similar to: oh323 implementation 0.67 has call-id problem

We come into this section of the dialplan: exten => 88670333333,1,Wait(1) exten => 88670333333,n,SayUnixTime exten => 88670333333,n,NoOp(If you know the extension ...) exten => 88670333333,n,Dial(${PHONE_6003}) The caller from the GK hears only ringing, not the time. The extension 6003 rings and I can pick up, but without any voice nor video. athome*CLI> -- Executing

How do I route all the outgoing calls through a SIP gateway, this should send more than one outgoing call to the sip gateway at once. please show me the sample configurations on how to do this. my SIP gatway can accecpt direct IP traffic or SIP proxy traffc. Thank You Kanishka -------------- next part -------------- An HTML attachment was scrubbed... URL:

I've got the same problem. MusicOnHold works if I use something like: Exten => 1111,1,MusicOnHold() but if I try to answer a call and then transfer or put on hold the call, I get no music. Does anyone have any idea? Bye, Gianluca. _____ Da: Kanishka Somaratne [mailto:kani@technoportal.biz] Inviato: gioved? 17 marzo 2005 5.53 A: asterisk-users@lists.digium.com

Banging my head against a brick wall trying to get a working H.323 implementation for CVS-HEAD. (The ONLY H.323 I have had working is OH323 v0.6.5 with CVS-STABLE - see my other post regarding compile problems on OH323 for HEAD) So, I thought, lets try this wonderful chan_woomera (dubbed "H.323 for Asterisk that works!"). I get exactly the same kind of problem as I have previously had

I was under the impression that this is a good list. But maybe that isn't the case. I have asked multiple questions and have done tons of research before hand and tried to be as specific as possible. So far I haven't received any answers. All I wanted was information on getting a NT server to accecpt connections from UNIX. The documentation is to clumsy and very hard to read. The

Hi, when I receive a call from ISDN BRI (with a Sangoma A500) and I try to playback something I get the following error: **[WOOMERA]** HW DTMF supported s1c1- -- Executing [number at from-pstn:1] Answer("WOOMERA/g1/1-7b29", "") in new stack **[WOOMERA]** +++ANSWER WOOMERA/g1/1-7b29 -- Executing [number at from-pstn:2] Playback("WOOMERA/g1/1-7b29",

The update worked like a charm! Hold music is as cheesy as ever! Thanks much, this list is a life saver! Dan ------------------------------ Message: 2 Date: Fri, 18 Mar 2005 09:16:59 -0600 From: Eric Wieling <eric@fnords.org> Subject: Re: [Asterisk-Users] Redhat 9 Music on hold To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com>

On Wed, 16 Apr 2003, Ron Bramblett wrote: > Sorry about the previous post. > > It seemed no matter how detailed of a message I asked, I did not ask it > the right way so no responses. > > What I am trying to do is this. > > I have a NT server (I don't have access to it or even a user account on > it.) You must have an account to gain access to the NT server. Your

Hello, there has been a post to this list somewhere arount april which said that it is possible to use a Sangoma BRI A500 card with zaptel and asterisk bristuff. That is, without sangoma_brid and sangoma_mgd daemons and without woomera channels. Could anybody give me a short hint how to configure this? I tried wanpipe-driver + zaptel + asterisk-bristuffed, but I couldn't get zaptel to

Hi, folks! Sangoma has an informal user survey up on their home page (at http://www.sangoma.com) asking if people would use their A500 card for North American BRI, if it were supported. I encourage anyone with an interest in voice BRI in North America to vote; this information will be used for deciding whether to make the investment in developing a National ISDN driver layer for it. Benefits to

Through the generosity of our Premier sponsor, Sangoma Technologies we are proud to welcome Craig Southeren all the way from Australia. Mr. Southeren.s work has pioneered the development of open source telephony applications with his ground-breaking OpenH323 protocol stack that stood alone as the only open source VOIP software for quite some time. Today, Craig continues to raise the bar

Hello, This problem is regarding the configuration directive called 'LoginGraceTime'. Problem Description: Tests were done with OpenSSH -3.6.1p2 and 3.7.1p2 on HP-UX. sshd is started with LoginGraceTime as 1 minute.Three windows were used to initiate the ssh client.After launching two clients wait for a sometime without issuing the password so it exceeds the grace period for login.when

Hi how do i set an SIP users to make outgoing calls that is worth only $5. if they exceed $5 they can't make any calls. what i need is not a calling card, but to limit outgoing calls for SIP users depedning on a value i give. I use realtime asterisk. Thank You Kanishka -------------- next part -------------- An HTML attachment was scrubbed... URL:

how to adjust voice volume for sipura 2000 and cisco ata186?

1and1 dedicated server's service has been down for a few hours , unable to reach them by phone or email. do anyone know what is going on there ? Mario -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070731/74328b51/attachment.htm

Hi I want to create 2 groups of extensions, for example group 1 can't make outgoing calls they can only call other extensions and extensions of group 2. group 2 can call any of the extensions + they can make out going calls using our SIP server. Please let me know how to do this. I was going through the docs and I sae that I have to specify a group in zapta.conf , this is not clear please

Unsubscribe Elvin G. Nodalo -----Original Message----- From: asterisk-users-request at lists.digium.com Sent: 7/10/2013 1:00 AM To: asterisk-users at lists.digium.com Subject: asterisk-users Digest, Vol 108, Issue 14 Send asterisk-users mailing list submissions to asterisk-users at lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit

Unsubscribe Elvin G. Nodalo -----Original Message----- From: asterisk-users-request at lists.digium.com Sent: 7/10/2013 1:00 AM To: asterisk-users at lists.digium.com Subject: asterisk-users Digest, Vol 108, Issue 14 Send asterisk-users mailing list submissions to asterisk-users at lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit

are there any open source sip softphone (Window OS version )?

IPSwitchBoard Version 0.67 Release notes: CRM integration, can call a web page with callerid when there's an incoming call. You can specify the min. and max. length of the callerid. Drop any active call. Help file integrated in IPSwitchBoard. Play button for sound files. Bug fixes - thank you for all your feedback. Download IPSwitchBoard for FREE here: