similar to: RE: [Asterisk-Dev] SetVarCDR

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

I recently upgraded my Asterisk system to the latest CVS-HEAD Asterisk CVS HEAD built by bill@localhost on a i686 running Linux on 2005-10-12 13:34:09 UTC Ever since this upgrade, the system is jumping n+101 if it gets a busy on a Dial command, it is now proceeding to the next priority (n+1) Has something changed with this? Is there a way to change it back? Thanks, Bill --------------

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 =

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 It is out of date, and doesn't support VAD nor silence detection.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

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

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 -----

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

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

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

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

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

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 --

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 all, I’m running Shorewall 2.2.1 on linux kernel 2.6.10 with iptables 1.2.11. I recently ran a nessus scan of my firewall from a machine outside of the firewall and the nessus report told me that there are some ports open that I did not specify to be open. The ports are 32772/udp, 123/udp, 111/tcp, 32772/udp, and 53/udp. Why are these ports open when I did NOT specify them to be open

OK. I just downloaded asterisk-1.0.3.tar.gz and did a 'cvs co -r v1-0 asterisk' into 2 seperate directories. I then did 'diff -ur asterisk-cvs/ asterisk-1.0.3/' and there were source code line differences between the two. Some code that was in asterisk-cvs wasn't in asterisk-1.0.3 and vice versa. Which of those is "the most recent"? If someone wants to use cvs to

Hi, On Thu, Nov 21, 2013 at 8:16 PM, Richard Sandiford < rsandifo at linux.vnet.ibm.com> wrote: > İsmail Dönmez <ismail at donmez.ws> writes: > > Using openSUSE 13.1 on s390x machine I get two new regressions with llvm > > 3.4rc1: > > Hmm, I don't see this locally. Just to rule out one possibility, > which compiler are you using to build? Do you see the

We should find a code owner for the scalarizer (lib/Transforms/Scalar/Scalarizer.cpp). I nominate Richard Sandiford, who added it in r195471. Let me know what you think. Thanks, Hans

Hey gang, We currently have a class 3 switch (CSX) that..well..it sucks. It does terrible CDR writes, doesn't support LCR, the list goes on and on. We want to replace this with several asterisk boxes each running one or two 4 port PRI cards. The problem is: I can plug in 20 PRI lines into the CSX (from PSTN) and have 1 come from CSX into asterisk. If 1 call comes in on each of the 20 pris,

Richard Sandiford <rsandifo at linux.vnet.ibm.com> writes: > Are you worried that adding it to PMB will increase compile time? > The pass exits very early for any target that doesn't opt-in to doing > scalarisation at the IR level, without even looking at the function. As an alternative, adding Scalarizer and InstCombine passes to SystemZPassConfig::addIRPasses() would probably

I met a strange SIP problem recently. In an ordinary procedure, when asterisk loads sip module, a series of functions are called sequentially: load_module()->restart_monitor()->ast_pthread_create()->pthread_create()->do_monitor() However in my system, pthread_create() failed to create a child thread to execute do_monitor(), (though pthread_create() returns a successful signal to