Hi
Use meetme() function for release sip channel .
your dahdi channels are working or not .Without gotoif( ) its work r not .
On Fri, Jun 11, 2010 at 1:27 PM, dinesh bansal
wrote:
> Hi regarding this,
>
> I receive the calls form sip then forward it to ss7 can this help you ? as
> ther
hi,
i'm testing 1.6.2.x asterisk with chan_ss7 1.4.1.
there is problem with jitterbuffer enabled in chan_ss7.
i cannot send fax. without jitter buffer it is ok.
do you know what changed between asterisk 1.4 and 1.6 in jitter buffer
code?
are you using jitter buffer enabled in libss7 without pro
One of my customer is not getting any ringback from me. He is sending sip
to my asterisk ss7 box using libss7 with TE410P card.
I tried the various option (yes, no and never) for progressinband in the sip
profile and none worked.
Customer is using genband SBC
The customer wants:
100 trying
180
Asterisk does not provide SIGTRAN implementation.
Only MTP and ISUP are implemented.
You can use the MTP stack and implement SCCP, TCAP,...etc.
A question: Can you give us an idea about what you want to do with BSSAP.
Just for information. And, does it relate to Asterisk.
--
Hi,
Try the dialplan like this to get Ringback Tone,
[wholesale]
exten => _473.,1,Dial(DAHDI/g1/${EXTEN},r)
exten => _473.,n,Hangup
On Mon, Jun 14, 2010 at 10:28 PM, dave george wrote:
> One of my customer is not getting any ringback from me. He is sending sip
> to my asterisk ss7 box using
