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Flowroute: Howto Set Outbound Callerid (ast 1.4)?

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.

1.8 busypatterns

Hi,

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.

using Wifi smartphones as SIP clients

All,

has anyone any experience in using Wifi smartphones as SIP clients? Does
this work properly? What models/brands are optimal for this (in terms of
ease of use, battery life etc)?

Thx!!

B.

Lifetime & Replacement

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 are still good)
> work with a TDM410P?

Yes, the module are compatible with a TDM410P, or any other Digium card
supporting analog modules except the TDM2400P.

HELP!! Caller ID “unknown” for all inbound call (Satria Anamarta)

Hi Anam

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
E-Mail mcgrathr@mail2web.com

Call Transfer Not Working

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