Hello,

We are in the process of debugging a voice quality issue for a client of
ours that is a VoIP services provider. The client uses a softphone that
runs on a pjsip stack.

When placing a call using the softphone, it negotiates the use of G729
codec with the remote endpoint (ptime = 20ms). The endpoint transmits RTP
packets with encoded G729 payload. VAD/DTX is enabled. We see that the
last frame transmitted by the carrier side endpoint, before the beginning
of a period of discontinuous transmission has 20 bytes of payload. We have
verified that VAD/DTX is used by the carrier side endpoint by noting that
there exist successive RTP packets that differ by 1 in their sequence
number but have a timestamp difference > 160 and MARK bits are set in the
RTP header.

Our understanding is that for G729B, the SID frame that is transmitted
before a period of discontinuous transmission has a size of 2 bytes.
However we see that ALL RTP packets sent by the carrier side end point has
a length of 20 bytes.

Has anybody else seen this behavior from a carrier side endpoint ? Is
there an RFC or document that specifies

Vitelity seems to be offline to both IP and voice traffic. Is there any
place to find out what their status is?

Roger Marquis

Is anyone else using Vitelity right now and having an issue with a global
outage of sorts? Potral/WWW arent accessible and it would appear through
monitoring that the outbound is flapipng like mad. The outbound can be
rerouted, I know, but inbound is a huge problem right now.

[Sep 4 10:26:13] NOTICE[27507]: chan_sip.c:15679 sip_poke_noanswer: Peer
‘vitel-outbound’ is now UNREACHABLE! Last qualify: 1193
[Sep 4 10:26:23] NOTICE[27507]: chan_sip.c:12528 handle_response_peerpoke:
Peer ‘vitel-outbound’ is now Reachable. (176ms / 2000ms)

why does this not work? i simply want to hear the recorded message

exten => s,1,Answer()
;exten => s,n,Record(zipcodegutter1.gsm) ;zcg1
exten => s,n,Playback(zipcodegutter1)
exten => s,n,Dial(SIP/c000001s/12222222259,120,A,(demo-thanks))

Hi,

I am implimenting a solution for a radio station where by calls are
first received by an attendant, who interviews the caller and then
places the call in a queue along with some information about the
caller. The radio presenter can then choose which call to pick up
depending on those in the queue.

My question is, how can it be possible for call to skip other calls
in the queue and be picked up? Are queues the best mothod of
implimenting this?

Thanks very much for your help.

Tim