com>
Sent: Thursday, January 08, 2004 1:32 PM
Subject: Re: [Asterisk-Users] * and Cisco Gateways


> >Anybody on the list who implemented Cisco ATA + * + Cisco 2600?  I
> >cannot get my calls from ATA to terminate to the Cisco gateway via
> >*.  I am not sure if it is my hardware problem.  I'm getting the
> >following "codec negotiation problem" from Cisco.
> >
> >23:39:08:  Unexpected VoIPCodec Type :g729br8
> >23:39:08:  Unexpected VoIPCodec Type :gsmefr
> >
> >
> >I appreciate any help I can get.  Thanks.
>
> Go into sip.conf, and add these lines to the SIP peer for your Cisco
2600:
>
> disallow=all
> allow=ulaw
> allow=alaw
>
>
> This will force G.711 codec usage, which may solve your problems
> though it will increase your bandwidth.
>
> JT
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users

_______________________________________________
Asterisk-Users mailing list
Asterisk-Users at lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________
Asterisk-Users mailing list
Asterisk-Users at lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users