We solved the problem by adding this string:
p->dialing = 0;
in chan_dahdi.c, static void *ss7_linkset(void *data) after the case
ISUP_EVENT_CON:
case ISUP_EVENT_ANM:
Thanks,
Dave George
Teletone Inc.
561 674 3838
From: asterisk-ss7-boun...@lists.digium.com
[mailto:asteris
Hi Gustavo,
I think one solution for this case is send and receive the audio during the
early media (183).
Asterisk when receive a ANM from pstn side not forward the 200 Ok to SIP
side and establish the audio during the early media (183).
Does anyone know if it is possible ?
Regards,
Bruno Rod
I tried that several months ago with libss7, but remember that 183 with no 200
means that the A side will wait for a 200, so you can have the call active for
2 minutes in some countries (less time on others), after that timer expire the
call should be released. If Ast receive an ACM with optiona
The linkset is up but it seems as if the second D channel on slot 25 is not
aligning. I am using 1 linkset and I am able to pass calls on both T1s.
I am getting the following error:
aislecomss72*CLI> ss7 debug linkset 1
Enabled debugging on linkset 1
Link state change: NOTALIGNED -> I
I have a many diferents devices in other side like cisco gateways, ATA and
asterisk box.
For my problem 2 minutes is a good time because it's happens when telco send
a error message and this messages has a small time (15s).
To this error messages 2 the audio in early media will work but if you
For more informations bellow the RFC 3398 about ACM
http://www.packetizer.com/rfc/rfc3398/
7.2.6 ACM received
Most commonly, on receipt of an ACM a provisional response (in the
18x class) SHOULD be sent to the SIP network. If the INVITE that
initiated this session contained legitimate
Under what circumstances should you legitmately have early media up for
longer than 30 seconds?
Bruno Rodrigues de Mello wrote:
> I have a many diferents devices in other side like cisco gateways, ATA and
> asterisk box.
>
> For my problem 2 minutes is a good time because it's happens when telco
I've seen some complex announcements that may have more that 30 seconds, like
numbering changes.
On 6 Feb 2010, at 19:08, Paul Timmins wrote:
> Under what circumstances should you legitmately have early media up for
> longer than 30 seconds?
>
> Bruno Rodrigues de Mello wrote:
>> I have a man
That's was exactly my issue, in several countries the 800 numbers MUST be sent
as No Charge, and it's illegal send an ACM with charge for that kind of
traffic. I solved that using X- header in SIP to let the other side knows that
call must not be billed. Let me try to find those patches for libs
