similar to: Could not stop autoservice on calling channel

Hello, I am currently load testing some new hardware and have been receiving the following warning. Does anyone happen to know if there are any risks or performance implications for increasing the MAX_AUTOMONS value? The current value is 1500. asterisk[30322]: WARNING[30423]: autoservice.c:110 in autoservice_run: Exceeded maximum number of automatic monitoring events. Fix autoservice.c Steven

After updating to 1.4.15, I have the following issue: When I try to use the "M" macro option in the Dial() option, I get the following in the console: -- Executing Dial("Zap/1-1", "Zap/g2/w5051234|60|M(set-userfield^local)KT") -- Called g2/w5051234 -- Zap/3-1 answered Zap/1-1 [Dec 6 12:10:58] ERROR[19496]: app_dial.c:1541 dial_exec_full: Unable to start

hi, all actually i can't understand what is the use of autoservice.c file. can anybody help me. thnks in advance. Bhrugu mehta

Hi All, I am using an Asterisk 1.6.2.6, and when I use this part of the dialplan: exten => 8355,1,Dial(SIP/${EXTEN}&IAX2/${EXTEN},18,tTWwr) exten => 8355,n,Dial(IAX2/8366,48,tTWwr) (i made that simple to exhibit issue) I got just 1 ring in 8366 extension before it hangup, what i noticed is the total time spent on ringing is 30s that means if i use 12s in the first dial i get 18s left

hi, i am getting these errors while installing asterisk. i reconfigured kernel and i have all the modules installed. kernel-source readline readline-devel openssl openssl-devel this is the error: (at the last part of the installation) gcc -g -o asterisk -Wl,-E io.o sched.o logger.o frame.o loader.o config.o channel.o translate.o file.o say.o pbx.o cli.o md5.o term.o ulaw.o alaw.o callerid.o

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

[Apr 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 -

Has anyone else got 1.2 compiled from cvs ? I've posted the question below to the -dev list but got no answers: 1) No-one else is trying beta 1 2) No-one else is having any issues (I must be the idiot) 3) No-one else saw my message :) I have been trying to compile 1.2 beta 1 on a centos 4 box, to no avail. The "make" command seems to compile ok, but "make install"

Hi, I'm having a problem with the queue behaviour in my place: I have two ISDN channels to the outside (Zap/1) and two channels two a Siemens Gigaset (Zap/4). I also use a SIP gateway to call outside and have a couple of IP phones around as well (SIP). The Gigaset has about 5 phones connected to it (+base station). Whenever two people are using those, I always am blocking two internal

hi!, Asterisk Version:1.6.1.20 OS: CentOS release 5.3 (Final) uname: 2.6.18-128.el5PAE #1 SMP Wed Jan 21 11:19:46 EST 2009 i686 i686 i386 GNU/Linux Application: Queue Specific Details: Obtain Acknowledgement from queue member before bridging the caller. Language: AEL Similar Example:http://www.voip-info.org/wiki/view/Asterisk+tips+Queue+Member+ackcall Scenario: 1. User calls in a General Number

Hi, I am trying to use asterisk open source version(asterisk-1.6.0.5) with MySQL (using res_odbc)support for extensions and users list. The call rate is 7 calls / second and each call stays for 120 seconds. after making 25000 successful calls , calls started failing with following message on CLI. [Nov 11 08:50:04] WARNING[2258]: app_dial.c:1502 dial_exec_full: Unable to create channel of

the gdb of the core taken from the asterisk as the time of crash is as below I run asterisk-1.2.5 on fedora core 3 with chan_ss7 can someone help out? #0 ast_var_name (var=0x1) at chanvars.c:71 71 if (var->name[0] == '_') { (gdb) bt #0 ast_var_name (var=0x1) at chanvars.c:71 #1 0x0808934e in pbx_builtin_getvar_helper (chan=0x0, name=0xf5bc2d46

Hello Just installed fresh Debian testing box, checked out Asterisk and others from CVS stable (-r 1.0), and now trying to 'make install' in Asterisk. I get this error: if [ -d CVS ] && ! [ -f .version ]; then echo CVS-v1-0-03/29/05-15:19:53 > .version; fi gcc -g -o asterisk -Wl,-E io.o sched.o logger.o frame.o loader.o config.o channel.o translate.o file.o say.o pbx.o

hello, i try to compile * in uclibc enviroment (uclibc 0.9.19 ), but still getting following error does anyone know how to solve it ? regards Marian --------- gcc -g -o asterisk -rdynamic io.o sched.o logger.o frame.o loader.o config.o channel.o translate.o file.o say.o pbx.o cli.o md5.o term.o ulaw.o alaw.o callerid.o fskmodem.o image.o app.o cdr.o tdd.o acl.o rtp.o manager.o asterisk.o

I am still getting an error compiling the 1.2-Beta version. The tarball works fine, but I have never been able to compile the 1.2beta from CVS. I have been compiling CVS-HEAD on the machine for quite some time. It goes into this loop: if cmp -s include/asterisk/version.h.tmp include/asterisk/version.h ; then echo; else \ mv include/asterisk/version.h.tmp include/asterisk/version.h ; \

Sorry for posting this again, but it seems to have become attached to another thread. Guess I replied to another message instead of starting a new one... Hi, I'm trying to setup a call forwarding rule so that when an extention doesn't answer the call is forwarded to my mobile. I'm using voiptalk.org for incoming and outgoing calls and SIP phones for extentions (so all IP based -

Hello, About the INBOX location when using maildir, in the wiki, http://wiki.dovecot.org/MailboxFormat/Maildir, I can read: " Directory Structure ~/Maildir/new, ~/Maildir/cur and ~/Maildir/tmp directories contain the messages for INBOX. The tmp directory is used during delivery, new messages arrive in new and read shall be moved to cur by the clients. " But in my Maildir, I see

Hi, I was running asterisk 1.4, but recently upgraded to 1.6 (for fax support) at the same time as moving from Ubuntu hardy to I have a single TDM400P rev I with two fxo and two fxs modules, these were working perfectly for years on Asterisk 1.4 using Zaptel drivers with Oslec. Now I've moved to 1.6 so I am using Dahdi. Distribution is stock ubuntu package. After several hours (perhaps 24

Dear all, we have encounter problem when starting asterisk in the foreground, "asterisk -vvvvgc" with more 100 SIP calls concurrently. we have set ulimit to the highest value. still has this problem. Is this the problem keeping asterisk in the foreground or this is a bug in SVN 1.2 16771? Apr 5 08:48:36 WARNING[14887]: channel.c:562 ast_channel_alloc: Channel allocation

Hi, I am using Asterisk 1.2.9.1, with chan_h323. The problem I am coming across is when trying to bridge an incoming H323 call with another H323 call: phone1 dials into asterisk with H323, for extension 111 in asterisk: exten => 111, 1, Dial(chan_h323, H323/111@phone2) (in my extensions.conf the syntax is good ... this is no). I can see how the first call is partially processed, then the