similar to: RE: Asterisk-Users Digest, Vol 8, Issue 186

I have googled, and wiki'ed until blue. Is it possible to put T1---*----Toshiba CTX ? I have a TE405P, with one interface programmed for the T1, I am not sure how to program the 2nd port to mimick the T1 to the Toshiba. The Zapata.conf [channels] switchtype=national context=from-pstn signalling=pri_cpe usecallerid=asreceived echocancel=yes echocancelwhenbridged=no echotraining=400

Has anyone had success with the AGI STREAM FILE command with the CVS? I can't get it to work with the debian 1.0.5 package or the CVS on Redhat or Debian. It's not syntax, I'm doing that right. It doesn't give me an error when I use AGI DEBUG, it doesn't even give a response, just goes right on to the next command. I put a "SAY NUMBER 123 #" before and after

Does anyone know how to remove the browse list comments listing? In other words, in our instance, when you browse the my network places, both the resource name and comments are lengthy Samba 3.0.13. listings. Thanks,

Hello I Installed Asterisk on RedHat 9. I am currently try to configure minimum with two softphone talking to each other over the LAN. I am using X-Lite softphones from xten.com site. I defined 3 phones in sip.conf and also specifies in extensions.conf file. I am able to ring other computer but there is no voice exchange ( i can't hear any think except ring). Here is the portion of sip.conf

I have the following asterisk configuration (sip.conf) : [General] externip=82.79.81.3 localnet=192.168.10.0 localmask=255.255.255.0 [Phone1] type=friend host=dynamic nat=yes qualify=yes context=sip callerid="Phone1" <1> disallow=all allow=gsm [Phone2] type=friend host=dynamic qualify=yes context=sip callerid="Phone2" <2> disallow=all allow=gsm [Phone3]

That could be it. What I want to do is set a group of callers and have the event cause the phone to ring them in order. I will tie it to my IVR portion and thus I can make sure peole in sales get calls based on our hierarchy in the office. So if I am reading your example right the syntax is.... Exten => 501,1,Dial(SIP/PHONE1&SIP/PHONE2&SIP/PHONE3), rtf) Is that a valid way to cause

Hello again, I am using mix monitor and the majority of the sound records perfectly. I then get a "Internal Data Stream Error" near the end of the sound file. Has anyone ever seen this? I am allowing the ULAW amd ALAW codecs and an example dialplan entry is ; ; phone line phone1 exten => phone1,1,Answer() exten => phone1,2,MixMonitor(test.wav|av(0)V(0)) exten =>

Actually doing both sounds good to me. Can you explain further about ringing them all at once? Here is how I tried to make mine work and failed... {global} PHONES0=SIP/2000 PHONES1=SIP/2001 [local] exten => 6001,1,Dial(${PHONES0&PHONES1),20,trf) When I dial 6001 I see my debugger tell me that I am using the wrong syntax. Do you know the correct syntax for ringing them all at once? I

Hello! Is anybody having stability problems with if_re under FreeBSD 6.3-stable? I know about PR kern/118719[1] but it doesn't look like the problem I'm having - at least my machine doesn't panic. My machine[2] runs FreeBSD 6.3-stable / amd64: root@kg-vm# uname -a FreeBSD kg-vm.kg4.no 6.3-STABLE FreeBSD 6.3-STABLE #1: Sun Jan 27 02:10:15 CET 2008

I have read every thread, I have Redhat 9, Asterisk 1.0.6, 2 T1 lines connected via TE405P. Everything works great, except MOH. I added an exten with MusicOnHold(30), and it plays just fine. Conferences have music when no one is in. I have SIP phones. When I place a call on hold, the CLI give no indication the call is on hold. I have set musiconhold(default) everywhere, removed it from everywhere,

I'm new in hire so Hello to everyone! I'm beginner user of Asterisk CVS-HEAD-10/01/04-14:31:34 . I just installed it on my Mandrake Linux 10.0 kernel 2.6.3-4mdk with sample configuration (used make samples). I would like to make phone connections between X-Lite (SIP) installed on computers in LAN. How to make this? I was reading manual, and tried to make changes in sip.conf but this all

Thanks Eric, the hold music is back, and cheesy as ever! This list is a live 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> Message-ID:

Hi can i set up a group of SIP users and forward a call to it? I am looking for a group, not for a queue. I won't listen any musinc on hold, and i won't that someone has to pay if nobody of the user's in the group accept the call. Can i do that? Thanks to all

This updates Ruby to 1.8.6 patch level 111 (lots of bug fixes since 1.8.6 was released). It also updates most of the included extensions to their latest versions (see the change log), and adds FastCGI and ruby-fcgi support. These two new extensions allow you run your Rails application under IIS7! I''d like to thank Luis Lavena for taking charge of the One-Click Ruby Installer project and

Hi people, I have a Cisco ATA 186 which understands only the SCCP protocoll, therefore I am a pure beginner and I hope that anybody of you could help me. How will I configure the ATA which has 2 analog ports? For any support I would kindly thank you Tamer

http://bugzilla.mindrot.org/show_bug.cgi?id=186 Summary: Build failure against openssl-0.9.5a Product: Portable OpenSSH Version: 3.1p1 Platform: ix86 OS/Version: Linux Status: NEW Severity: normal Priority: P2 Component: Documentation AssignedTo: openssh-unix-dev at mindrot.org ReportedBy:

http://bugzilla.mindrot.org/show_bug.cgi?id=186 markus at openbsd.org changed: What |Removed |Added ---------------------------------------------------------------------------- Status|NEW |RESOLVED Resolution| |DUPLICATE ------- Additional Comments From markus at openbsd.org 2002-03-27

https://bugzilla.mindrot.org/show_bug.cgi?id=1647 Summary: Implement FIPS 186-3 for DSA keys Product: Portable OpenSSH Version: 5.2p1 Platform: Other OS/Version: All Status: NEW Severity: enhancement Priority: P2 Component: ssh-keygen AssignedTo: unassigned-bugs at mindrot.org ReportedBy:

https://bugzilla.mindrot.org/show_bug.cgi?id=1647 Damien Miller <djm at mindrot.org> changed: What |Removed |Added ---------------------------------------------------------------------------- Status|NEW |RESOLVED Resolution|--- |WONTFIX CC|

https://bugzilla.mindrot.org/show_bug.cgi?id=1647 Damien Miller <djm at mindrot.org> changed: What |Removed |Added ---------------------------------------------------------------------------- Status|RESOLVED |CLOSED --- Comment #5 from Damien Miller <djm at mindrot.org> --- Close all resolved bugs after 7.3p1 release