similar to: chan_skinny issue

Hi, After reading this valuable forum and the voip-info wiki and follow all the steps , but my Cisco 12SP+ remains unregistered. These are my config files: skinny.conf [general] port = 2000 ; Port to bind to, default tcp/2000 bindaddr = 172.20.1.1 ; Address to bind to dateFormat = D-M-Y ; M,D,Y in any order (5 chars max) keepAlive = 120 languaje=es allow = all ; disallow

I have a bunch of 7910's that I managed to get registered with Asterisk 1.2.14: managed5*CLI> skinny show devices Name DeviceId IP TypeId R Model NL -------------------- ---------------- --------------- ------ - ------ -- test7 SEP0004C1878F8E 192.168.0.226 6 Y 7910 1 The problem is that the phone resets when I attempt to make a call from it or place a call to it. If I pick up I have

Hi all: I've got some core dumps with chan_skinny. The client is ATA186 with v2.16.1.ms ata18x (Build 030814a). The * version is CVS-10/05/03-16:03:26. When I make a call, the phone connected with ATA rings only 1 time and * dies. Maybe I have some errores in ATA config. If someone has proven configs for ATA, please send me the details. Thanks in advance, Gus The logs: *CLI> Version

Went from 1.6.0.6 to 1.6.0.8 and resulted in segmentation fault. Reverted to 1.6.0.6 and back to normal. ------------------ Linux asterisk.hulber.com 2.6.18-128.1.1.el5 #1 SMP Mon Jan 26 13:58:24 EST 2009 x86_64 x86_64 x86_64 GNU/Linux Apr 3 11:49:56 asterisk kernel: asterisk[3780]: segfault at 00002ce1ac0537a8 rip 0000003e980715a8 rsp 00007fff5bf00c30 error 4 Apr 3 11:50:00 asterisk

Can anyone recommend a good voip provider? Thanks

Hi iam not able to start asterisk give me following error any help STARTING ASTERISK /usr/sbin/safe_asterisk: line 42: 4633 Illegal instruction (core dumped) ${ASTSBINDIR}/asterisk ${CLIARGS} ${ASTARGS} >&/dev/${TTY} </dev/${TTY} Asterisk ended with exit status 132 Asterisk exited on signal 4. Automatically restarting Asterisk. /usr/sbin/safe_asterisk: line 42: 4637 Illegal

i m using asterisk 1.8.3 on a centos 5.5 computer when i try to change my name on the voicemail asterisk dump core here what i got on the console -- User ended message by pressing # -- <SIP/6672-00000002> Playing 'auth-thankyou.alaw' (language 'fr') -- <SIP/6672-00000002> Playing 'vm-review.alaw' (language 'fr') -- Saving message as

Hi I was trying to have realtime voicemail working with ODBC Driver. Everything works fine with MySQL Realtime access, BUT as I want to implement ODBC Storage as well, it seems that everything should go through ODBC ( what I read on voip-info wiki page ) But I do not manage to make it work with ODBC. Outside Asterisk, ODBC works fine, I can access my databases & tables ! Asterisk fails to

Hi, My asterisk is frequently dead by itself. It leaves messages: /usr/sbin/safe_asterisk: line 40: 24890 Segmentation fault (core dumped) asterisk ${CLIARGS} ${ASTARGS} >&/dev/${TTY} </dev/${TTY} Asterisk ended with exit status 139 Asterisk exited on signal 11. Automatically restarting Asterisk. Anyone has any idea of the cause? Thanks.. Best Regards, Stevanus

Hi all, I use asterisk-1.6.2.7 and asterisk addon version 1.6.2.1. It started ok with out cdr_addon_mysql.o. But when I put cdr_addon_mysql.o in to modules folder, it fail at start and the following out has been thrown: ---------- [root at localhost modules]# /usr/sbin/safe_asterisk: line 145: 13270 Segmentation fault (core dumped) nice -n $PRIORITY ${ASTBINDIR}/asterisk -f ${CLIARGS}

I'm seeing what looks to me like odd behaviour when I use na.omit on a simple "length" function, as follows. > sno [1] 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 > a [1] 0 1 0 0 1 1 0 0 0 0 0 0 0 0 1 1 0 1 1 0 0 0 0 0 0 0 0 0 0 0 0 1 0 0 > b [1] 1 1 0 1 1 1 0 0 NA 0 0 0 NA 0 1 NA 0 1 0 0

Hi all, I've just built a new installation of CentOS release 5.3 (Final) and have installed both <http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-1.6.1.1.tar.gz>Asterisk 1.6.1.1 and subsequently Asterisk 1.6.0.10 (thinking that I was maybe trying to be too cutting edge) on a Dell PowerEdge sc440 server (nothing complex - Pentium Dual core 2ghz - 1gb ram - 70gb

I upgraded * via svn and it did not work !!! 1. asterisk-addon did not compile! pbx:/usr/local/src/svn-versions/asterisk-addons # make ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql `ls *.c` make -C format_mp3 all make[1]: Entering directory `/usr/local/src/svn-versions/asterisk-addons/format_mp3' gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes

I've just installed asterisk on a Debian Linux (apt-get it) And i have placed two sip phones in sip.conf and i'm testing parking with them I have phone1-SIP/1000 and phone2-SIP/1007 The following happens if i park from calling party and everything is OK 1. Pickup Phone2 and call to Phone1 2. Talk 3. Phone2 dials #700 and parks the call (it is placed in 701) 4. Phone2 is hangup 5. Pickup

Hi, I want to extract information from a number of text files in a folder. The files are named as : 82534.txt, 82555.txt, 8282787.txt etc. I give below a sample of the kind of the information in the text file : ######## #(a lot of preceding text) 2008-10-01????? 06:30:12??? ??? ??? ??? 2 of 3 page #(some lines of text - varies from file to file) sekvens??? 890 # lines of text sNo???? start???

I was trying to enable CDR in a mysql-database when the following occured : asterisk*CLI> cdr status CDR logging: enabled CDR mode: simple CDR output unanswered calls: yes CDR registered backend: cdr_manager CDR registered backend: cdr-custom asterisk*CLI> exit [Jun 29 21:56:52] Executing last minute cleanups [root at asterisk asterisk]# vi modules.conf [root at asterisk asterisk]# service

Hi, in following of a recent discussion I got to work on MGCP with the Cisco ATA186 again, and got it to work very nicely. However, there is a little thing with transfers I would like to get comments on: Call comes in from PSTN and goes to an ATA186 (MGCP) Call is answered and then, using flash, transferred to another extension If the extension is available, there can be an announcement and

Hi, I've spotted weird crash of Asterisk cvs Stable. I have defined queue in queues.conf : [prodaja] music = default announce = queue-markq strategy = ringall context = from-pstn timeout = 15 retry = 5 maxlen = 0 announce-holdtime = no announce-frequency = 30 announce-holdtime = yes monitor-format = gsm|wav|wav49 monitor-join = yes eventwhencalled = yes member => Agent/1000

Hey guys, can you shead some light on this? I will copy my mgcp.conf and post below, but here is the problem. I can't get call waiting to work with my MGCP device. I already have one call going, and I can hear the second call come in, I flash over to it, but all I get is a dial tone, * puts the 1st call on mute/hold, but I never get the second, and it terminates. I flash back over and pick

Hi, Anybody seen this before? (using a pre-compiled asterisk from the OBS on a sles11sp1) (I mean, i did the same with a 1.6 without any problem, but i need 1.8) after starting: kc3004:~ # /usr/sbin/safe_asterisk: line 145: 16133 Segmentation fault (core dumped) nice -n $PRIORITY ${ASTSBINDIR}/asterisk -f ${CLIARGS} ${ASTARGS} > /dev/${TTY} 2>&1 < /dev/${TTY} Asterisk ended with