similar to: Outgoing Context being mistake for dialplan?

Hi, I have some really disturbing problems with Asterisk 1.4.1 and my dialplan for outgoing calls. First of all i switched some weeks ago from * 1.2 (bristuffed version ) to this version and in my opinion a lot more troubles arose.... For outgoing calls I use a Digium B410P with chan_misdn (before a Junghanns QuadBRI with zap). 1) So first thing is, that a user reports to me (highly

Hi *, now for a long time i am trying to set the outgoing callerid, without luck. I am here in Germany, my asterisk has a pri interface connected to a PMX installed by Telekom. All telephone calls are preselected to EcoVoice. I am using asterisk 1.2.7.1, zaptel 1.2.5 and libpri 1.2.2. A week ago we tried with a device able to simulate a telephone system so send out a callerid, and that

Per my conversation below with digium, are there any legal alternatives to digium's G729? It is out of date, and doesn't support VAD nor silence detection. Digium has stated that they have no plans to update it anytime soon. VAD/Silence is a big deal with major carriers and we are having to fight a battle to get them to make special arrangements to turn off VAD/Silence in their

Even though they make the cards and advertise that they support data modes, digium won't support data mode on the $500 card they sold to me, so I must turn to the list. Has anyone configured a T100P to use a T1 strictly as bandwidth? Is there a HOWTO somewhere? Wiki has nothing I could find. I've got plently of public IPs I can assign to it but don't know how. Thanks, Matthew -----

Hi, Let's see if any of the questions gets answered or at least I get ponte dto something that can help me. I followed this wiki: http://wiki.samba.org/index.php/Samba4/HOWTO/Join_a_domain_as_a_DC#Getting_ready_for_joining_Samba_as_a_DC_to_an_existing_domain I have my S4 domain running, I compiled and installed another S4 to replicate the first server and joined successfully to the

All of the stuff I've googled for and read on wiki all relate to "Outlook". Has anyone been successful in getting "Outlook Express" to do click to dial? -Matthew -- ------------------------------------------------------------------------ Matthew Boehm, IT Director Cypress Telecommunications mboehm@cytelcom.com 3838 N. Sam Houston

All, I get this whenever trying to dial to a peer when the peer registered to another server. I'm basically trying to use realtime to check for the peer and dial it. Mar 24 09:16:47 VERBOSE[4527]: -- Executing Dial("SIP/brak-f69f", "IAX2/brak-test/107") in new stack Mar 24 09:16:47 DEBUG[4527]: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_users WHERE name =

We have asterisk running on a quad processor dell. The kernel has been compiled with SMP. However, asterisk seems to only use 1 processor. 3 of the 4 always stay at 100% idle. Is it pointless to have a multi-proc machine? I was going to buy a new dual 3.6Ghz Xeon server but if nothing will take advantage of the other proc... Perhaps my conception of multi-proc/threaded is warped. If asterisk is

I don't know...now I have a _X. in my CDR. -----Original Message----- From: Matthew Boehm [mailto:mboehm@cytelcom.com] Sent: Saturday, March 12, 2005 8:05 PM To: Asterisk Developers Mailing List <asterisk-dev@lists.digium.com> Subject: Re: [Asterisk-Dev] SetVarCDR You must have some fux0red config 'cause using _X. works fine here. I haven't had an 's' in my CDRs for

Hey gang, There must be any easy solution for this but my mind is frazzled on compiling 2.4 with RTC as module. Bleh. Currently extension 9000 is our VoicemailMain(@company) line. Some employee's are complaining that the old system was better because you didn't have to enter your mailbox number and that instead the old system took you right to it. I figured there was something similar

I tried putting this as the ata's dailplan: *St4-|#St4-|9|^9t4>$.- this is sip.conf [ata2001] type=friend username=ata2001 secret=SoMeSeCrEt host=dynamic context=fromata canreinvite=no and this in extensions.conf [fromata] ignorepat => 9 exten => _91700NXXXXXX,1,Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel) exten =>

Hey gang, We have about 30 Cisco 79XX phones all running latest 7.5 SIP. We are also using all Cisco Switches and Routers. Everything works great except that when you reboot a phone it takes like 3-5 minutes for it to come up. The phones spend tons of time 'Configuring VLAN..' We don't run any VLANs. Is there some way to skip this? In the 'Network Settings' I have

I've read the Wiki on using asterisk's built-in transfer options (#8 and #6). They work fine but how does one cancle an attended transfer? Example: I have person on phone, I hit #6 to being att-transfer. I enter Sally's extension. I let it ring for a few seconds. Sally never picks up but her voicemail does. How do I hangup her voicemail and resume the previous call? The example on the

Hey guys, I sent an email to the list at 2:57PM central. I just now see it on the list, and its 3:23PM. Anyone else experience this? I am sending this email at 3:24PM central. Lets see when it gets posted to the list. -Matthew

This is frustrating. Asterisk has crashed now twice today and neither crash has produced a core file. My ulimit is unlimited. I'm using safe_asterisk so asterisk is restarting immediatly, but how the hell am I suposed to find out wtf happened with no core file? Debug log doesn't say anything either. AGRHHHHHHHH -Matthew --

Hi Richard, Thank you for the reply Correct, I did mean 13.15. I set no optimize and better backtrace through "make menuselect" and the output is now [Aug 28 21:41:16] ERROR[17171][C-0000392d]: frame.c:343 ast_frdup: FRACK!, Failed assertion Excessive refcount 100000 reached on ao2 object 0x21962b0 (0) Got 26 backtrace records #0: [0x61923f] main/utils.c:2475

http://bugs.freedesktop.org/show_bug.cgi?id=25965 Summary: no overlay with kms Product: xorg Version: 7.5 Platform: Other OS/Version: All Status: NEW Severity: normal Priority: medium Component: Driver/nouveau AssignedTo: nouveau at lists.freedesktop.org ReportedBy: castet.matthieu at

Hey gang, I'm trying to setup the ability to dial a SIP user via their email address. I'm using SJPhone as my tester UA, but most clients will be using XTen Pro. I added an SRV DNS entry into our zone, and it returns: ; <<>> DiG 9.2.1 <<>> SRV _sip._udp.cytelcom.com ;; global options: printcmd ;; Got answer: ;; ->>HEADER<<- opcode: QUERY, status:

Lets say that I have 9 phones with extensions 1 thru 9. (All SIP) Parking extension is set to 700. When I try to park a call to 700 as soon as I press the 7 key (to start typing 700) the person at extension 7's phone rings. So instead of transferring to extension 700, it went to 7. How can I slow down the response time of * to let me get to 700? Thanks, Matthew

Hello: I found a reference to the application SetVarCDR in the following post but I don't seem to have this available to me in my version of *. HYPERLINK "http://lists.digium.com/pipermail/asterisk-cvs/2005-February/005337.html"http://lists.digium.com/pipermail/asterisk-cvs/2005-February/005337.html My version of * is CVS-HEAD-03/10/05-18:42:35 I would like to change the