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 chan-9-backward-804-20130507123100.call .....
[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]…
In article <4FECCD0C.email@example.com>,
> 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…
I short question:
I want to connect Asterisk to OpenBSC with mISDN, mISDNuser and LCR.
Do I need chan_lcr?
mISDN .v2 integrated in Kernel 3.0.22
HFC-E1 Evaluation board from cologne chip
I tried to configure Asterisk with <./configure --prefix=/usr/src/lcr
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 184.108.40.206 - 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]@sipgate.de). My sip.conf including the codec restrictions looks like this…
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: