The flowroute website mentions that they set callerid on outbound calls based on the presence of (in order of preference): "P-Asserted-Identity", "Remote-Party-ID" or "From:". I've been trying to make outbound callerid work via flowroute to no avail. Does anyone have an extensions.conf / sip.conf snippet howto make this work? This is for Asterisk 1.4.44.
is it possible to detect 4 length pattern busy cadence detection on FXO lines in 1.8?? Here the tones are: 425Hz Pattern(0.2ms on, 0.2ms off, 0.2ms on, 0.6ms off)
in asterisk 1.4 busy detect worked
in asterisk 1.6 didn´t work and i was told that 1.6 can´t handle 4 length patterns, but what about 1.8?? for now I can only hangup by asking the provider polarity switch. Thanks best regards.
On 05/06/2012 11:42 AM, Greg Woods wrote:
> Second, since the parts of this card are very expensive, I am wondering
> if these symptoms likely mean that the main board of the card is dead,
> but the FXS and FXO modules might still be good. In that case, I could
> just get a new main card and move the modules to the new main card. The
> problem is that I can't find any TDM400P cards anywhere, all I can find
> are TDM410P's. Will the modules I have (assuming they…
It seem should be work, but I just have a question about chan_dahdi.conf regardless to parameter
rxwink=300 ; Atlas seems to use long (250ms) winks
By the way gain parameters shouldn´t have any effect to CID signal processing, how about to comment, and test again, if still without working try to connect parallel phone with Asterisk it will check if could be a hardware problem or configuration parameter setting Good luck
Mc GRATH Ricardo
I am using asterisk 1.8.11 on centos 5. I have realtime sip peers with DTMF setting rfc2833 and inband. I have also enabled blind and attended transfer features in features.conf but still call transfers dont work. I have setup transfer feature in past but i dont think i am missing anything this time. I just dont have any clue why its not working. I have tried using ATAs and softphones but cant make it to work. Can anyone help? Am I missing anything? features show output: =========================== Builtin Feature Default Current
I made a new patch from irroot's branch and I ported it to 1.8.11.
Unfortunately latest one is still against 1.8.8 and porting from
subversion is quite time consuming, hopefully my work will be useful to
Today I had no time to properly test it, so feedbacks are welcome.
Squeeze debian packages with t38 gateway will follow. Cheers,
Thousands of people from all over the world are earning substantial incomes from home just by completing simple online surveys.
This new system realy work!!!
Your First Requirements:
1-) A computer or laptop.
2-) Access to the internet.
3-) A few extra minutes of your time.
If you have this go to url and star win now. This is first section.
So get really excited...
I was using the Playback application to play an MP3 file after compiling
and installing asterisk 18.104.22.168 with format_mp3 and it seems to me that
asterisk is transcoding the file to an slin on the fly rather than
playing the mp3 itself. Is this what it does? Also, does this mean I might as well change the format of MP3s to WAV
seeing as I'm used to doing that anyway? Thanks Ish
I'm trying to send a fax with sendafax aplication and receive the fax with
the receiveFax aplication on the same Asterisk Server (1.8..8.2). All work fine but the PBX always use T30 protocol. Is thes a variable or setting to configure Asterisk to send and receive this
fax with T38 protocol only? Thank you Regards