similar to: Native bridging with Polycom 600

Hello, Talking with Soren Sratje about Caller ID in Brazil, we compare ours DTMF tones captured by ztmonitor. wcfxo correctly recognize the "DTMF CLIP" and asterisk shot the AST_STATE_PRERING correctly. But the DTMF tones are not reconized. In the chan_zap.c, the code: if (f->frametype == AST_FRAME_DTMF) { (...) Does not occurs because the frametype is always reconized as voice

Hello guys, I have searched high and low, but not found any information about rules of using DTMF in SIP INFO method. Cisco has described something with Signal=, but it look like this feature is dependent on implementors? The problem is chan_sip.c cannot correctly translate received DTMF digits, especially #,*. At least with my Antek EGW-804 gateway. Looking into chan_sip.c, I found this code:

Hi, Try as I might, I can't get hangups detected on a Zap channel with loop start lines. So, after someone leaves a voicemail and then hangs up, Asterisk doesn't know it, exits VoicemailMain2, and loops back to the corporate greeting, tying up the line even though the outside caller has hung up. Therefore, I've added the following hideous hack - er, code - to voicemail2.c. It

Hi all ( dev & user list ), I'm starting to implement the missing features in Meetme application : 's' -- send user to admin/user menu if '*' is received Line 438 -------- app_meetme.c ----------------------------------------------------------------------------- else if ((f->frametype == AST_FRAME_DTMF) && (f->subclass == '*') &&

I would like to be able to stop the chanspy application and go to the next step in the dialplan but I do not see a way to do that. I have looked at the code and I do not see a way to stop the chanspy application. Even if there are no channels that match the chanprefix pattern the chanspy application is not exited. Hitting the * key stops spying on a channel but then starts spying on the same

Hi list! I'm having some problems with chan_sccp and a Kirk IP600. Basically the handsets work (they emulate a Cisco 7940) but I have the following issues: 1. If a handset is in a conversation and there is a new incoming call, the incoming audio is muted (but the other party can still hear anything spoken on the handset). What is normal Asterisk behaviour, that a handset is left alone

hi, I installed icecast-2.2.0.tar.gz and ices-2.0.1.tar.gz on Fedora3 linux-2.6.12-1.1372_FC3). It works fine for playlist.ogg from the other CPU, such as 'xmms http://192.168.0.3:8000/listplay.ogg'. But when I use 'stdinpcm' like 'asterisk-ices.xml' which send voip's voice udp packets to 'asterisk-ices.xml' such as; .......(snip)...... <stream>

Hi, I try to set up attended transfer in my Asterisk Box . My features.conf look like this: [general] parkext => 100 parkpos => 1-5 context => parkedcalls parkingtime => 100 transferdigittimeout => 3l courtesytone = beep xfersound = beep xferfailsound = invalid featuredigittimeout = 500 ;adsipark = yes pickupexten = *8 [featuremap] atxfer => *2 blindxfer => # disconnect

Hello All I'm settup my asterisk as belows: sangoma card, connected with E1, CAS Signalling. I have two problem. 1. The asterisk don't received any DTMF when caller input to 2. when i dial to system, the caller hear bad sounds. monitor on console. asterisk show error. Jun 11 12:15:45 WARNING[1496]: channel.c:1447 ast_read: Exception flag set on 'UniCall/6-1', but no exception

Hello All i have big problem for unicall. my system work successful with sangoma card, E1 and CAS signalling (vietnam). when at the some time. i have trouble then my system is half (CPU instructions = 100) i tested for some case as belows: - When i dial, then my system became answer, the caller hangup. system error message show (loop without condition and half machine) Jun 11 12:15:45

Hi When I make a call from the outside to asterisk and the call is asnwered, all is OK, but when I make a call from the outside to asterisk and hangup before the call is answered, you got this WARNING in the console: Jul 6 19:33:08 WARNING[10037]: channel.c:1521 ast_read: Exception flag set on 'UniCall/2-1', but no exception handler Jul 6 19:33:08 WARNING[10037]: channel.c:1521

This is a serious problem! I have brought up this issue in four previous attempts to get some feedback. I find it hard to believe that no one else is having this same problem. Apr 11 13:27:36 NOTICE[4446]: channel.c:1906 ast_read: Dropping incompatible voice frame on Local/103@sip-00f3,2 of format alaw since our native format has changed to slin Apr 11 13:27:36 NOTICE[4446]: channel.c:1906

Several times a day I get this meesage scrolling on one of our asterisk boxes: Mar 13 14:43:47 NOTICE[11264]: channel.c:1911 ast_read: Dropping incompatible voice frame on Local/928@sanset-d88e,2 of format slin since our native format has changed to alaw Mar 13 14:43:47 NOTICE[11264]: channel.c:1911 ast_read: Dropping incompatible voice frame on Local/928@sanset-d88e,2 of format slin since our

Hi all. I have some problems to hear clearly music on hold, the sound interrupting. this some logs what i have in asterisk : rtp.c:298 process_rfc3389: RFC3389 support incomplete RFC3389: 1 bytes, level 8... RFC3389: 1 bytes, level 8... RFC3389: 1 bytes, level 8... RFC3389: 1 bytes, level 8... RFC3389: 1 bytes, level 8... RFC3389: 1 bytes, level 8... RFC3389: 1 bytes, level 8... RFC3389: 1

Hi everybody i have a problem trying to connect an incomming phone call from pstn to my (soft phone) iaxcomm, the phone rings but when i try to answer the call, asterisk sends a message like this. Jun 1 19:33:17 NOTICE[5013528]: channel.c:1223 ast_read: Dropping incompatible voice frame on IAX2[192.168.222.99:4569]/16 of format GSM since our native format has changed to ALAW i'm working

All, Im having a problem with ReceiveFax where its generating a ton of these messages the entire time the receivefax app is running receiving my fax. [Apr 7 22:16:06] ERROR[26918]: channel.c:2520 __ast_read: ast_read() called with no recorded file descriptor. Im running on Centos 5.2 with all patches. asterisk-1.6.0.9 asterisk-addons-1.6.0.1 dahdi-linux-2.1.0.4 dahdi-tools-2.1.0.2

Hi, When I call forward on PAP2, the incoming call will right the forwarded number. However, there is one-way voice problem. The caller can hear the destination(the forwarded number), but after the called party answers, the caller can't hear anything. Then the CLI> produce continuous errors as following: Oct 6 10:57:45 NOTICE[11026]: channel.c:1409 ast_read: Dropping incompatible vo ice

hi,all i have just set up asterisk 1.6.0.7 rc1 with spandsp 0.0.5 pre4 to ReceiveFAX, link to a E1 (DE410P) using dahdi this can receive the fax from E1 successfully, but i see many error message in the log like this: [Mar 16 09:24:38] ERROR[23540] channel.c: ast_read() called with no recorded file descriptor. when i receive a 5 pages fax, i will see this error message over 200 lines..... it

our * crashed twice in a month with segmentation fault & a core dump. here's the stack trace: #0 0xb7e11965 in mallopt () from /lib/tls/libc.so.6 #1 0xb7e10c43 in malloc () from /lib/tls/libc.so.6 #2 0xb7e17090 in strdup () from /lib/tls/libc.so.6 #3 0x08057ada in ast_verbose (fmt=0x0) at logger.c:879 #4 0xb7b29009 in moh_files_release (chan=0x9455ca0, data=0xb657be48) at

I'm getting a SIGSEGV at ast_slinear_saturated_multiply at the line: 351 res = (int) *input * *value; It's called from ast_frame_adjust_volume. The frame looks like: (gdb) print *f $6 = {frametype = AST_FRAME_VOICE, subclass = {integer = 100021, format = { id = AST_FORMAT_SLINEAR16, fattr = {format_attr = { 0 <repeats 64 times>}, rtp_marker_bit = 0