mfcr2 channel state IDLE 0x00 and call trace log file not ended ??


Hi, would be glad to be contributing with this question to all the comunity.

I´m having a weird issue, suddenly I get Channels on IDLE 0x00 state until I do a dahdi restart.

Trying to do a call trace to see whats going on deeper, I get surprised when tried to open the .call file to see the log this is incomplete.

This is what I see

[root@localhost telefonica]# nano .....

[12:31:01:846] [Thread: 3067460464] [Chan 9] - Getting ANI digit 3 [12:31:01:846] [Thread: 3067460464] [Chan 9] - ANI so far: 1152723, expected length: 10 [12:31:01:846] [Thread: 3067460464] [Chan 9]…

Asterisk Users 2.3 years ago 0 Answers

PRI trunk between Asterisk servers does not work.


In article <>,
James Sharp wrote:
> On 6/28/2012 3:53 PM, Ernie Dunbar wrote:
> > We have three Asterisk servers, Voip1, Voip2, and Voip3. Voip1 has a
> > Wildcard TE405P (3rd Gen), Voip2 has a Wildcard TE410P/TE405P (1st Gen),
> > and Voip3 has a Wildcard TE110P. Voip1 is the server that handles our
> > PRI to the PSTN and we hope will allow us to failover to other Asterisk
> > servers (ie, Voip2 and Voip3). Voip2 is our current production server,
> > and Voip3…

Asterisk Users 3.2 years ago 4 Answers

Asterisk with LCR -> chan_lcr needed?


Hello, I short question: I want to connect Asterisk to OpenBSC with mISDN, mISDNuser and LCR. Do I need chan_lcr? I have:
Asterisk 1.8
mISDN .v2 integrated in Kernel 3.0.22
lcr 1.7
HFC-E1 Evaluation board from cologne chip
I tried to configure Asterisk with <./configure --prefix=/usr/src/lcr

Asterisk Users 3.2 years ago 0 Answers

Error SIP/2.0 488 Not acceptable here


Hello, a person trying to call me by my phone number is getting the error 488 Not
acceptable here. I googled that error, seems like this error is normally
caused by a failed codec negotation, though I have no clue how I could have
read this out of the logs. Anyway, my setup is as follows:
Asterisk - NAT - Sipgate SIP Provider
The user calling me is also using Sipgate and is calling my landline phone
number from Sipgate (not [my sip id] My sip.conf including the codec restrictions looks like this…

Asterisk Users 3.2 years ago 3 Answers

No compatible codecs, not accepting this offer! - after upgrading to 1.8.11


Hi, I've upgraded my asterisk 1.4 to the version 1.8.11. After making some
adjustments to the configuration files to port it to the new version, calls
between registered phones in asterisk, work fine, but inbound calls coming
from the SIP trunk I have with a telco to asterisk, don't work anymore. I
don't know why!... This is the SDP portion that comes in the INVITE messages of calls through
that trunk (let's say, whose endpoint has the IP x.x.x.x, purposely
omitted). Nothing seems to be wrong with that to me:

Asterisk Users 3.4 years ago 4 Answers

fake auth rejection??


Very occasionally in my logs I see things like this. In this case 7
lines starting with the first line and each ending with one of the
group of 7. Took about 10 seconds for the 7 tries. [2012-05-03 16:58:27] NOTICE[31850] chan_sip.c: Sending fake auth
rejection for device "unknown" ;tag=aTZ1eFu5Gi
;tag=CPW3Z9lDvN Should I be worried? Ira

Asterisk Users 3.4 years ago 0 Answers

chan_dahdi with asterisk 1.4 and new Linux versions


Hi All; First of all, I am trying to install vicidial and actually vicidial requires asterisk 1.4 and can not work with asterisk 1.8, in addition there is a special version of asterisk 1.4 that is required for vicidial which is asterisk- The problem that it look like there is a problem with that asterisk 1.4 is not able to buld chan_dahdi if the linux version is new (like fedora core 16), while asterisk 1.8 can work fine with new linux versions. What is happening with me that when I used fedora core 16, I compiled and installed dahdi 2.6…

Asterisk Users 3.4 years ago 3 Answers

Auto answer Asterisk ; Unable to create channel of type


To fix the issue below, Edit /etc/asterisk/modules.conf load => noload => load => Edit /etc/asterisk/extensions.conf exten => s,1,Dial(Console/snd/controlC0,20,A(trek)) exten => s,2,Hangup It worked perfect; one little issue that I have not fix is that I have to use this command #pulseaudio -vvvvvvvv Restart asterisk and it worked but if pulseaudio is not on asterisk won't work, I'm in the process of fixing this issue. Thanks,  

Asterisk Users 3.4 years ago 0 Answers

chan_mobile with Nokia 6021 - incoming SMS causes call to drop


Hello, I'm Using Asterisk on Debian Squeeze. I was experiencing problems with ${SMSSRC} being blank, so I applied this patch: but otherwise everything is standard. As the subject says, if I am making a call through the phone when an SMS is received, the bluetooth connection drops and the call ends. The SMS is delivered successfully. This happens on two different 6021 handsets, one is connected to a DBT-120 (csr) dongle and the other is connected to a no-name btusb dongle. Nothing unexpected shows up in the Asterisk console: default iconsmime.p7s

Asterisk Users 3.4 years ago 0 Answers

Operation not permitted ??


Anyknow know this problems ?
I read on the net that it's a possible network problems, but i don't think
because it's a VMWare server and in the same server i have other
asterisk without this problems. best regards
Le 25 avril 2012 09:35, Olivier CALVANO a écrit :
> Hi
> i have a lot of error in the CLI of one of my Asterisk:
> [Apr 25 09:30:46] WARNING[10542]: chan_sip.c:2901 __sip_xmit: sip_xmit
> of 0x8ef2130 (len 867) to returned…

Asterisk Users 3.4 years ago 2 Answers