similar to: Zaptel dialing too fast?

I have the latest bristuff, a zaphfc card for external calls and a PRI card for connecting to a PBX as a channel bank. With a BRI I would expect to be able to have two incoming calls going at the same time, but when I try it, one call connects and the other gives the following console message, a busy tone, and then a hangup: Mar 23 19:18:35 WARNING[5595]: chan_zap.c:7512 zt_pri_error: PRI:

# For linux-cifs-client: this paragraph is for the Debian bug tracking # system control robot. Please ignore it. reassign 447153 linux-2.6 thanks On Fri, Oct 19, 2007 at 12:03:01AM +0200, Michal Suchanek wrote: > On 18/10/2007, Colin Watson <cjwatson at debian.org> wrote: > > On Thu, Oct 18, 2007 at 03:32:27PM +0200, Hramrach wrote: > > > When copying to a cifs share scp

Hi, can anybody tell me what the errors mean and why my asterisk server falls from time to time. From time to time means several hours, not regularly. I also can provide a core if someone can debug? Thanks and regards Jan 11 14:34:59 NOTICE[13573] chan_zap.c: Hangup, did not find cref 83, tei 64 Jan 11 14:34:59 WARNING[13573] chan_zap.c: Hangup on bad channel 0/2 on span 8 Jan 11 14:35:03

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've been running Asterisk with a TDM400P for about 6months, no problems. 2 in/outgoing analog lines, one analog phone. Recently I was messing with the XTEN client, got to finagling with things, and not knowing what was wrong I upgraded from 1.0.7 to 1.0.9 (both asterisk + zaptel). I was testing various things, and found everything worked except outgoing calls. So I checked

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

This was posted at The Asterisk Blog Forums <http://asteriskblog.com/forum/> Click here for the original post.<http://asteriskblog.com/forum/viewtopic.php?t=20> I need someone to explain how the dialplan rules work? I'm having a hard time getting it. I know that to dial out you need a 9 and to ignore that 9 once your out... requires a switch of sorts that tells asterisk to ignore

Dear All, I just setup the Asterisk with E100P which it's no problem in Dial In but I have problem when outdial. The connection method is like this : E1 PRI <-SIGNAL-1-> MaxLink (PBX) <-SIGNAL-2-> E100P <-> Asterisk <--> SIP \-----> Analog PHone Now when I tried to dial out by SIP X-Lite on Windows, it shows me Connect, Trying,

hello everyone, Last week I installed asterisk 1.2.24 with digium TE220B card. I have a problem with our PRI and Asterisk: the call be interrupted.It happens either PSTN-to-SIP or SIP-to-SIP,almost every call. After spending several days searching on internet, I found a lot of discussion about this issue and I have tried many, but it still.I am totally new to Asterisk environment and suspect I

We use an IPcortex PABX running Asterisk 1.2.39-BRIstuffed-0.3.0-PRE-1y-y. We have recently implemented Call Queuing for our main incoming line with hold music. The call queue type is: Ring all - One call at a time (no position announcement). Since implementing this feature we've been receiving the below error daily in the mornings and lunchtime when the queue will jump to the next available

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

I have a trixbox 2.2 and Nortel santral that are speak each other. I use digium TDM100M 2 fxs-2fxo. After I made yum update I had met with some problems when I want to make any call from extension of trixbox to extension of nortel. When I attend to log (/var/log/messages) I meet with these messages as you see below. When I try to make any call from trixbox extension the call seems established but

Hi gang, Thanks to other help I''ve gotten on this list, I''ve got backgroundrb up and running using postgres alongside my rails app. I''ve designed the system to have several eternally-running workers that periodically poll the database to see if there''s any work for them to do, and set things up in the config file to make them restart quickly if any of them

Update: I can reproduce the error by putting the reception phone (call queue 0) in Do Not Disturb mode, then call in from outside using a mobile, then pick up the call from the 2nd phone in the queue. It will cause the error only if I hang up _before_ the mobile hangs up. The error doesn't seem to happen if the external call hangs up, or if the call is answered by the reception phone (first

It seems consistant after dialing dozens of times that the call that doesn't go through is the one the gets the log message "No echocancellation requested" (chan_zap.c) and the "Scheduleing timer" (channel.c) in the middle of receiving the DTMF tones. I'm now using the T400P card last week very simular problems the the T100P (although I think I was actually loosing

Dear All It is a little hard to give a good small example of my question,so I will show the full data on the bottom and the attachment.Maybe some one could tell me an appropriate way to show it.I'm sorry for the inconvenience. Q:How to generate a 53*53 diagonal matrix by my data Some problems confused me are that: 1.Since it is a diagonal matrix,I have tried to transform col1 and col2 to

Hi, I am a libvirt qe, I am testing the function about the new location of disk auth element (as sub-element of the source element) in Backing Chain Management. And I have a questions: When all the backing chain are iscsi network disk type(no matter if the authentication is exist), start the guest , it will only has the top level disk in xml and there are no other backingStore in it. The test

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

Hi Everybody, I am trying to use SIP (Sipura 2000) to connect to Asterisk which then dials out a local number using the Digium E100P. We have purchased the G729 codec licenses from Digium and loaded them into Asterisk successfully. However, the call drops immediately after being answered with the debug error message saying something like: "channel.c:2646 ast_channel_bridge: Didn't get a

Hi, Some calls I make trough my PSTN asterisk gateway just hangup after some minutes. Even if I'm using sip or iax. I have callprogress=no busydetect=no in my zapata.conf. Anyone help? Or tell me what to look at /var/log/asterisk/debug. I didn't find anything wrong. [endpoint]---iax or sip----[asterisk]----E&M----PSTN. As endpoint I had tested another asterisk box (with a FXS),