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@lists.digium.com
[mailto:asterisk-users-bounces@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@vbx/${EXTEN}
exten => _0031X.,3,Playback(invalid)
exten => _0031X.,4,Hangup
;US
exten => _001X.,1,SetCallerID("Not Available"
<7005551212>)
exten => _001X.,2,Dial,IAX2/jpr@vbx/${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@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@lists.digium.com on behalf
>of JP Russell
> Sent: Mon 7/11/2005 9:12 PM
> To: asterisk-users@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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