[asterisk-users] Code in extensions.conf to leave a voicemailin another PBX ?!

Wed May 5 09:37:45 CDT 2010

Hi Guys,
first of all, thanks Danny for your support trying to help is a big help
itself.
so the thing is:
from pbx1 to pbx2 which was able to leave VM, it was set up like this:
exten => 8021,1,Dial(IAX2/pbx2/${EXTEN},30,tTWwr)

but from pbx2 to pbx1 which was not able to leave VM, it was setup like
this:
exten => 8093,1,Dial(IAX2/pbx1/${EXTEN},20,tTWwr)

that seems to me suly, but though i wnet ahead and modified the only
difference which is the ring time from 20 to 30, and IT WORKED!!!
i wasted some time going over values, and it seems like it's working for 21
but not for 20, maybe a pro can give us precise explanation, but at least i
can leave a VM now :)!

2010/5/5 Danny Nicholas <danny at debsinc.com>

>  This is a little over my head, but the message indicates that you don’t
> have a fully authorized connection.  Can you post the iax.conf snippets
> relevant to the call?
>
>
>  ------------------------------
>
> *From:* asterisk-users-bounces at lists.digium.com [mailto:
> asterisk-users-bounces at lists.digium.com] *On Behalf Of *khalid touati
> *Sent:* Wednesday, May 05, 2010 8:36 AM
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Code in extensions.conf to leave a
> voicemailin another PBX ?!
>
>
>
> Thank you Danny, but it says in the link that it's an iptables issue,
> though i allowed everything on this network interface and even stopped
> iptables but still i have this issue.
>
> 2010/5/4 Danny Nicholas <danny at debsinc.com>
>
> See if this helps
>
> http://www.voipuser.org/forum_topic_3921.html
>
>
>
>
>  ------------------------------
>
> *From:* asterisk-users-bounces at lists.digium.com [mailto:
> asterisk-users-bounces at lists.digium.com] *On Behalf Of *khalid touati
> *Sent:* Tuesday, May 04, 2010 11:35 AM
>
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>
> *Subject:* Re: [asterisk-users] Code in extensions.conf to leave a voice
> mailin another PBX ?!
>
>
>
> Hi Guys,
> so when i dial from an asterisk 1.2 to asterisk 1.4 i get the following
> warning:
> WARNING[640]: file.c:738 ast_readaudio_callback: Failed to write frame
> is anyone familiar with?
>
> 2010/4/29 khalid touati <khalidtouati at gmail.com>
>
> Hi Guys,
> Danny: as i said from pbx1 (1.4) to pbx2 (1.2) it's working fine.
> Peder: i just didn't want to put a lot of lines, (by the way it's dialing
> talking fine), but here you are:
>
> [macro-stdexten]
>
> exten => s,n,Dial(SIP/${ARG1}&IAX2/${ARG1}@${ARG1},20,tTrWw)    ;Ring
> phone for 20 seconds
>
>
> exten => s,n,Goto(s-${DIALSTATUS},1)
>
> exten => s-NOANSWER,1,Voicemail(u${ARG1})
> exten => s-NOANSWER,2,Goto(default,s,1)
>
> exten => s-BUSY,1,Voicemail(b${ARG1})
> exten => s-BUSY,2,Goto(default,s,1)
>
> exten => _s-.,1,Goto(s-NOANSWER,1)
>
> exten => a,1,VoicemailMain(${ARG1})
>
>   2010/4/29 Peder <peder at networkoblivion.com>
>
> In PBX1, where are you actually dialing the phone?  The first line of the
> macro just says “goto dialstatus” with no Dial statement.
>
>
>
>
>
> *From:* asterisk-users-bounces at lists.digium.com [mailto:
> asterisk-users-bounces at lists.digium.com] *On Behalf Of *khalid touati
>
>
> *Sent:* Thursday, April 29, 2010 2:03 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>
> *Subject:* [asterisk-users] Code in extensions.conf to leave a voice mail
> in another PBX ?!
>
>
>
> Hi Guys,
> i spent some time to figure this out (since i love how dialplan is written)
> but i decided to ask for your help guys.
>
> i have two asterisk servers one is 1.2 the other is 1.4, from 1.4 (pbx1) to
> 1.2 (pbx2) i can leave a voice mail without any pb, but from pbx2 to pbx1 it
> just hang up.
>
> in pbx2 extensions.conf:
> i am using: exten => 8029,1,Dial(IAX2/pbx1/${EXTEN},20,tTWwr)
>
> in pbx1, i have:
> exten => 8029,1,Macro(stdexten,8029)
> and in stdexten macro:
>
> exten => s,n,Goto(s-${DIALSTATUS},1)
> exten => s-NOANSWER,1,Voicemail(u${ARG1})
> exten => s-NOANSWER,2,Goto(default,s,1)
>
> exten => s-BUSY,1,Voicemail(b${ARG1})
> exten => s-BUSY,2,Goto(default,s,1)
>
> exten => _s-.,1,Goto(s-NOANSWER,1)
> exten => a,1,VoicemailMain(${ARG1})
>
> when calling from 8021(pbx2) to 8029(pbx1) i get on CLI pbx1:
>
> -- Executing [s at macro-stdexten:6] Goto("IAX2/pbx2-15464", "s-NOANSWER|1")
> in new stack
>     -- Goto (macro-stdexten,s-NOANSWER,1)
>     -- Executing [s-NOANSWER at macro-stdexten:1]
> VoiceMail("IAX2/pbx2-15464", "u8029") in new stack
> *[Apr 29 14:36:35] WARNING[7307]: file.c:738 ast_readaudio_callback:
> Failed to write frame*
>     -- <IAX2/pbx2-15464> Playing
> '/var/spool/asterisk/voicemail/default/8029/unavail' (language 'en')
>   == Spawn extension (macro-stdexten, s-NOANSWER, 1) exited non-zero on
> 'IAX2/pbx2-15464' in macro 'stdexten'
>   == Spawn extension (default, 8029, 1) exited non-zero on
> 'IAX2/pbx2-15464'
>     -- Hungup 'IAX2/pbx2-15464'
>
> any other ideas how to be able to leave a voice mail from 1.2 to 1.4 or fix
> the issue I'm having, thanks a lot!
>
> --
> Abdullah
>
>
>
> --
>
>
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>
>
>
> --
> Abdullah
>
>
>
>
> --
> Abdullah
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>               http://www.asterisk.org/hello
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>
>
>
> --
> Abdullah
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>               http://www.asterisk.org/hello
>
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>

-- 
Abdullah
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