Voicemail Playback Converting

Asterisk version 1.6.2.7 on CentOS 5.5

We are getting voice mail playbacks that looks like Asterisk is converting the format.

A voice mail is left

-- Accepting AUTHENTICATED call from 192.168.1.9: > requested format = ulaw, > requested prefs = (), > actual format = ulaw, > host prefs = (), > priority = mine -- Executing [*199@longdistance:1] VoiceMail("IAX2/apc-3316", "199,u") in new stack -- <IAX2/apc-3316> Playing 'vm-theperson.gsm' (language 'en') -- <IAX2/apc-3316> Playing 'digits/1.gsm' (language 'en') -- <IAX2/apc-3316> Playing 'digits/9.gsm' (language 'en') -- <IAX2/apc-3316> Playing 'digits/9.gsm' (language 'en') -- <IAX2/apc-3316> Playing 'vm-isunavail.gsm' (language 'en') -- <IAX2/apc-3316> Playing 'vm-intro.gsm' (language 'en') -- <IAX2/apc-3316> Playing 'beep.gsm' (language 'en') -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/default/199/tmp/DVoCbD format: wav49, 0x14024538 -- x=1, open writing: /var/spool/asterisk/voicemail/default/199/tmp/DVoCbD format: gsm, 0x13ff75e8 -- x=2, open writing: /var/spool/asterisk/voicemail/default/199/tmp/DVoCbD format: wav, 0x13ff60d8 -- User hung up

Then when you listen to the voice mail it tries to play back a slin file.

-- Executing [*98@iax-trunk:1] Ringing("SIP/199-00000000", "") in new stack -- Executing [*98@iax-trunk:2] Wait("SIP/199-00000000", "2") in new stack -- Executing [*98@iax-trunk:3] VoiceMailMain("SIP/199-00000000", "") in new stack -- <SIP/199-00000000> Playing 'vm-login.gsm' (language 'en') -- <SIP/199-00000000> Playing 'vm-password.gsm' (language 'en') -- <SIP/199-00000000> Playing 'vm-youhave.gsm' (language 'en') -- <SIP/199-00000000> Playing 'digits/1.gsm' (language 'en') -- <SIP/199-00000000> Playing 'vm-INBOX.gsm' (language 'en') -- <SIP/199-00000000> Playing 'vm-and.gsm' (language 'en') -- <SIP/199-00000000> Playing 'vm-first.gsm' (language 'en') == Parsing '/var/spool/asterisk/voicemail/default/199/INBOX/msg0000.txt': == Found -- <SIP/199-00000000> Playing 'vm-message.gsm' (language 'en') -- <SIP/199-00000000> Playing '/var/spool/asterisk/voicemail/default/199/INBOX/msg0000.slin' (language 'en')

Voice mail is recorded in WAV, WAV49 and GSM.

What do we need to do to have Asterisk just playback the voice mail with out converting it to slin?

It appears the voice mail is being recorded in this way but only from calls over the IAX trunk between our two Asterisk servers. Voice mail coming from calls from SIP or DAHDI sound correctly. The calls over the IAX trunk sound good, just the voice mail is broken up.

Hi Check your timing, also add transcode via slin to yes

and try setting internal timing to yes.

Ian

Ian,

Thanks for the reply.

made those changes to both Asterisk servers asterisk.conf files and restarted the services.

Still get the same results.
Voicemail left from DAHDI and SIP sound perfect.
Voicemail left from IAX are sped up messages.

Called in through a DAHDI and IAX connection and counted to 10 in the voicemail. Looking in the /var/spool/asterisk/voicemail/default/ext_num/INBOX and comparing the file size the message left from the IAX connection are considerably smaller. Also comparing the .txt files the DADHI duration shows 10 and the IAX duration shows 3.

If you pull the files over and play them on a workstation you get the same sped up glitch sound. I believe that they are being recorded this way and not a playback issue at this point.

Hi

It sounds like a packet size issue.

are all 3 files sped up ?

Ian

Ian,

Yes all three files that are recorded from an IAX trunked call are sped up.

The connection between the two Asterisk server is a 3Mb point to point connection. If someone answers the call on the other side of the IAX trunk, the conversation sounds fine, no lag or break ups. The issue only seems to be when a voice mail is recorded. Before we upgraded to 1.6.2.7 the voice mails were being recorded as expected.

Ended up saying forget it and now have a SIP trunk between the two plants for all the calls to go through. I will try the IAX trunk again when we update Asterisk again.

I was just wondering if anyone has found a solution for this besides setting up SIP trunks between two separate Asterisk boxes. We are experiencing the same issues, but we would like to use IAX trunks instead of SIP. Any help would be appreciated.