Dear In essence Caller ID ETSI and FSK US (Bellcore) is based on the same pattern as;
_____________ ___________________ _________ ________________ _____________
|First Ring burst |_500ms_|Channel seizure 300 bits|__|Mark Signal|__| Caller ID Message|_200 ms_|Second ring burst | So basically any kind of device should be work without any problem, unfortunately during these process if some noises (as miss ground connection or others) happens during the process can make failed to process caller-id information, by the modem. Mc GRATH Ricardo
E-Mail mcgrathr@mail2web.com

Asterisk Users 3.2 years ago 0 Answers

sip show peers


I have a process that runs on a server and does a simple 'asterisk -rx
"sup show peers' > /tmp/peers"
and then looks for any "(Unspecified)" items and reports them as having
lost connection.
My server is running 1.4.43 and the two boxes I am monitoring are also
running 1.4.43.
Once in a great while 1 of my boxes reports "(Unspecified)". I am trying
to find out why. How can I make the remote boxes have a shorter heart beat to checking
more frequently
with the server so as not to…

Asterisk Users 3.2 years ago 0 Answers

Slow AMI Originate


Hello, We use AMI to originate calls. Sometimes, lately every morning, the AMI Originate process operates extremely slowly. I cannot see the calls in "core show channels verbose", I don't know where they are, what state they are in, after 2-3 minutes the calls go through one after the other. As mentioned, it usually happens in the morning as soon as people start their workday, where there are a lot of logins and calls being made, but no where close to a peak in terms of simultaneous channels, etc. In some cases restarting asterisk, in others just taking…

Asterisk Users 3.3 years ago 1 Answer

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 3.3 years ago 4 Answers

Asterisk Directmedia


What is directmedia?

"directmedia" is the new configuration option name for "canreinvite"; they are the same feature.

To put it simply, is the process where Asterisk tries to redirect the RTP media stream to go directly from the caller to the callee. Be careful that some devices do not support this (especially if one of them is behind a NAT). The default setting is YES, for example in the SIP protocol configuration file sip.conf.…

Asterisk Tips 3.3 years ago 0 Answers

asterisk 1.4.39 and dahdi 2.6 on Ubuntu


On 04/19/2012 05:59 PM, bilal ghayyad wrote:
> Dears;
> I see this at the /var/log/asterisk/messages:
> [Apr 20 01:49:48] ERROR[1657] codec_dahdi.c: Failed to open /dev/dahdi/transcode: No such file or directory If you aren't using a DAHDI transcoding card, then you don't need to
load the codec_dahdi module in Asterisk. Since it was built, though, you
clearly have DAHDI built and installed properly, and the Asterisk build
process was aware of that. >
> Again, I am installing asterisk and dahdi at Ubuntu (uname -a
> Linux House…

Asterisk Users 3.3 years ago 0 Answers

Monitoring voice-quality with sip/rtp/rtcp


After some customer complaints I find myself tcpdumping, gzipping and
transferring large packagedumps over the network to be analyzed. While this manual process isn't a long-term solution, I'm evaluating
different options. Aside from the manual thing I could see two variants:
- Dump the traffic (on the server or another via switch port
mirroring/monitoring) and analyze it with tshark
- Analyze the traffic in asterisk How do you monitor call quality for you services? (Right now I use
asterisk 1.4, but are in the process to migrate to 1.8 or 10.) I looking

Asterisk Users 3.4 years ago 4 Answers

Multiprocess Asterisk


Hi, Is anybody running multiprocess of Asterisk on a server ? Does it work
well? My configuration is too complicated. I know Asterisk on a
virtual machine works well. but OS overhead is considerable. that is
why I want to divide a process. Regards,

Asterisk Users 3.5 years ago 2 Answers

Deadlock detected in asterisk- x86_64


I am having problems with a deadlock in Asterisk The system is a x86_64 machine that is being used as a callcenter. The agents log in via the AgentLogin application, and each Agent/XXXX channel is assigned to one or more queues. A custom separate process generates calls into the queues for the agents to
answer. The calls all go out through a SIP trunk, and all of the agent extensions are SIP. After an hour or so, asterisk deadlocks. Any attempt to run "agent show" or "agent show online" through the console hangs. Also, AMI events seem to…

Asterisk Users 3.6 years ago 0 Answers

audio , Failing due to no acceptable offer found


On 01/28/2012 10:22 AM, Din Assegaf wrote:
> Hi All,
> I'm trying to upgrade asterisk server to 1.8.x from my asterisk 1.6,
> But when making A Call from SIP Client, I got cli Warning ... and no
> call has been made.
> My Sip Client is using lib java peers client http://peers.sourceforge.net/
> with standard codec PCMU/PCMA
> [Jan 28 23:03:32] WARNING[1654]: chan_sip.c:8942 process_sdp:
> Unsupported SDP media type in offer: audio 0 RTP/AVP 0 8 101

Asterisk Users 3.6 years ago 1 Answer