similar to: Use of the "hint" modifiers - examples, anyone?

Hi Is there any work being done on implementing IM/SIMPLE support for SIP on Asterisk? Like a presence server? rdgs, /Staffan Kerker

Hi My hardware PBX run asterisk on vxworks,Because the vxworks not support perl. Now I want to add a callback function to my pbx. now it can store Caller and Called party numbers in queue when Called party is busy Then I malloc a new ast_channel to call.It is should use ast_get_channel_by_exten_locked() or ast_channel_alloc() , my program as follow,But it isn't work, anyone know how to

I'm tying to use park and announce for call park on Asterisk 1.4.2 but I'm having trouble getting it to work properly. This use to work with an older version of Asterisk. A telephone on the PSTN calls an IP phone. The IP phone is assigned extension 3-8396. 3-8396 answers the call and attempts to perform a blind transfer to x700, the parking lot number. The transfer gets to Asterisk,

a) jerjers been doing a lot commendable work for * b) support is not mandatory, and i agree with royk it should not be withheld based on political viewpoints, that's pointlessly draconian c) choice is always good, so people should have the option of oh323 or h323, let them decide, and not limit them, unless astmaster chooses to limit them, and that too based on valid points d) jerjer gave a

I have the programable button led's working properly on my snom 200 except they don't flash during a ring event. I found a post by Andre Bierwirth saying he had a patch that he submitted but didn't make it into CVS. I would like to get a copy of that as a starting point to implement button flashing on ring. I have read through all the code and it looks like it should be pretty

Hi, This patch fixes a couple of segfaults in music-on-hold, frame smoother routines and channel allocation in Asterisk. Mark, feel free to apply it in CVS (if approved). Regards, Michael. -------------- next part -------------- Index: channel.c =================================================================== RCS file: /usr/cvsroot/asterisk/channel.c,v retrieving revision 1.25 diff -u

Hello folks, I'm trying to get an 8x8 DTA-310 running mgcp to work. I get no dialtone & can't get it to ring. My mgcp.conf says: ; ; MGCP Configuration for Asterisk ; [general] port = 2427 bindaddr = 0.0.0.0 [172.16.2.25] host = 172.16.2.25 context = default line => aaln/1 And here's the interesting bits of extensions.conf: [globals] ... TRUNK=H323/BYEXTENSION@pstn_gw ...

Hi I am new here and am wondering if I have the correct list to subscibe to. I am looking for a user forum; technical mutual help/tutorial type list; would this be that type of thing? So far the messages I am seeing are mainly intercommunications between what appear to be developers working on assigned sub-projects of various flavors of samba. I don't want to spam a list with

Does anyone object to me removing the andersaa pass from mainline for 2.7? It is buggy and unmaintained, and people keep filing bugs about it because it is tantalizing. If someone wants to complete the work in the future, they can always resurrect the code from SVN. -Chris

Hi All I just completed a custom application for Asterisk (i m not a C guru so i just copy codes from other application and alter according to my needs) attached files is the source file this application is working fine but still i need you people to give suggestion to improve it Primary task of this application is to get a parameter from extensions.conf, query sql server and play a files

Hello, I managed to terminate calls from cisco: as5300 and 7206 to asterisk over h323. I tested both oh323 from inaccessnetwork and JerJers chan_h323. I used 1.12.2 version of oh323 and 1.5.2 version of pwlib. After latest changes from JerJer chan_h323.c works ok when receiving traffic from ciscos. I havnt found any audio problems although I didnt send much traffic. Latest oh323 has some

Hi, Since Carl has kindly provided us with fax support for CAPI based cards, we have been using it with much success. Today I have modified app_capiFax so that it now supports a dynamic CSID. The following example uses the DNID created by chan_capi on an AVM Fritz! card. * Receive a fax with CAPI API. * Usage : capiAnswerFax2(path_output_file.SFF|stationID) * * This function can be

static void senddialevent(struct ast_channel *src, struct ast_channel *dst) { manager_event(EVENT_FLAG_CALL, "Dial", "Source: %s\r\n" "Destination: %s\r\n" "CallerID: %s\r\n" "CallerIDName: %s\r\n" "SrcUniqueID: %s\r\n" "DestUniqueID: %s\r\n" "CDRUserfield: %s\r\n", src->name,

Address 0xfffffffe out of bounds why and how to solve.MyConfbridgeCount(conferencenumber,variablename )return total number of user in conference given by conferencenumber otherwise zero.At runtime using MyConfbridgeCount(4000,count ).now app2: MyConfbridgeCount will call function count_exec(struct ast_channel *chan, const char *data).But at compile time char * data cause core dumped.

I'm still getting duplicate process but the results of gdb are different. Can anyone shed any light onto what is causing this? (gdb) info threads 1 Thread 1091845040 (LWP 31287) 0xffffe410 in __kernel_vsyscall () (gdb) thread apply all bt Thread 1 (Thread 1091845040 (LWP 31287)): #0 0xffffe410 in __kernel_vsyscall () #1 0x4004f13e in __lll_mutex_lock_wait () from

Hi, I'm writing asterisk application in C language. I need to know what is state of my asterisk user, so I have found command: ast_device_state(data); . So if my IP phone is reachable I get status 1 (AST_DEVICE_NOT_INUSE<http://www.asteriskpbx.com/doxygen/1.2/devicestate_8h.html#42ea804da1426b4117686332400b27c2>). But when I have unplugged my phone's cable , and sip show peer

Hi there, We're trying to get IMAP voicemail storage working on an MS Exchange server - I would be grateful if anyone who has successfully done this could post the magic soup here, as extensive Google searching has yielded nothing other than tantalizing references to it being done without any specifics. -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web:

In conjunction with Mark Spencer's visit to our Paris office, we'll be kicking off Asterisk TV (http://asterisktv.com) live during the weekls Asterisk Users Conference. I believe someone from Lumenvox will be back with us on the conference, now that I've had a chance to play with their speech recognition product. I think we're starting to get some great info from the user

I've been searching around on the internet trying to find out about conduits and options for syncing programs with my Palm Pilot device. Google searches turn up a lot of discussion on the matter over a two or three year period, everything from people talking about doing it or how it should be done or how much they want it done. But I can't actually determine what, if any, tangible

Hello. Once again, I came up with a problem for which I can't seem to find a solution. I'm not able to make BLF work with Thomson ST2030 phones and Asterisk (1.2.13). I've set up hints in dialplan, as well as Subscibe keys on the phone. The LED status gets updated according to the associated line status. However, when a phone is ringing, If I try to pickup the call by pressing the