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 3.1 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 3.2 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 3.2 years ago 4 Answers

Asterisk Directmedia


What is directmedia?

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.

When should I use directmedia in Asterisk?

If you have all clients behind a NAT, or for some other reason want Asterisk to stay in the audio path, you may want to turn this off. If you want to allow media…

Asterisk Tips 3.3 years ago 0 Answers