search for: chan_local

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

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 ..." The sound stops but the call continues. Please help! Debian 5.0 - Asterisk 1.6.2.6-1 Mickael. -------------- next part -------------- An HTML attachment was scrubbed... URL: h...

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 d...

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...

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 this work or will I have to change my dialplan so it do...

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 extension 100 in the callb...

...l, 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 channel [May 1 16:54:02] WARNING[4658]: app_dial.c:1090 dial_exec_full: Unable to create channel of type 'Local' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) +++++its a lie, the user...

...' I have tried with both phones individually, and both are "asterisk-compatible" with H323. Bridging works if the originating call is SIP, for example. But if I try H323 with H323, it's a nono. Am I doing something wrong? do I need to set up some parameter? I thought about using chan_local, but I came across this: *CLI> -- Executing Dial("H323/ip$192.168.1.100:1940/4096", "local/811@to_WAVE_from_incoming_SIP/n") in new stack May 7 11:31:47 WARNING[860]: channel.c:2512 ast_request: No translator path exists for channel type local (native -1) to -2033656 May...

...solve a problem I have with agents hanging up on callers before they even talk to them (caused by agents dropping their handset or something.) What I want is something like AgentLogin() where the agent has to press '1' to accept the call. Does anyone know how to get this to work with chan_local ? Thanks! Robert

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 u...

...pressing # -- <Zap/1-1> Playing 'auth-thankyou' (language 'en') -- <Zap/1-1> Playing 'followme/pls-hold-while-try' (language 'en') -- Started music on hold, class 'default', on channel 'Zap/1-1' [Jan 25 11:34:32] NOTICE[23725]: chan_local.c:571 local_alloc: No such extension/context FM1 at default creating local channel [Jan 25 11:34:32] WARNING[23725]: app_followme.c:863 findmeexec: Unable to allocate a channel for Local/FM1/2000 at longdistance cause: Unknown [Jan 25 11:34:32] NOTICE[23725]: chan_local.c:571 local_alloc: No...

I have a confbridge setup that feeds the conference from the ALSA microphone input (this is the conference leader) and then uses app_ices to send the conference audio to icecast. I start the conference leader like this: console dial 1000_admin at conferences I join the ices user to the confbridge with a call file: Channel: Local/1000 at conferences MaxRetries: 2 RetryTime: 60 WaitTime: 30

...onf" are some commented lines just above the feature map that said: ; Note that the DTMF features listed below only work when two channels have answered and are bridged together. ; They can not be used while the remote party is ringing or in progress. If you require this feature you can use ; chan_local in combination with Answer to accomplish it. But I don't have a clue how to use chan_local in combination with Answer to accomplish it. Does anybody knows how to do that???, with the transfer button preferably... Thanks a lot... -- Raul Linux Counter #156439 -------------- next part ------...

I am using Asterisk 1.4.4 on a CentOS 5 machine for a small call center with 6 agents. I am using realtime for queues and sip and I am also trying to use realtime static to load agents.conf. The only problem I am having is that no agents are loaded when I start Asterisk. I have to manually do a "module reload chan_agent.so" so the agents get loaded from the database. Obviously

...ch.so [CC] chan_agent.c -> chan_agent.o [LD] chan_agent.o -> chan_agent.so [CC] chan_iax2.c -> chan_iax2.o [CC] iax2-parser.c -> iax2-parser.o [CC] iax2-provision.c -> iax2-provision.o [LD] chan_iax2.o iax2-parser.o iax2-provision.o -> chan_iax2.so [CC] chan_local.c -> chan_local.o [LD] chan_local.o -> chan_local.so [LD] gentone.c -> gentone ./gentone busy 480 620 Wavelength 1 (in samples): 16.66667 Minimum samples (1): 50 (3.000000.3 wavelengths) Wavelength 1 (in samples): 12.90323 Minimum samples (1): 400 (31.000000.3 wavelengths) Need...

...? (assume I somehow know which calls should be paired up...) I could dump them both in a meetme - but that seems wasteful as i _know_ there will only ever be 2 parties. (And I need DTMF to flow through). I may want to record the bridged call, but that isn't vital. I'm thinking of dialing chan_local with a call-id but I'm sure I am missing something simpler. Tim.

...ice is appreciated. thanks Todd +++++++++++++++++++++++ /var/log/asterisk/full++++++++ Mar 15 16:12:37 VERBOSE[31964] logger.c: [chan_mgcp.so]Mar 15 16:12:37 VERBOSE[31964] logger.c: [chan_mgcp.so] => (Media Gateway Control Protocol (MGCP)) Mar 15 16:12:37 VERBOSE[31964] logger.c: [chan_local.so]Mar 15 16:12:37 VERBOSE[31964] logger.c: [chan_local.so] => (Local Proxy Channel) Mar 15 16:12:37 VERBOSE[31964] logger.c: [chan_phone.so]Mar 15 16:12:37 VERBOSE[31964] logger.c: [chan_phone.so] => (Linux Telephony API Support) Mar 15 16:12:37 VERBOSE[31964] logger.c: [chan_zap...

...ISend at default [Oct 30 23:44:45] DEBUG[16150] devicestate.c: No provider found, checking channel drivers for Local - MWISend at default [Oct 30 23:44:45] DEBUG[16246] devicestate.c: Notification of state change to be queued on device/channel Local/MWISend at default [Oct 30 23:44:45] DEBUG[16150] chan_local.c: Checking if extension MWISend at default exists (devicestate) [Oct 30 23:44:45] DEBUG[16150] devicestate.c: Changing state for Local/MWISend at default - state 2 (In use) [Oct 30 23:44:45] DEBUG[16176] app_queue.c: Device 'Local/MWISend at default' changed to state '2' (In use) b...

Hello asterisk users, Compiled asterisk-11.13.0 on openSUSE 13.1, however Channel driver chan_sip is XXX in menuselect --- it depends on: chan_local(M), res_crypto(M), res_http_websocket(M) chan_local is [*] chan_local in menuselect, res_crypto is in Resource Modules, Depends on: openssl(E) --- I don't know what (E) means ??? res_http_websocket is [*] res_http_websocket in menuselect. So this means that openssl(E) is holding ev...

...imit to the highest value. still has this problem. Is this the problem keeping asterisk in the foreground or this is a bug in SVN 1.2 16771? Apr 5 08:48:36 WARNING[14887]: channel.c:562 ast_channel_alloc: Channel allocation failed: Can't create alert pipe! Apr 5 00:48:36 WARNING[14887]: chan_local.c:523 local_new: Unable to allocate channel structure(s) Apr 5 08:48:36 NOTICE[14887]: app_dial.c:1042 dial_exec_full: Unable to create channel of type 'LOCAL' (cause 0 - Unknown) Apr 5 08:48:36 WARNING[14893]: res_agi.c:246 launch_script: unable to create fromast pipe: Too many open f...