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 <da...@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-boun...@lists.digium.com [mailto: > asterisk-users-boun...@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 <da...@debsinc.com> > > See if this helps > > http://www.voipuser.org/forum_topic_3921.html > > > > > ------------------------------ > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@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 <khalidtou...@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/${ARG1}&IAX2/${ar...@${arg1},20,tTrWw) ;Ring > phone for 20 seconds > > > exten => s,n,Goto(s-${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(${ARG1}) > > 2010/4/29 Peder <pe...@networkoblivion.com> > > In PBX1, where are you actually dialing the phone? The first line of the > macro just says “goto dialstatus” with no Dial statement. > > > > > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *khalid touati > > > *Sent:* Thursday, April 29, 2010 2:03 PM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > > *Subject:* [asterisk-users] Code in extensions.conf to leave a voice mail > in another PBX ?! > > > > Hi Guys, > i spent some time to figure this out (since i love how dialplan is written) > but i decided to ask for your help guys. > > i have two asterisk servers one is 1.2 the other is 1.4, from 1.4 (pbx1) to > 1.2 (pbx2) i can leave a voice mail without any pb, but from pbx2 to pbx1 it > just hang up. > > in pbx2 extensions.conf: > i am using: exten => 8029,1,Dial(IAX2/pbx1/${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 [...@macro-stdexten:6] Goto("IAX2/pbx2-15464", "s-NOANSWER|1") > in new stack > -- Goto (macro-stdexten,s-NOANSWER,1) > -- Executing [s-noans...@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 > > > > -- > > > _____________________________________________________________________ > -- 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 > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > -- > 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 > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > -- > 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 > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- 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 asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users