similar to: Getting the right DST in CDR

Hi guys, i just came to know that CDR(dst) field is set to current extension instead of the dialed no. i need to set it to DNID because our every user has 5 dids and i want to show the caller at the end of the month which numbers he dialed for every call, along with other cdr info. Our rating depends on the dialed number also. here is my extensions.conf exten=> 1212,1,Dial(SIP/rizwan)

Here is the setup: SIP/3044 -> SetCallerID(5551212) -> Call out PRI The CDR shows a src of 5551212. That is a lie! The src of that call was not 5551212, the source was 3044! The "translated source" of that call was 5551212. How can I get "real" source of this call and not some faky nonsense? The "reason" behind using the SetCallerID is because if I

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]:

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 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

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 It is out of date, and doesn't support VAD nor silence detection.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

This has been briefly talked about before but no real resolution was found. If someone dials a DID, it comes to asterisk via PRI. I translate that DID into an extension by means of: exten => 7145254474,1,Dial(SIP/3022,30) The destination number of the caller is '7145254474' yet for some reason the CDR shows a destination of 3022. 3022 was 'not' the destination dialed, it was

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 =

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

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 --

A simple question but one I can't seem to find easily... I have 90 or so DIDs. For all outbound calls, I edit the callerid so that it will always read out main line's number. This poses a problem though, because the CDR detail isn't written until after everything is done. So when you look at the detail, the number you are calling is over written with this new caller id for some

I am using VoiceOne http://voiceone.it/ as my management interface. I am not 100% sure when it started, but my CDR is now full of "s" as the DST instead of the actual dialed number. As I understand it - it is because it is being recorded in the CDR while in a macro (as below). Is there any work around so that I can record the actual dialed number? [macro-dialout] exten =

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 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 have a PRI comming into each of 2 buildings. How do I redirect an incomming call on PRI_A of particular DIDs to arrive at PRI_B instead? Thanks, John

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 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: