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

[Danny Nicholas]

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

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