search for: app_dial

Hello, any idea why the Asterisk CLI gets flooded by these messages ? How can the SIP peer /vita3/ cause this flood ? [Apr 2 11:45:48] VERBOSE[17029] app_dial.c: [Apr 2 11:45:48] -- SIP/vita3-000010af requested media update control 26, passing it to SIP/708708-000010b3 [Apr 2 11:45:48] VERBOSE[17029] app_dial.c: [Apr 2 11:45:48] -- SIP/vita3-000010af requested media update control 26, passing it to SIP/708708-000010b3 [Apr 2 11:45:48] VERBOSE[170...

hi all i am using CVS with Realtime mysql on backend. Dial application is invoked again and again what is the reason. I have tested it by printing some message to debug. this application is invoked again and again here is debug you can see lot of messages from app_dial.c at the end. Any one tell me what is the reason. Is this a bug or what Kamran Ahmad ------------------------------------------------------ *CLI> sip debug SIP Debugging Enabled *CLI> Sip read:...

...r 26 10:47:01] NOTICE[32151]: chan_sip.c:16223 sip_poke_noanswer: Peer '3516533812' is now UNREACHABLE! Last qualify: 86 [Apr 26 10:47:11] NOTICE[32151]: chan_sip.c:12723 handle_response_peerpoke: Peer '3516533812' is now Reachable. (98ms / 2000ms) [Apr 26 12:08:49] WARNING[32273]: app_dial.c:1242 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) [Apr 26 12:51:20] WARNING[32281]: app_dial.c:1242 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) [Apr 26 12:52:56] WARNING[32284]: app_dial.c:1242 dial_exec_full: Unable t...

...le switch on 'Zap/1-1' Feb 3 12:11:17 NOTICE[6424]: chan_zap.c:5363 ss_thread: Got event 2 (Ring/Answered)... -- Executing Dial("Zap/1-1", "SIP/2001&SIP/2002&IAX2/2003&IAX2/2101&SIP/2102&SIP/2103|10") in new stack Feb 3 12:11:17 NOTICE[6424]: app_dial.c:746 dial_exec: Unable to create channel of type 'SIP' Feb 3 12:11:17 NOTICE[6424]: app_dial.c:746 dial_exec: Unable to create channel of type 'SIP' Feb 3 12:11:17 NOTICE[6424]: app_dial.c:746 dial_exec: Unable to create channel of type 'IAX2' -- Called 2101 Feb...

I made a patch to app_dial.c to make Dial(ext1&ext2&ext3,tumeout,B) return busy when just one extension is busy. http://www.neland.dk/app_dial.c.diff It works, but... I can't figure out setting/reading an option. It looks fairly easy, but the flag is always set. *** app_dial.c.org 2009-11-04 22:15:50.0000...

...1226204480]: channel.c:1858 ast_channel_make_compatible: No path to translate from SIP/405-db6d(256) to Zap/1-1(72) Jun 9 06:30:42 WARNING[1226204480]: chan_sip.c:1322 sip_write: Asked to transmit frame type 8, while native formats is 256 (read/write = 256/256) Jun 9 06:30:42 WARNING[1226204480]: app_dial.c:331 wait_for_answer: Unable to forward voice Jun 9 06:30:42 WARNING[1226204480]: app_dial.c:331 wait_for_answer: Unable to forward voice Jun 9 06:30:42 WARNING[1226204480]: app_dial.c:331 wait_for_answer: Unable to forward voice Jun 9 06:30:42 WARNING[1226204480]: app_dial.c:331 wait_for_answe...

Hi. Since a customer requested us that feature, I wrote this little patch for app_dial to allow to play an announcement to the called party, as soon he answers. you can define the file to play in the dial() option, using A(filename). for example: exten => blah,1,Dial(Zap/blah,30,rA(/my/own/announce)Tt) that doesn't break anything ... feel free to blame me for anything bad th...

Looks like half a patch has been applied to app_dial in cvs head could someone with commit rights fix it. Thanks Chris

...starwars*CLI> sip show registry Host Username Refresh State 192.168.7.16:5060 7062210 105 Registered -- Executing Dial("SIP/192.168.7.14-085a4790", "Zap/1/67742142") in new stack Nov 30 01:41:52 NOTICE[507921]: app_dial.c:742 dial_exec: Unable to create channel of type 'Zap' == Everyone is busy/congested at this time -- Executing Dial("SIP/192.168.7.14-085a4790", "Zap/1/61002663") in new stack Nov 30 01:41:52 NOTICE[507921]: app_dial.c:742 dial_exec: Unable to create channel of...

...run\" -DASTSPO OLDIR=\"/var/spool/asterisk\" -DASTLOGDIR=\"/var/log/asterisk\" -DASTCONFPAT H=\"/etc/asterisk/asterisk.conf\" -DASTMODDIR=\"/usr/lib/asterisk/modules\" -DASTAGIDIR=\"/var/lib/asterisk/agi-bin\" -DBUSYDETECT -fPIC -c -o app_dial.o app_dial.c app_dial.c: In function `wait_for_answer': app_dial.c:232: parse error before `o' app_dial.c:242: parse error before `o' app_dial.c:285: parse error before `o' make[1]: *** [app_dial.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk/apps' make: *** [subdirs]...

...it to 30. Now I get two rings. Other extensions ring once or twice. After some time has gone by since this was first reported, all phones in my random sample ring only twice. As I trace a call to that extension through the log, I notice it setting the ring timer properly (I think) and then I see app_dial.c - SIP/1234-00001111 is ringing Then eventually app_dial.c: -- Nobody picked up in 30000 ms But according to the timestamps, the time from the one event to the other is ten seconds! Here's another example- call starts: [2019-01-14 08:17:33] VERBOSE[13311] pbx.c: -- Ex...

...h shut down the Asterisk box. When the power went up, Asterisk came back up okay but the ports on the card were all red. Zttool show red alarm and cat /proc/zaptel/1 show red alarm today. Both incoming and outgoing cannot be made. When a outgoing call was made, we got the following error message: app_dial.c: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) We suspect it was the ISDN line problem and so we waited a whole day for the engineer to arrive. He plugged an ISDN phone into the line and found it was working because he could call out. We are perplexed a...

Just a quick question ---cut--- [Sep 17 15:52:14] WARNING[8232] app_dial.c: Unable to create channel of type 'IAX2' (cause 34 - Circuit/channel congestion) [Sep 17 15:52:14] WARNING[8232] chan_iax2.c: No more space [Sep 17 15:52:14] WARNING[8232] chan_iax2.c: Unable to create call [Sep 17 15:52:14] WARNING[8232] app_dial.c: Unable to create channel of type '...

Hi Everyone, I am currently running Trixbox 2.6 and I have a problem with Asterisk. /etc/init.d/asterisk status Asterisk dead but subsys locked I deleted all files in /var/run/asterisk folder and asterisk restart... It's ok for a while. But some days after Asterisk again is dead. Can anybody help me? Rgs / budacsik

Wondering if anybody else is experiencing this: Using June 21st CVS Call made internally from one Polycom IP600 to another. Core dump with the last message in log as: NOTICE[17426]: app_dial.c:681 dial_exec: Unable to create channel of type 'SIP' Happens a couple of times a day. No, I haven't done any backtracing, verbose logging, etc., (first thing in the morning, I promise) I just wanted to see if anybody might have a quick fix. Thanks John Baker

...'show warranty' for details. ================================================ The problem lies when I try to compile rxfax and txfax. The compiler jumps out of the /usr/src/asterisk/asterisk/asterisk-1.2.30.2/apps/ directory: /bin/sh: curl-config: command not found cc -fPIC -c -o app_dial.o app_dial.c app_dial.c:37:22: error: asterisk.h: No such file or directory app_dial.c:39: error: expected declaration specifiers or ?...? before string constant "asterisk.h" is located: # find / -name "asterisk.h" /usr/src/asterisk/asterisk/asterisk-1.2.30.2/include/asterisk...

...0xb659fcd0, c1=0x9455ca0, config=0xb6c4feb0, fo=0xb6c4f988, rc=0xb6c4f98c) at channel.c:3524 #10 0xb78fad29 in ast_bridge_call (chan=0xb659fcd0, peer=0x9455ca0, config=0xb6c4feb0) at res_features.c:1319 #11 0xb7099301 in dial_exec_full (chan=0xb659fcd0, data=0xb6c4feb0, peerflags=0xb6c50568) at app_dial.c:1577 #12 0xb7097dc5 in dial_exec (chan=0xb7ed1900, data=0xb7ed1900) at app_dial.c:1619 #13 0x0808e445 in pbx_extension_helper (c=0xb659fcd0, con=0xb7ed1900, context=0xb659fe20 "op05_x", exten=0xb659ff14 "00116", priority=1, label=0x0, callerid=0x0, action=0) at pbx.c:553...

...bout 20 seconds, then the message "all circuits are busy now. please try your call again latter" followed by the congestion tone. Instead, I want this to busy ring and then hang up without any message. Here is a snippet from site A: ... [2014-07-09 09:56:16] VERBOSE[21606][C-0000dab7] app_dial.c: -- Called DAHDI/g5/5551212 [2014-07-09 09:56:17] VERBOSE[21606][C-0000dab7] app_dial.c: -- DAHDI/i7/5551212-411b is proceeding passing it to SIP/260-0000a2f1 [2014-07-09 09:56:17] VERBOSE[21606][C-0000dab7] app_dial.c: -- DAHDI/i7/5551212-411b is ringing [2014-07-09 09:56:17] VERBOSE...

A nice flag in app_dial ? would be f and F to indicate weather to send flash to the fxo or fxs device in a bridged call. so if you have a pots line on an x100p bridged to a tdm400p and the pots line has call waiting you hear the call waiting signal 'f' would behave as normal and flash on the tdm400p and &...