similar to: Unabled to exit console

> -----Original Message----- > From: Ryan Parlee [mailto:listbox@jesca.com] > Sent: Tuesday, August 10, 2004 6:10 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] CVS version tags > > > > Can someone please tell me which version tags I should be > using when checking out with cvs. Right now I have two > directory trees: > > 1)

Can you post some of your sip configs and your extension configs. Thanks, -gcc -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Ryan Parlee Posted At: Sunday, April 04, 2004 12:10 AM Posted To: Asterisk User Group Conversation: [Asterisk-Users] Asterisk - Cisco 7960 - NAT Subject: [Asterisk-Users] Asterisk -

I have searched all over the web and have not really found anything related to this error.... The only thing I found is related to a system stops responding on DTMF, which does not happen here... THe following is the output from the CLI: *CLI> 2004-08-09 17:36:29 DEBUG[229390]: chan_sip.c:2332 sip_alloc: Allocating new SIP call for 640E2D47-E98B11D8-8FDBE54A-5FB5A0CB@65.67.76.30

Does anyone know the cause of these BAD BAD BAD messages? I think I lost all my calls when it happened too. We have nagios running against IAX and nagios reports that IAX is down. It would seem that the entire application locks up when this happens and calls are dropped. Connected to Asterisk 1.2.14 currently running on flexo (pid = 26846) Verbosity is at least 3 flexo*CLI> show channels

Does anyone know a place in Australia that sells a power supply suitable for the IAXy? I haven't had any luck tracking one down. -- jeremy bogan [ jeremy@segpub.com.au ] segment publishing - design.develop.host

HI all, Recently, I got the following message from CLI and finally the asterisk will hang. Anyone can tell me how to fix the problem or why it will happen. Thanks. Jun 17 14:18:02 DEBUG[24573] channel.c: Avoiding initial deadlock for 'SIP/1127-008d65f0' Jun 17 14:22:45 ERROR[24696]: chan_sip.c:11337 sipsock_read: We could NOT get the channel lock for SIP/1589-0087cdd0! Jun 17

Greetings I am receiving following error message. Any idea as to why? WARNING[147466]: File chan_sip.c, Line 4530 (sipsock_read): Recv error: Resource temporarily unavailable WARNING[147466]: File chan_sip.c, Line 4530 (sipsock_read): Recv error: Resource temporarily unavailable Frank...

i have created some ''public'' controllers like so: ./script/generate controller example and an ''admin'' controller like this: ./script/generate controller admin/example now i want to have caching in the ''public'' controllers so i add something like this: caches_action :show, :whatever but i want the ''admin'' controller to do

hello all, I am using the msql cdr module to store cdr in db, I realised that it does't capture the start and end time af a particular call record. Therefore I dive into the source code to add the start and end time into the query (add something like cdr->start, cdr->end), but end up getting segfault. the original version of cdr_mysql.so works fine but I need the start time and end

Hi, sometimes on by Asterisk 1.2.10 box I get these errors, there are about 50 active SIP channels so I dont know if calls are getting dropped or not. Should I be worried? 2006-10-18 09:33:59 WARNING[4375]: channel.c:787 channel_find_locked: Avoided deadlock for '0xb7341470', 10 retries! -- Executing GotoIf("SIP/sipCSC-b737f9e8", "0 ? 15") in new stack

I just got the following error messages displayed on my Asterisk console: ========================================== Oct 18 21:18:32 ERROR[22095]: chan_sip.c:11323 sipsock_read: We could NOT get the channel lock for SIP/5058977054-e577! Oct 18 21:18:32 ERROR[22095]: chan_sip.c:11324 sipsock_read: SIP MESSAGE JUST IGNORED: BYE Oct 18 21:18:32 ERROR[22095]: chan_sip.c:11325 sipsock_read: BAD!

Hi, sometimes on my Asterisk 1.2.10 box I get these errors, there are about 50 active SIP channels so I dont know if calls are getting dropped or not. Should I be worried? 2006-10-18 09:33:59 WARNING[4375]: channel.c:787 channel_find_locked: Avoided deadlock for '0xb7341470', 10 retries! -- Executing GotoIf("SIP/sipCSC-b737f9e8", "0 ? 15") in new stack

I too am having this problem reported by Frank Hoonhout. Asterisk runs fine for a few minutes and then stops accepting new calls. (I have a standalone server with SIP phones and I'm not doing any external registration). Asterisk CVS-04/07/03-09:28:50 0x420e0037 in poll () from /lib/i686/libc.so.6 (gdb) info threads 16 Thread 14351 (LWP 7258) 0x420e187e in select () from

Hi, I just got my TDM400P card (2 modules) and i installed it no probs. The card is detected fine, but for some reason when I add the card to zaptel.conf i get the following error: --snip-- ZT_CHANCONFIG failed on channel 2: Invalid argument (22) Did you forget that FXS interfaces are configured with FXO signalling and that FXO interfaces use FXS signalling? --snip-- My /etc/zaptel.conf looks

For some unknown reason, two of my IAX peers started registering on strange ports. Nothing has changed in the config, but they cannot make calls to me, however I can still make calls to them. In my IAX2 peers, the following is showing: user1/user1 203.XXX.XXX.XXX (D) 255.255.255.255 4585 OK (21 ms) user2/user2 203.XXX.XXX.XXX (D) 255.255.255.255 11280 OK (37 ms) I've

I have problem to change from handsfree mode to handset mode. When I switch from handset to handsfree while waiting for connection I press the green speakerphone button once. It is all well. Once it is connected I don't want to give the called party too much echo and I want to switch it back to handset. If I press the green button again I lose the call. Anyone knows whether it is possible

Dear All, I am using asterisk 1.4.21.2. I have used Originate manager application to to call the two persons. I have called AGI application to call another person. There I have used GET FULL VARIABLE AGI command to get the value. I am able to call another person form AGI. But when one end cut the call another one didn't disconnected. The following errors are displayed in Asterisk console,

Hi All, Something really weird is going on. I just stopped and started my asterisk server, and it just decided to stop starting. I only made one change to the config file for the musiconhold.conf, and now for some strange unknown reason, i'm getting the errors: [chan_modem_i4l.so]Apr 28 15:26:42 WARNING[16384]: loader.c:240 ast_load_resource:

I have finally crashed Asterisk for the first time and I'm wondering if anyone has seen this. This is a configuration with SIP endpoints and an IAX2 channel to another Asterisk PBX. The main PBX dropped a core file after a SEGV (signal 11 ) with the following trace: #0 0x42079133 in strchr () from /lib/tls/libc.so.6 #1 0x41bb0f9c in _fini () from /usr/lib/asterisk/modules/chan_sip.so #2

I frequently get this error message, it repeats itself hundred/thousands of times and never stops. chan_sip.c:7467 sipsock_read: Failed to grab lock, trying again... During this period, I can make no SIP calls what-so-ever. The only way I've been able to stop it is to killall -9 asterisk. Doing a restart now doesn't respond. Anyone know why? -- Daniel Jimenez