[Asterisk-Users] Unable to dial certain calls

Tue Jul 12 07:49:11 MST 2005

First off kill the Glaw.  It doesn't exist.
Then try your call.  But also why are you sending the line congestion
when you first start to make a call.  That's normally used as a closure.

But from what I can see about the only thing wrong is the GLAW.  Kill
that and you should be good to go.

..o-------------------------------------------------------o.
Brian Fertig
NOC/Network Engineer
Planet Telecom, Inc.
Tampa, FL Office

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of JP Russell
Sent: Tuesday, July 12, 2005 5:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Unable to dial certain calls

Of course.  Note that I have no idea what "glaw" is but 
someone on some board 
shttps://paranoia.demon.nl/SkinFiles/jpruss.com/GoldenFleece/send.gif
https://paranoia.demon.nl/SkinFiles/jpruss.com/GoldenFleece/send.gifugge
sted 
it as a resolution to a similiar problem so I put it in.

The entry from the iax.conf file is:
[vbx]
type=peer
host= 213.61.187.150
secret=-my password-
notransfer=yes
context=def
allow=glaw
allow=ulaw
allow=gsm

and from extensions.conf I guess you need the [def] 
context entries.

they are:

;NL
exten => _00316.,1,Congestion
exten => _00319.,1,Congestion
exten => _0031X.,1,SetCallerID("Not Available" 
<7005551212>)
exten => _0031X.,2,Dial,IAX2/jpr at vbx/${EXTEN}
exten => _0031X.,3,Playback(invalid)
exten => _0031X.,4,Hangup
;US
exten => _001X.,1,SetCallerID("Not Available" 
<7005551212>)
exten => _001X.,2,Dial,IAX2/jpr at vbx/${EXTEN}
exten => _001X.,3,Playback(invalid)
exten => _001X.,4,Hangup

Finally sip.conf includes the below paramaters:

[general]
disallow=all
allow=ulaw
allow=glaw
allow=gsm
port = 5060			; Port to bind to
bindaddr = 0.0.0.0		; Address to bind to
context = from-sip		; Default for incoming calls
callerid=No CallID

[2203]
port=5061
username=-thisusername-
secret=-this password-
host=dynamic
type=friend
nat=1
qualify=no
;reinvite=no
canreinvite=yes
context=intern

On Mon, 11 Jul 2005 22:55:49 -0400
  "Brian C. Fertig" <brian at planet-telecom.com> wrote:
> Check your codecs..  Can you post a sniplet of your IAX, 
>SIP, and extensions.conf for dialing the US so we can see 
>were the problem may lie?
> 
> Brian Fertig
> 
> 
> ________________________________
> 
>From: asterisk-users-bounces at lists.digium.com on behalf 
>of JP Russell
> Sent: Mon 7/11/2005 9:12 PM
> To: asterisk-users at lists.digium.com
> Subject: [Asterisk-Users] Unable to dial certain calls
> 
> 
> 
> To begin with, I am new to both asterisk and VOIP and 
>although I've 
> gotten pretty far with my Asterisk setup and have two 
>different sip 
> accounts working fine for outgoing calls I am having 
>trouble with one 
> issue.
> 
> My problem is that I have another provider who uses IAX2 
>that I wish 
> to use for calling various countries, including local 
>(The 
> Netherlands) calls and calls to the US to name two.  I 
>am able to 
> call local numbers without a problem through this 
>provider with 
> Asterisk, but calling US numbers is not working.
> 
> I CAN call the same US numbers with the service by using 
>a direct 
> connection from a softphone for example.
> 
> The entries that show up in the log after failed 
>attempts to call the 
> US are:
> 
> Jul 11 20:04:04 WARNING[25225728]: File channel.c, Line 
>1851 
> (ast_channel_make_compatible): No path to translate from 
>SIP/2203-2929
> (4) to IAX2[vbx]/1(16)
> Jul 11 20:04:04 WARNING[25225728]: File app_dial.c, Line 
>672 
> (dial_exec): Had to drop call because I couldn't make 
>SIP/2203-2929 
> compatible with IAX2[vbx]/1
> 
> I don't see anything suspicious entries in the CLI 
>logging with IAX2 
> debugging on.  Searching the archives and google didn't 
>turn up a 
> solution to this or even point me in the right direction 
>I'm afraid.
> 
> Anyone have any idea on what my problem is or I can go 
>for this issue?
> 
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