similar to: Channel Variables in a Call file?

hi, i have a new install, 1.6 2, 2 extension, but the call doesnt get thorugh: here is my sip debug outout: thx for ur help!! <asterisk-users at lists.digium.com> --- (13 headers 16 lines) --- Sending to AA.BBB.CCC.DD : 28127 (NAT) Using INVITE request as basis request - Y2QxNTg4NjE3MTZjNGMzZGM5NzE3YWY4NjAyOTYzMjk. Found user '701' for '701' Found RTP audio format 107 Found

Hello to all I'm trying to record the calls going to my queues, but asterisk creates 2 files, one with the inbound and another with the outbound sound. I know Sox should mix the 2 files automatically in the end, but this isn't happening. I have sox installed in my server. How can I force Sox to mix the files? Here is my config: queues.conf----------------------------- [general]

Hello I have an old Asterisk where I need to listen to Agent calls. So I created this code: exten => _555,1,ChanSpy(Agent) exten => _555,n,Hangup() But I always get: 2009-10-22 16:00:38 WARNING[5695]: pbx.c:1720 pbx_extension_helper: No application 'ChanSpy' for extension (default, 555, 1) It seems that Asterisk doesn't have ChanSpy enabled... is this possible? Which

Hello I'm trying to use an AGI that returns the queues status (numbers of available agents, etc ), but I'm having some problems with it (it's still very buggy). Is there any AGI repository with source code samples? Had anyone used an AGI to check queues and agents status? Thanks regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip:

List, I've got 2 queues with 10 agents in both queues. One of the agents is mainly responsible for queue_1, and the others mainly for queue_2 so i've defined the following in my queues.conf [queue_1] strategy=ringall member=>Agent/1,2 member=>Agent/2,1 member=>Agent/3,1 member=>Agent/4,1 [queue_2] strategy=ringall member=>Agent/1,1 member=>Agent/2,2

Hello I'm trying to register an Asterisk working behind Nat. Here is the trunk: register=username:password at sip.startel.pt [startel] type=peer host=sip.startel.pt username=username fromuser=username secret=password qualify=yes disallow=all allow=ulaw allow=alaw allow=gsm insecure=very port=5060 nat=yes canreinvite=yes The problem is: Asterisk is registering with its

Hello Im using Asterisk as a SIP client of Kamailio with RTPproxy. Asterisk is behind NAT. X-Lite and SNOM phones behind NAT work fine. But when I try to connect with an Asterisk behind NAT, the Asterisk client doesn't receive sound. I already tried in 2 different NATs, with no firewalls. This is my Asterisk config: [kamailio] type=peer host=xxx.xxx.xxx.xxx disallow=all allow=ulaw

Hello: I have the linux version 2.0 of x-lite downloaded. Does anyone know exactly what settings needed to reach the asterisk server on my home network? Internet ->DSL transparent bridge ->router ->asterisk ->softphone x-lite attempts to login and register, but times out. There must be some setting I'm

I wonder if Asterisk's skinny/sccp channel driver could be used as a "client" to register with a Cisco PBX. That is, along with a SIP client, say, have Asterisk and said SIP client stand in for a Cisco phone, or an IP Communicator. Anyone done this? Cheers, b. -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type:

Hello to all I'm using asterisk 1.4 and dahdi. I had everything working fine, and I could place calls through my R2 channel. But now the channel is always "RED" and Im getting this error message: TE4XXP: Version Synchronization Error! Here is my chan_dahdi.conf------------------------------ [channels] language=en context=incomingr2 signalling=mfcr2 mfcr2_variant=ar

Hi all, We have our system hosted publicly and 4 phones are connected remotely at employee's home, and when they try to do a assisted transfer to one of the employee at the main office, the call is lost. For ex: person A calls person B, person B calls person C for assisted transfer, and as soon as person B hits transfer button again to transfer person A to C, the call is lost. But in the

Hello I just updated the kernel, dahdi-linux and dahdi-tools Im also using now asterisk 1.4.26.1 And im still with a red light (not RED/YELLOW anymore): [root at catumbela ~]# /etc/rc.d/init.d/dahdi status ### Span 1: TE4/0/1 "T4XXP (PCI) Card 0 Span 1" (MASTER) HDB3/ RED 1 PRI CAS RED 2 PRI CAS RED 3 PRI CAS RED 4 PRI

Dear all on a debian amd64 i've installed (from source) asterisk 1.4.30 On the system we have in average 50 concurrent calls in queue and 40 sip members. I'm experiencing an apparently random problem: sometimes some users receive 2 calls from asterisk, apparently ignoring the ringinuse=no settings. It appears on users that are members of many queues As you can see from the log, the

I've been having a problem with Win95 clients watching a printer queue on the Samba server (called labserv2). labserv2 is a Solaris 2.5.1 machine running samba 1.9.18p4. The printing system is LPRng 3.4.2. A user can watch the printer queue from a Win95 client machine. However, when the printer queue gets backed up (usually about 12 jobs), the queue window locks. The corresponding smbd

There’s a few passes in LLVM that make heavy use of a big DenseMap, one that potentially gets filled with up to 1 entry for each instruction in the function. EarlyCSE is the best example, but Reassociate and MachineCSE have this to some degree as well (there might be others?). To put it simply: at least in my profile, EarlyCSE spends ~1/5 of its time growing DenseMaps. This is kind of… bad.

Hi! I want to know the way that calls are answer in this case... I have queue1 and queue2, one agent that receive call from both queues. queue1 <- call1 queue1 <- call2 queue2 <- call3 queue2 <- call4 In my test the agent answer calls in this order: call1,call3,call2 and call4. I think this must be in this order call1,call2, call3, call4 like a big FIFO. Its ok this behavior? Could

Curious to know if anyone's created something similar to the following, if so and you'd care to share an AGI or dialplan, much appreciated. I will be eventually write a script to answer the nTH call. (if I can't find it (why reinvent wheels). Looking to do some testing sending anywhere between 50-200 calls to a machine. I'd like a Snom/Polycom/whatever to pick up after the nTh

Ex-girlfriend logic not working in latest CVS? Incoming sip calls don't work. Anyone else seen this problem? Extension logic looks good: exten => 6153248305/_931NXXXXXXX,1,Queue(queue1); exten => 6153248305/_615NXXXXXXX,1,Queue(queue2); ;exten => 6153248305,1,Queue(queue3); show dialplan looks good: -- Added extension '6153248305' priority 1 (CID match

> On Mar 15, 2016, at 4:09 PM, Philip Reames <listmail at philipreames.com> wrote: > > > > On 03/15/2016 03:07 PM, via llvm-dev wrote: >> There’s a few passes in LLVM that make heavy use of a big DenseMap, one that potentially gets filled with up to 1 entry for each instruction in the function. EarlyCSE is the best example, but Reassociate and MachineCSE have this to

What should we use instead of DenseMap? —escha > On Mar 15, 2016, at 3:30 PM, Xinliang David Li <xinliangli at gmail.com> wrote: > > yes it makes sense. Avoid using DenseMap when the size of the map is expected to be large but can not be pre-determined. > > David > > On Tue, Mar 15, 2016 at 3:07 PM, via llvm-dev <llvm-dev at lists.llvm.org <mailto:llvm-dev at