Displaying 12 results from an estimated 12 matches for "ast_control_progress".
2004 Jul 12
2
Indications missing on Cisco FXO -> ATA-186 (SIP)
...- Called 5503
-- SIP/5503-f6b5 is ringing
WARNING[-1323201616]: channel.c:1375 ast_indicate: 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.
2009 Dec 19
0
E1 ingress to SIP egress problem with 183 response
...res = process_sdp(p, req);
if (!ast_test_flag(req, SIP_PKT_IGNORE) && p->owner) {
/* Queue a progress frame only if we have SDP in 180 or 182 */
ast_queue_control(p->owner, AST_CONTROL_PROGRESS);
}
}
check_pendings(p);
break;
case 183: /* Session progress */
if (!ast_test_flag(req, SIP_PKT_IGNORE) && (p->invitestate != INV_CANCELLED) && sip_cancel_destroy(p))...
2013 Jul 15
2
ignore 183 session progress in parallel call scenarios
Hi,
I am using asterisk 1.8.22 and have a problem when calling in parallel
several SIP endpoints and I am not sure how to resolve it. In this case
Asterisk will not bridge any audio to the caller before the 200 OK. Which
means any progress announcements, including remotely generated ringback,
are not passed back to the caller.
This behavior is completely correct, because there is no way to know
2004 Nov 20
0
Can anyone shed some light on wht these calls were dropped?
...ov 20 16:52:05 DEBUG[-1109894224]: Made call 5 into trunk call 16384
Nov 20 16:52:05 DEBUG[-1109894224]: Created trunk peer for '111.222.333.444:4569'
Nov 20 16:52:05 DEBUG[-1109894224]: Expanded trunk '111.222.333.444:4569' to 6400 bytes
Nov 20 16:52:07 DEBUG[-1109894224]: Received AST_CONTROL_PROGRESS on Zap/2-1
Nov 20 16:52:07 DEBUG[-1095144528]: Ooh, voice format changed to 1024
Nov 20 16:52:15 DEBUG[-1109894224]: Took Zap/2-1 off hook
Nov 20 16:53:35 DEBUG[-1090942032]: Setting NAT on RTP to 0
Nov 20 16:53:36 DEBUG[-1090942032]: Stopping retransmission on '689f34b07622328c08f11b0a14c10311...
2003 Oct 06
1
chan_zap.c - echo cancelation getting in the way of dialing????
...: Found
Oct 6 10:59:57 DEBUG[5126]: File chan_sip.c, Line 568 (__sip_semi_ack): (Provisional) Stopping retransmission (but retaining packet) on
'5ecb985524e3ec232cf3e54d59674900@172.20.1.67' Request 102: Found
Oct 6 10:59:57 DEBUG[28688]: File chan_zap.c, Line 3612 (zt_indicate):
Received AST_CONTROL_PROGRESS on Zap/19-1
***********************************
A call that got all the digits but still fialed:
************************************
Oct 6 11:08:41 DEBUG[40976]: File chan_zap.c, Line 3428 (zt_read): DTMF digit: 6 on Zap/6-1
Oct 6 11:08:42 DEBUG[40976]: File chan_zap.c, Line 3428 (zt_read):...
2009 Feb 14
1
Progress() and Proceeding()
...ocol PBX so it's quite obvious that
what the applications really do largely depends upon the channel
driver.
But OTOH "Indicate progress" or "Indicate proceeding" doesn't
mean anything for the end user.
For SIP Progress() seems to send
"183 Session Progress" (AST_CONTROL_PROGRESS),
Proceeding() is "100 Trying" (AST_CONTROL_PROCEEDING).
183 starts early media, 100 does not.
Are there any situations where it makes sense to use either of
these applications from the dialplan?
Philipp Kempgen
--
AMOOCON 2009, May 4-5, Rostock / Germany -> http://www.amooc...
2007 Jul 17
1
Music on hold problem
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]
2006 May 26
0
SIP call problem
...et_read_format: Unable to find a path from g723
to ulaw
May 26 09:49:05 NOTICE[3227]: channel.c:1724
ast_set_write_format: Unable to find a path from ulaw
to g723
-- SIP/SIP_PROVIDER-77e1 is making progress
passing it to Zap/3-1
May 26 09:49:05 DEBUG[3242]: chan_zap.c:4479
zt_indicate: Received AST_CONTROL_PROGRESS on Zap/3-1
May 26 09:49:05 WARNING[3242]: chan_sip.c:1829
sip_write: Asked to transmit frame type 4, while
native formats is 1 (read/write = 4/4)
May 26 09:49:05 WARNING[3242]: chan_sip.c:1829
sip_write: Asked to transmit frame type 4, while
native formats is 1 (read/write = 4/4)
May 26 09:49:05 WA...
2005 Jan 27
2
Busy - problem with Asterisk spliced between Arcor E1-PRI and Ericsson BP250
...t;Zap/35-1", "134|1") in new stack
-- Goto (pri-external,134,1)
-- Executing Dial("Zap/35-1", "Zap/g3/134") in new stack
-- Called g3/134
-- Zap/38-1 is making progress passing it to Zap/35-1
Requested indication 14 on channel Zap/35-1
Received AST_CONTROL_PROGRESS on Zap/35-1
Dunno what to do with control type 15
-- Zap/38-1 is busy
Set option AUDIO MODE, value: ON(1) on Zap/38-1
Hangup: channel: 38 index = 0, normal = 63, callwait = -1, thirdcall = -1
Not yet hungup... Calling hangup once with icause, and clearing call
disabled echo cancellation on ch...
2005 Jun 22
3
Loosing hair on connecting Panasonic PBX- * - Euroisdn Italy
...uot;) in new stack
Jun 22 16:25:21 VERBOSE[5536]: -- Called g1/000038670613063
Jun 22 16:25:32 DEBUG[5536]: Queuing frame from PRI_EVENT_PROCEEDING on
channel 0/2 span 1
Jun 22 16:25:32 VERBOSE[5536]: -- Zap/2-1 is making progress passing it
to Zap/11-1
Jun 22 16:25:32 DEBUG[5536]: Received AST_CONTROL_PROGRESS on Zap/11-1
Jun 22 16:25:32 DEBUG[5536]: Dunno what to do with control type 15
Jun 22 16:25:34 VERBOSE[5536]: -- Channel 0/2, span 4 got hangup
Jun 22 16:25:34 DEBUG[5536]: Set option AUDIO MODE, value: ON(1) on Zap/2-1
Jun 22 16:25:34 DEBUG[5536]: Hangup: channel: 2 index = 0, normal = 33,
cal...
2004 Apr 07
1
Out of trunk data space on call number 16386, dropping
...51]: -- Call accepted by 65.113.15.19
(format ULAW)
Apr 7 09:42:23 VERBOSE[163851]: -- Format for call is ULAW
Apr 7 09:42:23 VERBOSE[770067]: -- IAX2[woodlane]/1 stopped sounds
Apr 7 09:42:23 VERBOSE[770067]: -- IAX2[woodlane]/1 is ringing
Apr 7 09:42:23 DEBUG[770067]: Received AST_CONTROL_PROGRESS on Zap/22-1
Apr 7 09:42:26 VERBOSE[770067]: -- IAX2[woodlane]/1 stopped sounds
Apr 7 09:42:26 VERBOSE[770067]: -- IAX2[woodlane]/1 answered Zap/22-1
Apr 7 09:42:27 DEBUG[163851]: Ooh, voice format changed to 4
2004 Jun 01
1
Zap and call pickup -- it don't work.
...located here:
http://lists.digium.com/pipermail/asterisk-users/2004-May/048527.html
(I've since changed channel 1-16's pickupgroup to just 3, it didn't help.)
What I see in *'s debug log:
Jun 1 15:21:13 DEBUG[213006]: SIMPLE DIAL (NO URL)
Jun 1 15:21:13 DEBUG[213006]: Received AST_CONTROL_PROGRESS on Zap/24-1
Jun 1 15:21:16 DEBUG[229391]: DTMF digit: * on Zap/2-1
Jun 1 15:21:16 DEBUG[229391]: DTMF digit: 8 on Zap/2-1
Jun 1 15:21:16 DEBUG[229391]: Enabled echo cancellation on channel 2
Jun 1 15:21:16 DEBUG[229391]: No call pickup possible...
Jun 1 15:21:16 DEBUG[229391]: No call pickup p...