Hi,

Asterisk is making a call to a peer. In 200 ok, peer is sending its
application server ip in contact field, so asterisk sends ACK to that IP.
RTP starts flowing between endpoints and peer plays an IVR and asks for
destination number. After entering destination number peer’s application
server sends INVITE again with different media IP and asterisk accepts with
200 ok. RTP from peer comes from new media IP but asterisk keep sending to
their old media IP that came in their 200 ok before and don’t send to new
one. Hence, I can hear their voice but they can’t.

Does anyone know how to make asterisk send RTP to new media IP that came in
new INVITE within the call?

Thanks

Deepika

Hi,
I dial on A* from a linphonec to a Playback() extension, then suddenly
the sound stops after a while, without any notice.
I enabled debug both in linphone and A*, and the RTP packets are sent
from A* and received from linphone. It doesn’t matter whether I choose
alaw, ulaw, gsm as codec (besides changing cpu load of course).

How can I debug it? I’m using A* 1.6.2 and both linphone 2.x and 3.x.

I just need a console scriptable softphone, so maybe there’s an
alternative to linphone (which seemed good enough anyway!)…

Thank you,
Matteo

Hi All

In the console I am seeing warring rtp.c:1635 ast_rtp_read: RTP Read too
short

I get these all of the time things seem to be working fine but I am trying
to figure out if there is a way to resolve these Warnings.
I am running asterisk 1.6.2.13

Any direction is appreciated.

Thanks
Bryant

Hi. I am having a very strange problem –aren’t they all — with the
release candidate. I have softphone which talks to asterisk from behind
nat — the asterisk is on a public ip — and when I hit mute on the
softphone, all rtp traffic ceases! Now, a version which does work is
r281875, this does not happen in that vrsion, but right after that this
strange thing starts and is not fixed in the current one.

Any assistance here would be appreciated.

I’m building a paging system composed of roughly 10 switches in daisy
chain, with an embedded box with a speaker and a microphone for each
switch. The embedded box runs my software.

I need the system to be resilient to any network partition, so that
anyone can send announces from any mic to all the reachable clients. I’d
need also to page a subset of all the speakers.

I’m currently using some software I wrote which sends voice over
multicast RTP and coordinates all the sites with multicast messages.

I don’t own the switches so each site will be assigned an address by
DHCP, that’s why I’m using multicast.

I heard of asterisk and SIP as a possible alternative to my software,
and I’d rather use tested and widely adopted software.

Is there a way asterisk could be of use, or would I need to bend it too
much?

Thank you in advance,
Matteo