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mdshs

@teksavvy.com

[Asterisk] Forwarding voicemail never works?

Hey everyone, I use a hosted FreePBX running 2.9 with Asterisk 1.6 on PIAF. The issue I'm having is if someone leaves me a voicemail and I forward it to another user, they never get it. They get an email saying you have a new voicemail but their phone doesn't have anything if they call in. The weird thing is I was with one hosted provider before and it did this, and then I switched and same thing. I had them do a factory restore of my PBX and still the same thing. Does anyone have any idea what might cause this? Receiving voicemails works fine we just can't forward them from the phone so instead we have to email the attachment to the other person. Any ideas?


mdshs

@teksavvy.com

Odd thing too is just realized playing the attachment as a wav file it says this is an invalid file type that itunes cannot recognize.



nunya
Who is John Galt?
Premium,MVM
join:2000-12-23
O Fallon, MO
kudos:5
Reviews:
·Charter
·voip.ms

On the extension to which your are trying to forward to, look in the extension settings on FreePBX.
Make sure "Delete Voicemail" is set to "NO".
--
...because I care.


mdshs

@teksavvy.com

Just checked they're all set to no..



Trimline
Premium
join:2004-10-24
Windermere, FL
Reviews:
·voip.ms
·Callcentric
·RoadRunner Cable

reply to mdshs
I had some notes on this as I couldn't get the email attachments to play on my Android. After changing, it worked just fine. Here's what I recall doing...

"the band-aid solution for me was to change the voicemail.conf file on my asterisk server to just use the format wav and not wav49 or gsm. The files are larger now, but it's worth it in my opinion. I still would love to see native support for playing wav49 files."



nunya
Who is John Galt?
Premium,MVM
join:2000-12-23
O Fallon, MO
kudos:5

reply to mdshs
I never used 1.6, I went straight from 1.4 to 1.8. Could this be something inherent with 1.6?

Can you try to forward a voicemail to another extension and then post an asterisk logfile?
--
...because I care.



mdshs

@teksavvy.com

Wav only didn't help. Here's the log file of me calling my voicemail for extension 4945 forwarding the first message to 84945 without prepending it and then calling in *98 to check it from 84945. Not sure if this helps at all doesn't seem too detailed.

== Parsing '/etc/asterisk/asterisk.conf': == Found
== Parsing '/etc/asterisk/extconfig.conf': == Found
Connected to Asterisk 1.6.2.16.1 currently running on voice (pid = 1324)
Verbosity is at least 3
-- Remote UNIX connection
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [*97@from-internal:1] Answer("SIP/4945-00000007", "") in new stack
-- Executing [*97@from-internal:2] Wait("SIP/4945-00000007", "1") in new stack
-- Executing [*97@from-internal:3] Macro("SIP/4945-00000007", "user-callerid,") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/4945-00000007", "AMPUSER=4945") in new stack
-- Executing [s@macro-user-callerid:2] GotoIf("SIP/4945-00000007", "0?report") in new stack
-- Executing [s@macro-user-callerid:3] ExecIf("SIP/4945-00000007", "1?Set(REALCALLERIDNUM=4945)") in new stack
-- Executing [s@macro-user-callerid:4] Set("SIP/4945-00000007", "AMPUSER=4945") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/4945-00000007", "AMPUSERCIDNAME=First Last") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/4945-00000007", "0?report") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/4945-00000007", "AMPUSERCID=4945") in new stack
-- Executing [s@macro-user-callerid:8] Set("SIP/4945-00000007", "CALLERID(all)="First Last" ") in new stack
-- Executing [s@macro-user-callerid:9] GotoIf("SIP/4945-00000007", "0?limit") in new stack
-- Executing [s@macro-user-callerid:10] ExecIf("SIP/4945-00000007", "0?Set(GROUP(concurrency_limit)=4945)") in new stack
-- Executing [s@macro-user-callerid:11] ExecIf("SIP/4945-00000007", "0?Set(CHANNEL(language)=)") in new stack
-- Executing [s@macro-user-callerid:12] GotoIf("SIP/4945-00000007", "0?continue") in new stack
-- Executing [s@macro-user-callerid:13] Set("SIP/4945-00000007", "__TTL=64") in new stack
-- Executing [s@macro-user-callerid:14] GotoIf("SIP/4945-00000007", "1?continue") in new stack
-- Goto (macro-user-callerid,s,25)
-- Executing [s@macro-user-callerid:25] Set("SIP/4945-00000007", "CALLERID(number)=4945") in new stack
-- Executing [s@macro-user-callerid:26] Set("SIP/4945-00000007", "CALLERID(name)=First Last") in new stack
-- Executing [*97@from-internal:4] Macro("SIP/4945-00000007", "get-vmcontext,4945") in new stack
-- Executing [s@macro-get-vmcontext:1] Set("SIP/4945-00000007", "VMCONTEXT=default") in new stack
-- Executing [s@macro-get-vmcontext:2] GotoIf("SIP/4945-00000007", "0?200:300") in new stack
-- Goto (macro-get-vmcontext,s,300)
-- Executing [s@macro-get-vmcontext:300] NoOp("SIP/4945-00000007", "") in new stack
-- Executing [*97@from-internal:5] Set("SIP/4945-00000007", "VMBOXEXISTSSTATUS=SUCCESS") in new stack
-- Executing [*97@from-internal:6] GotoIf("SIP/4945-00000007", "1?mbexist") in new stack
-- Goto (from-internal,*97,106)
-- Executing [*97@from-internal:106] VoiceMailMain("SIP/4945-00000007", "4945@default") in new stack
-- Playing 'vm-password.gsm' (language 'en')
-- Playing 'vm-youhave.gsm' (language 'en')
-- Playing 'digits/3.gsm' (language 'en')
-- Playing 'vm-INBOX.gsm' (language 'en')
-- Playing 'vm-first.gsm' (language 'en')
== Parsing '/var/spool/asterisk/voicemail/default/4945/INBOX/msg0000.txt': == Found
-- Playing 'vm-message.gsm' (language 'en')
-- Playing '/var/spool/asterisk/voicemail/default/4945/INBOX/msg0000.slin' (language 'en')
-- Playing 'vm-extension.gsm' (language 'en')
-- Executing [h@macro-dialout-trunk:1] Macro("SIP/1007-00000004", "hangupcall,") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/1007-00000004", "1?theend") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] Hangup("SIP/1007-00000004", "") in new stack
== Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/1007-00000004' in macro 'hangupcall'
== Spawn extension (macro-dialout-trunk, s, 20) exited non-zero on 'SIP/1007-00000004' in macro 'dialout-trunk'
== Spawn extension (from-internal, 4166363472, 5) exited non-zero on 'SIP/1007-00000004'
== MixMonitor close filestream
== End MixMonitor Recording SIP/1007-00000004
-- Playing '/var/spool/asterisk/voicemail/default/84945/greet.slin' (language 'en')
== Parsing '/var/spool/asterisk/voicemail/default/4945/INBOX/msg0000.txt': == Found
-- Playing 'vm-forwardoptions.gsm' (language 'en')
-- Playing 'vm-msgsaved.gsm' (language 'en')
-- Playing 'vm-advopts.gsm' (language 'en')
-- Playing 'vm-repeat.gsm' (language 'en')
-- Playing 'vm-next.gsm' (language 'en')
== Spawn extension (from-internal, *97, 106) exited non-zero on 'SIP/4945-00000007'
-- Executing [h@from-internal:1] Hangup("SIP/4945-00000007", "") in new stack
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/4945-00000007'



mdshs

@teksavvy.com

One other thing to add is I notice that if I get a voicemail, in /var/spool/asterisk/voicemail/default/4945/INBOX it shows it there. If I forward it though to another extension, it's never in their /INBOX so it's like it just does nothing.



mdshs

@teksavvy.com

Any ideas anyone?


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