similar to: Agent Privacy - chan_local

Hi, I am upgrading from Asterisk 1.4 to 12.4. I am able to authenticate the user and call AgentLogin. But after that when I call AgentRequest I keep getting Agent '1234' is busy. If I put a delay of 5 second or more before calling AgentRequest then it works most of the times. Here's my dialplan: [login] exten => s,1,Background(thank-you-for-calling) same =>

I'm having some difficulty with chan_local dial requests. It seems that when a chan_local call is picked up, that the native bridge "pops" the environment back to the settings of the original call. This is unexpected and leads to very frustrating results. My example below is a very distilled sample of a much more complex dialplan problem I'm having with chan_local, but it

Hi, Asterisk 1.4 Working (jitter buffers created as expected): ZAP -> SIP SIP -> ZAP Not working (no jitter buffers created): SIP -> chan_local (with /nj) -> ZAP SIP -> chan_local (with /j) -> ZAP SIP -> chan_local (with no flags) -> ZAP I have this in zapata.conf: jbenable=yes jbforce=no jbimpl=fixed jbmaxsize=300 Is there something I haven't tried that will make

Happened twice. There might also be a race condition and some bad pointers in chan_local.locals_show. First the segfault. CLI> show locals <unowned> -- 6001@default Segmentation fault (core dumped) [root@mars asterisk]# ll -tr total 22260 [...] Loaded symbols for /usr/lib/asterisk/modules/chan_local.so #0 __pthread_mutex_lock (mutex=0x5d8) at mutex.c:99 99 mutex.c: No such file

I had my asterisk server working fine with FWD as a SIP provider, so I now added a second SIP provider (voctel). The addition to my sip.conf file is almost identical to FWD, however, asterisk now generates lots of debug messages for some strange reason! In particular, the line "##### Testing 127.0.0.1 with 172.31.0.0" shows up every two seconds! (See my log below). If I comment out

Hi I?m a new user and I do test with my hardware . I have a x100p and telephone vozip. And when I run this command asterisk ?vvvvc for to test it . My computer show it ?warning? [chan_iax.so] => (Inter Asterisk eXchange) == Manager registered action IAX1peers == Parsing '/etc/asterisk/iax1.conf': Not found (No such file or directory) Apr 2 07:45:12 ERROR[16384]:

Hmm i found the problem... I?m using a Grandstream BT100. The transfer just works in a queue if I first acknowledged the call using the # key, and then press the TRANSFER key in the Grandstream. In the asterisk console I receive a: -- SIP/4002-4563 acknowledged Then I can transfer the call... Weird because i?m using ackcall=NO in agents.conf ... Diego Magalh?es diego@redetaho.com.br +55 24

Hi folks, I'm running asterisk 1.2.10 and I need to use agent dynamic login. I read some doc and follow some tutorials but the agents can't login into the queue. Asterisk ask to me to dial the password agent and after this, it doesn't do nothing ( it doesn't tell login ok or login incorrect..). In the * console if I do show agents, any agent are logged. Any help will be

Hi, If agents is already logged in via AgentLogin() and users dialled extension 300 which will be placed in Queue(support-queue). How to find out which agent is available I can put their Agent id in AgentRequest() ? If this is not a good approach then how it should be done? Agent should automatically get next call when he/she is available. extensions.conf [LocalSets] exten =>

I am migrating my app to Asterisk12 and pjsip, but I cannot find chan_local, what happened?

Hi, I am using Asterisk 1.2.9.1, with chan_h323. The problem I am coming across is when trying to bridge an incoming H323 call with another H323 call: phone1 dials into asterisk with H323, for extension 111 in asterisk: exten => 111, 1, Dial(chan_h323, H323/111@phone2) (in my extensions.conf the syntax is good ... this is no). I can see how the first call is partially processed, then the

Hi all, my local channel seems to be not working properly. im doing this: exten=> s,1,Dial(Local/123@users,,Tt) some times it rings the phone at extension 123, and sometimes it doesn`t. When it doesnt, it actually displays a msg that it could not find that extension. [May 1 16:54:02] NOTICE[4658]: chan_local.c:563 local_alloc: No such extension/context 12129339038@users creating local

Hi All; Why we use the agent password when we configure the agent in the agents.conf if the agent login by dialing the number configured in the extensions.conf? example: exten => 28, 1, AgentLogin(1001) I know that agent username is used to assign the agent to the queue, but when we use the agent password? Regards Bilal

Which kind of transfer do you use? Try using the # transfer. Hope that helps.. Guido Hecken -----Urspr?ngliche Nachricht----- Von: Diego Magalh?es [mailto:diego@redetaho.com.br] Gesendet: Mittwoch, 2. Februar 2005 17:21 An: asterisk-users@lists.digium.com Betreff: [Asterisk-Users] AgentLogin / AgentCallbackLogin transfer problem Hello guys, I?m running Asterisk CVS-HEAD-02/01/05-12:22:46 and

I've been running into an issue where chan_agent gets stuck and all queues stop working. Here's a show channels from when it's stuck: Channel Location State Application(Data) SIP/56-be24 s@macro-stdexten:10 Ring Dial(Agent/19|50|tw) Local/*14@agentlogin *14@agentloginoff:1 Up AgentCallbackLogin() Local/*14@agentlogin *14@agentloginoff:1

Hello guys, I?m running Asterisk CVS-HEAD-02/01/05-12:22:46 and having a problem with call transfers using the cmds AgentCallBackLogin and AgentLogin First Case (using cmd AgentCallbacklogin): When the incoming call comes and enters the queue, the agent logged in answer the call. But when I try to transfer this call to another agent, the incoming call is dropped. I don?t receive any error

I just thought i'd post a small sample that uses the new chan_local to show one way of doing variable callfwding This sample extension.conf uses's the ast DB to store a users current extension, in a db family of CallFWD and the unique Key is based on the current channel the user is assigned. In the globals var section each key is hardcoded EXT1, EXT2 this is used in the [incoming] context

http://bugs.digium.com/bug_view_page.php?bug_id=0001693 This patch adds a lot of options for AgentLogin/AgentCallbackLogin Please test and respond in the bug tracker! /O ------------------------------------------------------------------------------------- "This patch adds quite a few new features into __login_exec () of channels/chan_agent.c for AgentLogin() and AgentCallbackLogin(). Only

Hello I just coding a AGI script for billing. - For external calls, I pass the call directly on a trunk. I do : Dial(trunk1/extension) -> OK ! - For internal calls (shortcode, others users ...) I am Dial(Local/extension at context/n) The problem is that through chan_local.so, I sound as it cut! Example if I call the voicemail ... "You have No messa ..." or "You have

Hi, I've recently upgraded from Asterisk 1.4.22 to 1.4.29, and I'm now having a problem with Originate and chan_local. I'm using the following Manager API action to originate a call: Action: originate Priority: 1 Context: trunk Callerid: 100 Channel: Local/100 at callback/n Exten: 123456789 Variable: USERFIELD=127.0.0.1|USEREXT=123456789 WaitTime: 30 This is intended to first call