similar to: Avoiding deadlock... Problem

I have a very serious problem with calls between PAP2-NA and a TDM2400 (8 FXO). Almost every call dropped after between 20 and 30 seconds with conversation. I disable the sound card, serial and other things on my server, but the problem still continues. I've changed the RPT Packet Size to .20 on PAP2-NA, but nothing. Here a piece of my log: [Jan 31 07:10:43] DEBUG[3131] channel.c: Hanging up

Hi What causes deadlock? Apr 5 14:02:43 WARNING[2413] channel.c: Avoided initial deadlock for '0x82acb10', 10 retries! Apr 5 14:02:43 WARNING[2413] channel.c: Avoided initial deadlock for '0x8298160', 10 retries! Here is the portion of the log: Apr 5 14:02:42 NOTICE[23363] chan_zap.c: Got event 18 (Ring Begin)... Apr 5 14:02:42 VERBOSE[23363] logger.c: -- Executing

Hi everybody, Well, I've finally got asterisk to to talk nicely with my Intertel pbx. Currently there is a outside T1 line (e&m wink start, esf, b8zs) connected to asterisk, and then asterisk connected similarly to my Intertel pbx. For right now all asterisk is doing is passing calls between the two. When I call out from the pbx, I can connect perfectly to the outside world. When I

Greetings, I've been having a large number of deadlock issues lately on chan_sip occurring only on VIA EPIA ML6000 boards. I'm curious if anyone else is having similar issues. My Config (have multiple systems all running the same hardware with the same problem) VIA EPIA ML6000 1GB RAM 80GB HDD Various Digium Cards (T1 and TDM cards) Trixbox 1.2.2 (though running stock asterisk code)

On 1.2.x I have a random problem when a Zap/x channel flashes to transfer or make a three way call. The Zap/x-2 channel is created and the transfer or three way proceeds, but on hangup the Zap/x-1 channel fails to destroy the old bridge and asterisk goes crazy logging the problem. Here is an example debug log. This happens only once a day or so, with 100 or so users transfering and three

Hello, I have an Asterisk 1.2.10, with a TDM with 2 FXO modules, and 2 GSM Nokia32. I configured the TDM with loopstart signalling. For a few days, all works great: Nov 9 09:28:54 VERBOSE[19103] logger.c: -- Called g1/6XXXXXXXX Nov 9 09:28:54 DEBUG[19103] chan_zap.c: Exception on 14, channel 15 Nov 9 09:28:54 DEBUG[19103] chan_zap.c: Got event Hook Transition Complete(12) on channel 15

Hello to all, I post again (last time subject: Line drops strange problem(got event On hook) because i have caught in debug a situation where i get a call and the line drops and i get a call from the same caller and the line works well and the call normally closes by both parties. The only differences i find are underlined. If someone can understand the reason why the line drops from the debug

Hi All, I have a rather peculiar problem. Whenever I dial out over ZAP/g0 the phone will just ring and ring, even if I answer the phone on the other end. Whats strange is that the * phone will continue to ring even after I've answered and (sometimes) hung up the dialed phone. If I make an extension to just directly dial out on ZAP/1, its almost the same behavior, it will continue to

Hi, Does any one experience that SIP phone to SIP phone (Polycom phone) calls can't hear each other, but Monitor application records both end's voices. It also happens in group pickup calls. Zap calls to queue (Local channel) also experience this problem (sometimes, our SIP phone can't hear any voice from incoming Zap calls when pickup, sometimes this happens after 10-50

Hi all, Have been experimenting with agents and queues instead of placing calls direct to a user's phone extension, but I've run into problems with calls to both the agent and the extension which creates a zombie and double records calls abandoned etc. We're using a unique queue for each agent (only a handful of users) to try and get some agent/queue information to see what the

I have a situation when I dial out my Zaptel I am getting a recording that I need to add a 1 or a 0 and the area code with this number. I have tried appending this and the number going out the zap is 1NXXNXXXXXX so it is going out with 1 and the area code. Someone has suggested that maybe the zaptel is dialing too fast. My question is how can I add a pause before dialing to test this out. I am

Hello to all, I have a strange problem with my asterisk. Line drops while i am in a call and without a reason.The line drops no matter if it is a incoming or outgoing call and it happen while i am talking on the phone (no silence detection problem). I tried to debug the situation and the only strange thing is the "got event On hook" (i guess..). I am thinking that it is a problem

hi there, I've been using sms a few months ago with * v1.0.9, but now I need it, so I'm testing it out again. But for some reason the SMS receiving doesn't work like it should. It receives the "call" from the telecom operator, and it starts the SMS application, but then i get the following error.. Any idea if this is due to bristuff or my implementation of SMS ? (I used

Hello List, I am having a strange issue with a trixbox system we installed for a client, and I would appreciate any help on this one. The issue is that occasionally when they go to answer an inbound call from the Sangoma A200 - there is no one there, and they are presented with dial tone. The calling party is hung up. A bit of background: The client actually has two systems install (one at

Hi guys, First of all, I know that this server must be upgraded asap, I'm just wondering if anyone of you has already faced this problem and , if so, would the upgrade solve my problems... CAPI version 0.6 Asterisk 1.2.5 AGI scripts are being used Main problems: -Dropped Calls - ps aux | grep asterisk shows that asterisk (that is started with safe_asterisk) is generating multiple

Hello, Can somebody please assist me with my problem. Currently I am using a Asterisk@HOme version 2.4 with a TE406P digium card. One the E1 is connected to a telco switch via an ISDN. May issue is that may incoming calls in the zap channels gets disconnected or release after 1 ring. I really dont know what setting should I change to increase the timeout of the ring. I have even tried upgrading

Hi, sorry for the long debug output below. I configured Asterisk with AMP to send the whole number including the extensions of the callers to the called party. Whatever I configure in AMP it looks like it is used, In my eyes it is ok, but doesn't seem to work. 033811234451 is the call id i configured, and it seems to use them, but the caller will only see a 0338189040 instead of my

hi, after upgrade from 1.0.x to 1.2.x i cannot send faxes my topology: PSTN<-wct4xxp-asterisk- -sip- ata (ht496,ht488,asus vp100) - samsung sf2500 fax log: Feb 13 23:50:35 DEBUG[27914] chan_sip.c: Allocating new SIP dialog for 20d700003cb20000@192.168.1.209 - INVITE (With RTP) Feb 13 23:50:35 DEBUG[27914] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE Feb 13 23:50:35

After a power interruption, asterisk doesn't seem to be routing calls and there seems to be a premature timeout and hangups occurring. I am clueless where to look. Can someone in the know, look at the following log and enlighten me if there's a problem, or if it looks normal. From the calling phone, it keeps ringing as if never picked up. Thanks soo much. -braman

I have the following in my extensions.conf [ext-local] exten => _53XX,1,Wait(2) exten => _53XX,2,NoOp,Dialing ${EXTEN} from ext-local-custom exten => _53XX,3,Macro(dialout-trunk,2,${EXTEN},,) This is used to match inbound caller-id for my legacy PBX. It works fine for inbound calls, but not for internal SIP calls. If I call from a SIP phone that is also in [ext-local], it looks like it