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...011,1,Macro(std) exten => 6012,1,Macro(std) ... exten => 6503,1,Answer ; directory dialing exten => 6503,2,Wait(1) exten => 6503,3,Directory(default|voip) ... voicemail.conf contains:- [general] format=wav attach=yes [default] 6010 => 1234,Customer Services,xxx@linguaphone.co.uk 6011 => 1234,user one,xxx@linguaphone.co.uk 6012 => 1234,user two,xxx@linguaphone.co.uk ... Any ideas?

I have about 30 GXP-2000 phones running 1.0.1.9 which have all been configured using the provisioning feature so the configuration is all identical. The problem I am having is that they randomly seem to stop registering with asterisk. When they stop registering they can still make calls but oviously asterisk cannot ring the phone so all incoming calls go to voicemail. Has anyone else had similar

...; An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk-users/attachments/20070315/e9cae81c/attachment-0001.htm > > ------------------------------ > > Message: 16 > Date: Thu, 15 Mar 2007 10:35:16 +0000 > From: Gareth Blades <list-asterisk@linguaphone.co.uk > > Subject: Re: [asterisk-users] busy/hangup/answer detection in PRI E1 > channels > To: Asterisk Users Mailing List - Non-Commercial Discussion > < asterisk-users@lists.digium.com> > Message-ID: > <1173954916.31406.2.camel@gblades-suse.ling...

Hello, I'm running asterisk-1.0.6 on a centos3.4 box. I'm still in testing phase and so far everything is running smoothly. I'm now trying to play a soundfile or an mp3file using 'MP3Player', 'Playback' or the 'Background' commands, but don't get any sound. The logfile says: -- Executing BackGround("SIP/joa-9def", "tt-weasels") in

Here is what I am attempting to do (which works well on Cisco Call Manger). I have some 7960's that have multiple lines on them. The second line specifically is a "helpdesk" line that is shared among multiple phones. Here is how I am making that line ring on multiple phones, maybe you have other suggestions: exten => 135,1,Dial(SIP/135@100&SIP/135@101,20,rt) So this

I have found out that the reason why my call transfers are not working when using the IAX protocol is because Asterisk is performing a native bridge. If I force the user of one of the clients to use a different codec so that Asterisk is unable to do a native transfer then it works. How can I disable native bridge for IAX calls? I know for SIP you can put 'canreinvite=no' but this does

Just noticed that I occasionally get these messages:- Apr 12 09:27:03 WARNING[11390] chan_iax2.c: chan_iax2: ast_sched_runq ran 281 scheduled tasks all at once Apr 13 09:13:18 WARNING[11390] chan_iax2.c: chan_iax2: ast_sched_runq ran 1987 scheduled tasks all at once Apr 13 12:47:56 WARNING[11390] chan_iax2.c: chan_iax2: ast_sched_runq ran 1804 scheduled tasks all at once Are they anything to be

I am having problems getting any form of call transfer working. I have reconfigured blind transfers to be #1 and assisted transfers to be *2 but these are not working. Looking at the wiki (http://www.voip-info.org/wiki-Asterisk+cmd+Transfer) it it does not mention IAX so I assume I have to use the native IAX transfer supported by Diax? I have tried using Diax but am getting a problem that after

I have a Digicom TE110p card installed in our exchange. I have compiled and installed libpri, zaptel and recompiled and installed asterisk. I have configured udev as I am running Fedora Core 3. The problem that I have is that when zaptel is not running everything works fine. However when I start zaptel (service start zaptel) then I can make normal calls ok but the 'Playback()'

I have loaded festival-1.4.3 patched with the 1.4.3.diff file. Festival source is in /usr/src/festival dir. When I try to use it I get this from asterisk: -- Executing Answer("SIP/chad-57a4", "") in new stack -- Executing Festival("SIP/chad-57a4", ""I am talking"") in new stack == Parsing '/etc/asterisk/festival.conf': Found

List, The past few days the asterisk service on my server has crashed several times. I have had it running for months and have made no changes to it. When it crashes, I am unable to make calls or gain access to the CLI. The service has been stopped. If I try to start it again (service asterisk start), it will start and run for a few seconds then crash again. After a reboot, it will run

I am trying to use a the SIP tapi from www.enum.at <http://www.enum.at/> . This works fine from all kinds of applications which support TAPI, like outlook and Dialer Pro. However when making tapi controlled calls, the signaling to and from PSTN seems to fail. I have used the digium hardware ISDN PRI boards, but also a SIP gateway. Both result in a audio message from asterisk

If we get an incoming call I can edit the callerID provided to add the leading '90' and set the name so that sales calls can be identified according to the number called. If however the callerID is unavailable then setting the callerID name or number fails (it shows as unavailable on the phone). This is the call log from such an incoming call without callerID. == Spawn extension (voip,