search for: ast_control_ringing

...ate: Unable to handle indication 3 for 'SIP/10.10.2.250-9903' -- SIP/5503-f6b5 answered SIP/10.10.2.250-9903 Looking at channel.c, I can see that this means that 'condition' is neither of 'AST_CONTROL_PROGRESS' or 'AST_CONTROL_PROCEEDING'. Presumably it's 'AST_CONTROL_RINGING', so why is this not handled? (NB Calls go through fine - all ulaw currently) Thanks a lot, Fran.

...otifying that line is out of service , temporary unavailable ., what to do to solve this problem In other words how to stop MOH since asterisk detect 183 and even if i can do that when the 183 comes from my soft switch which will allow user to hear the Ring Back Tone i found in the app_dial.c case AST_CONTROL_RINGING: Thanks in advance ********************************************* No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual...

Hi all, i have noticed that i can't hear a ring tone if i make a call from my TDM40B to a chan_modem_i4l endpoint. I had a look in the code from chan_modem_i4l and there is a function calling "i4l_handle_escape" that gives a AST_CONTROL_RINGING frame back. But this seems not work ...(or i4l is not signaling it ?) Til now i have used the Dail app like Dial, Zap/g1:XXXXXX|60|r so it is no wonder that i never noticed that the ring tone not working.... Have anybody an idea ? Thanks for help, Thomas. ***********************************...

....\n"); free(p->cidspill); p->cidspill = NULL; p->callwaitcas = 0; } p->subs[index].f.frametype = AST_FRAME_CONTROL; p->subs[index].f.subclass = AST_CONTROL_RINGING; break; *********************************************************************** I am seaching Why loop exits before reaching limit of 8867 or what makes zt_handle_event to control the flow. Please help me with any idea you have. Also tell if I am on wrong path for right problem PS: I have tried b...

i am creating call files, and catching successfully the ones that don't connect in a 'failed' extension. can anyone tell me how to find out the reason for the failure (ie busy, no answer). ${DIALSTATUS} doesn't appear to get set (presumably because Dial() isn't used) and channel_status doesn't seem to be any good. thanks in advance. -- - Rich Doughty

Hi, I am using asterisk 1.4. I have confgured the musiconhold.conf file. However, when i make a call and then hold the call it does nothing. in the CLI i do not see the starting/stopping musiconhold messages. i am making calls from sip to h323 using asterisk assip/h323 gateway (with gnugk and ooh323). i get the following messages when putting the call on hold: -- Executing [204 at default:1]

I am running asterisk 1.4.26.1 and using ALSA not oss dahdi 2.2.0 and libpri-1.4.10 I am calling into console/dsp I hear the audio just fine then after the hangup I hear ringing on the console/dsp. Why would that be? I found this bug for OSS https://issues.asterisk.org/view.php?id=13686 Does the same thing exist in ALSA??? some traces below Jerry == Parsing

...NV_CANCELLED) && sip_cancel_destroy(p)) ast_log(LOG_WARNING, "Unable to cancel SIP destruction. Expect bad things.\n"); if (!ast_test_flag(req, SIP_PKT_IGNORE) && p->owner) { ast_queue_control(p->owner, AST_CONTROL_RINGING); if (p->owner->_state != AST_STATE_UP) { ast_setstate(p->owner, AST_STATE_RINGING); } } if (find_sdp(req)) { if (p->invitestate != INV_CANCELLED)...

Hi there, Besides the problem of Voicetronix dialing too early before the carrier gives a dial tone, there also appears to be issues with the audio quality when more than 1 channel is utilized.