Hi,
This seems to be an issue of terminology,
On Thu, Sep 24, 2015 at 05:08:24PM -0500, Jeff LaCoursiere wrote:
>
> Hi,
>
> I have a client that has a 24 channel voice T1 that I have been using e&m
> signalling over for a number of years. The local telco finally got an ISDN
> switch and wants
On 09/30/2015 07:47 AM, Tzafrir Cohen wrote:
[snip]
Right. For Sangoma cards, lsdahdi can't tell if the port is E1 or T1 and
thus calls it "PRI". Note that "PRI" here is a poor name that refers to
the port type itself and not to the signalling in it (which don't have
to be ISDN).
Suggestions? Pa
Greetings everyone,
I was wondering if there was a way to change the codec that Asterisk uses when
streaming via MulticastRTP. Or perhaps a way to transcode the multicast stream.
In the CLI, when I have a multicast stream in progress, I am typing 'core show
channel MulticastRTP/0x7f7' to
Hi guys
i'm using asterisk 11.18.0.
I need to send the pound # sign to my SIP provider, but each time it's
reencoded in %23.
I try to put pedantic=yes in the sip.conf as advised here:
http://www.voip-info.org/wiki/view/Asterisk+SIP+pedantic
but nothing's changed.
Have someone already met this
Hi Matt
Interesting problem! I'm hoping those with knowledge about the internal
workings of the Page app and multicast will chime in, although it might pay to
quote your version of Asterisk).
I don't know enough to answer the question itself, but if it were me I would be
inclined to just work
On my Asterisk 11 system I have the following in extensions.ael for
chan_sip.
8001=> {
Set(SIP_CODEC=alaw);
//Dial(MulticastRTP/linksys/224.168.168.168:34567/224.168.168.168:6061);
Page(MulticastRTP/linksys/224.168.168.168:34567/224.168.168.168:6061,q,5);
