similar to: Running asterisk as non-root: Zaptel Permission Probs

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 =

Using CVS-HEAD libpri, CVS-HEAD zaptel, CVS-STABLE asterisk. Everything compiled fine. No problems loading chan_zap.so. Incomming calls to PRI work fine. Outbound is a different story: -- Executing Dial("SIP/64.72.107.4-4122fb40", "ZAP/R1d/18005551212|60") in new stack -- Called R1d/18005551212 -- Channel 0/23, span 1 got hangup Mar 13 13:19:29 WARNING[28835]:

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 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

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

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 It is out of date, and doesn't support VAD nor silence detection.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

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

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 -----

Any ideas on this warning? If I call this number, sometimes I get this error and sometimes the call goes thru fine. Why would it work sometimes? -- Executing Goto("SIP/3044-8d49", "cytel-outgoing|915124512424|1") in new stack -- Goto (cytel-outgoing,915124512424,1) -- Executing SetCIDNum("SIP/3044-8d49", "2814494000") in new stack --

I just started getting this error today (I've gotten this error befor) and its preventing me from having any incoming calls: chan_zap.c:7542 pri_dchannel: Ring requested on channel 0/2 already in use on span 1. Hanging up owner. PRI has been working fine. I didn't know anything was wrong until someone came and said their DID wasn't working. You call their DID, asterisk shows the

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

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 --

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

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

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

Why even have the ability to set callerid name/number if end offices don't honor it? For example, I have a SIP UA registered and in the sip.conf I have: callerid="Mark Mane <2815692712>" When that phone makes an outbound local call, asterisk will terminate it on PRI connected to asterisk box to Time Warner. When the called party looks at their caller id display screen

Hello. I've got 1 queue setup with 2 possible agents. Agent 1 is logged in and awaiting a call via AgentCallback. Agent 2 has not logged in. An outsider (caller A) calls in and is placed in the queue, cytelcs. Agent 1's phone rings and Agent1 and A talk. While they are talking, caller B calls in. Caller B is correctly placed in the queue and hears music, however this shows up in asterisk

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,