similar to: Frame too large?

Hi all, I must admit that I am stuck. I have a TDM400P card with two FXS and two FXO modules which I had set up and configured so that it was working beautifully. The only problem was that occasionally it would get itself into a state where outgoing calls would simply be met with a very loud static. A reboot would fix this issue and everything would work fine for a while. Recently however,

This was submitted to the Dev list last week, but there was no response, and perhaps it wasn't the right group. I am developing an application in which I need asterisk to pass on an incoming call to a separate IVR server. The problem is that asterisk appears to hang up while the IVR is playing back a sequence of recorded voice and systhesized voice prompts. My setup is: Analog line

Any ideas on this warning? If I call this number, sometimes I get this error and sometimes the call goes thru fine. Why would it work sometimes? -- Executing Goto("SIP/3044-8d49", "cytel-outgoing|915124512424|1") in new stack -- Goto (cytel-outgoing,915124512424,1) -- Executing SetCIDNum("SIP/3044-8d49", "2814494000") in new stack --

Hi All Just got a new quadBRI card and connected one port to our Old PBX. When I make a call from a sip phone to a phone number the phone rings, I hook up, and the call on the sip phone allmost imidialely disconnects, after a fiew seconds the "real" phone disconnects too. Here is a trace: -- Executing SetCallerID("SIP/cramer1-b718", "45") in new stack --

Hello, I have problem to make SIP calls, i have asterisk + PC InterP4 + Digium TDM400P here is the content of the sip.conf: [SIP_PROVIDER] type=peer fromuser=testcomclient username=testcomclient secret=testr host=IP_SIP_PROVIDER ;dtmfmode=rfc2833 context=interne canreinvite=no ;allerid=Beer disallow=all allow=ulaw allow=gsm allow=g723.1 ; Asterisk only

Any ideas on this? This has always worked in the past until we got this new customer and his new fax machine. Faxing from my machine works fine. But faxing from his gives this error. I can't imagine why his fax machine would be causing frame errors. -- Redirecting Zap/4-1 to fax extension -- Executing Macro("Zap/4-1", "fax-handle|7133950500|2815692740|Y") in

I am having some problems with g729 with SIP and ZAP channels. 1) I have two g729 licences. Very frequetnly (I don?t know what triggers the error) I get the following warnings and error when I try to place a call via SIP to my X100P. The only way to get out of this is through a restart of *. When the error ocurrs there are no other calls in place. Any ideas? Error Opening channel:2 not

Hello when i dial a toll free no using sipphone i get this error message. How do i solve this? Any help will be appreciated. console message: Starting simple switch on 'Zap/2-1' -- Executing SetCallerID("Zap/2-1", "17473863282") in new stack -- Executing SetCIDName("Zap/2-1", "Deepak JV") in new stack -- Executing

Sorry in advance if this is a FAQ... I've got a working Asterisk setup based on A@H 2.2. I have a TDM400 card with 2 FXS and 2 FXO ports; PSTN connections come in via the TDM card. I haven't been able to get inbound fax with spandsp and rxfax to work. Occasionally an all-text fax will come in, though it's usually badly corrupted, but in most cases, it would appear that the call is

Hi All, For the most part, the PBX works as it should. Occasionally people complain that they call and the PBX doesn't pick up. Other times it looks like the call is answered by Asterisk but I still hear ringing and I start listening to the IVR menu a few seconds into it. As for Asterisk not picking up, I see the following in the logs: [Mar 27 13:32:29] NOTICE[13197]: chan_zap.c:7071

Hi Guys, I have a really odd one here. We are dropping calls occasionally... there are no error messages being spat out, but I can see this suspicious behavior in the debug logs; Jun 30 14:58:48 DEBUG[19856] pbx.c: Function result is 'Other' Jun 30 14:58:48 DEBUG[19856] pbx.c: Function result is '(null)' Jun 30 14:58:48 DEBUG[19856] pbx.c: Function result is 's' Jun 30

Using tdm400. While transfering a call from outside to another extensions, while this "outside call" is waiting with music, the "another extension" call hangs up suddenly, and the call is back to the "outside call" suddenly. Wathcing logs: Jan 15 13:32:44 DEBUG[30148] res_musiconhold.c: Read 462 bytes of audio while expecting 640 Jan 15 13:32:55 DEBUG[27850]

Hi- I've posted this on the bugs list, but I'd also like to see if others have had similar problems when connecting via E1 trunks (E400P). I'm getting numerous errors like the following during inbound calls to my E1 channels. These occur when the system is under medium load: WARNING[196621]: File chan_zap.c, Line 5404 (zt_pri_error): PRI: !! Got reject for frame 78, retransmitting

Hi all, what could be the cause for the following messages? May 22 12:44:53 WARNING[3791]: chan_zap.c:8498 zt_pri_error: 2 !! Got reject for frame 0, but we only have others! May 22 12:44:54 WARNING[3791]: chan_zap.c:8498 zt_pri_error: 2 !! Got reject for frame 0, but we only have others! May 22 12:44:55 WARNING[3791]: chan_zap.c:8498 zt_pri_error: 2 !! Got reject for frame 0, but we only have

Hello friends, in aCentOS with a A400P01 OpenVox PCI I have a analog line connected. I am new in Linux and Asterisk, my steps are theese: 1. Install CentOS 4.4 (basic instalation). 2. Command line: yum -y update yum install gcc kernel-devel bison openssl-devel yum install openssl-devel 3. Download the source: wget http://ftp.digium.com/pub/asterisk/releases/asterisk-1.2.17.tar.gz

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

I have been trying for some time now to make the hook flash work on the FXS port. I am using Asterisk 1.4.10.1 with zaptel 1.4.4. When I "manually" flash the hook I can manage to find the duration to put a call on hold. However when pushing the flash button it never works. The phone's flashtime seems to be too short. I tried to set a shorter flashtime in the zapata.conf file, but it

I have a plain POTS line coming into FXO in a Digium card, this is developers kit card with 1 FXO and 1 FXS. The latency is very high, in that, it picks up after 8 rings. I don't know what I can tune to reduce to 2 or 3 rings. If it's of any help , I am posting a section of the log : ==== Apr 30 10:26:50 DEBUG[3050] manager.c: Manager received command 'Command' Apr 30

Hi, I am getting this message on the * console on my first pri span. Pri show span show it is down, and I can't make any calls from the span. Apr 27 07:40:23 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 27 07:40:23 NOTICE[23988]: chan_zap.c:8202 pri_dchannel: PRI got event: HDLC Abort (6) on Primary D-channel of span 1 Apr 27

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