similar to: asterisk voicemail question

Hello, How to reproduce this bug (?) : 1. register sipura spa2 with 2 lines on asterisk. 2. use first line to call somewhere. 3. while using first line try to call from second line somewhere else in 3 step i hear fast busy tones on second line and asterisk console gives me this short error: Mar 27 07:01:43 NOTICE[29656]: chan_sip.c:3629 process_sdp: No compatible codecs! My sipura adapter

Hello list, is there any open-source software that recodes g729 sound files to wav sound files ? The only way (at least) to do such transformation is with interactive form on: http://www.asteriskguru.com/audio_conversion.php Tofik Suleymanov

Hello folks, firstly, thank you for your useful and fast answers ! Is there anybody using D-Link SIP phones ? Are D-Link SIP phones ok to install in production environment ? Give your comments please. Tofik Suleymanov

Did anybody know, Is it possible to establish a ISDN DIAL up Connection and Analog Dial up Connection (V90) trough asterisk with Digium TE405? Thanks a lot for help. Nico Giefing -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060328/3b657058/attachment.htm

Hi to all, I already setup asterisk on REDhat 9.0 linux machine. I will have 4 physical phone lines and 10 IP phones for it to use. I have a network setup already. Is getting TDM400P - 4port FXO from digium enough to start? Do I need anything else? Thank you

I have a user, running CVS a/o 11/23/03, who has complained about "phantom" messages showing up days or even weeks after she has deleted them. So I asked her to let me know when it happened again, and she called a few minutes ago. The directory listing below shows a listing of the /var/spool/asterisk/voicemail/default/XXXX/Old directory, and to my surprise the messages are indeed

I'm using asterisk 1.2.14 When asterisk stores voicemail messages in /var/spool/asterisk/voicemail/default/EXTENSION/INBOX files are created with: -rwx------ 1 asterisk web-aster 6690 Apr 17 16:08 msg0002.WAV -rwx------ 1 asterisk web-aster 6732 Apr 17 16:08 msg0002.gsm -rw------- 1 asterisk web-aster 274 Apr 17 16:08 msg0002.txt -rwx------ 1 asterisk web-aster 65324 Apr 17 16:08

Hi, I'm having trouble getting asterisk to report MWI to a Cisco CCME. I record a message in mailbox 29, but the subsequent MWI notifications I see continue to report no messages waiting. Are they reporting for the wrong mailbox? Is there some other option I have to set or change? I'm running asterisk-1.4.22 Since the mailbox is in [home] in voicemail.conf, I've tried things like

Hello everybody, I was trying to install a web interface to my Voice Mail, Vmail.cgi I can log on it, list messages, but no play with the following error msg; "Hrm, can't seem to open /var/spool/asterisk/voicemail/default/234/INBOX/msg0001.WAV" Remark: playing the message msg0001.WAV directly OK Any smart guy up there could help ? Thanks, --------------------------------- Do

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Tofik Suleymanov wrote: > Steve Totaro wrote: > >> Give idefisk a try. It works very well for me, its free, and does >> not crash all the time like Cubix (formerly Firefly). >> >> > > Hello Steve, > > As far as i know 'idefisk' is a softphone, but i need a hardware phone. > Thank you for reply. > > Tofik Suleymanov > Oooops, sorry its

1.6 UPGRADE.txt: > * If you use any interface for modifying voicemail aside from the built in > dialplan applications, then the option "pollmailboxes" *must* be set in > voicemail.conf for message waiting indication (MWI) to work properly. This > is because Voicemail notification is now event based instead of polling > based. The channel drivers are no longer

Hi All, I have an issue with the ODBC voicemail storage option with asterisk. All appears to work fine, however, I get several sql execute warnings. I was wondering if anyone out there could help me get to the bottom of what is causing this and how I could possibly go about rectifying it. The warning message we are getting is as follows: WARNING[30115]: app_voicemail.c:1280 delete_file: SQL

On Wed, Jan 15, 2020 at 07:04:06PM +0100, Matthias Tafelmeier wrote: > > Hello, > > ran into oddish glitch on fedora 30 cloud image with tooling based on > libvirt-python onpython3.7. *qemu-img *is installed though. Could anyone > have a look. I don't see qemu-img installed there. You've requested 'qemu-kvm' and 'qemu-system-x86' which provide the

Hi, We can't read the messages in our mailbox always getting -- <SIP/tootaiAUDIO-00000001> Playing '/var/spool/asterisk/voicemail/default/100/Old/msg0002.slin' (language 'fr') [Oct 11 13:24:50] WARNING[26778]: app_voicemail.c:7802 play_message: Playback of message /var/spool/asterisk/voicemail/default/100/Old/msg0002 failed As you see Asterisk try to read

Does anyone have a quick and dirty script for defragmenting mailboxes? i.e.: -rwx------ 1 root root 80553 Oct 20 16:27 msg0000.gsm -rw-r--r-- 1 root root 218 Oct 20 16:27 msg0000.txt -rwx------ 1 root root 781164 Oct 20 16:27 msg0000.wav -rwx------ 1 root root 79360 Oct 20 16:27 msg0000.WAV -rwx------ 1 root root 7260 Oct

I have installed asterisk in a 4.11 RELEASE FreeBSD, and we are using two Zoom X5v with SIP and works fine, we can call each other and this is OK ----------extensions.conf---------- [general] static=yes writeprotect=no [sip] exten => 123,1,Dial(SIP/123,20) exten => 123,2,Voicemail(u${EXTEN}) exten => 123,3,Hangup exten => 123,103,Voicemail(b${EXTEN}) exten => 123,104,Hangup

Does any one knows of an Windows based SIP video phone???... Thanks... -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of asterisk-users-request@lists.digium.com Sent: Tuesday, January 11, 2005 9:27 AM To: asterisk-users@lists.digium.com Subject: Asterisk-Users Digest, Vol 6, Issue 142 Send Asterisk-Users mailing

I'm going to be hooking FXS lines on a TDM400 to a PBX which doesn't drop line voltage at the end of a call, so I'm going to have to use busy detection. A few questions - The tones are taken from the tones specified by the zone in zaptel.conf, right? Which tones cause hangup? The PBX may not use the national standard tones. Does anyone have any suggestion for how I can

I am trying to configure a VegaStream 50 FXO to work with asterisk. The problem that I am having is that the VegaStream does not support incoming registration from asterisk. VegaStream only allows outbound registration. My question is does asterisk allow incoming registration from an FXO? If yes how? Or better yet, has anybody been able to make the VegaStream FXO work with asterisk? According