On 100222 1313, JT wrote:
When a SIP device dials another SIP device...Asterisk connects the calls and
displays the channel information.
If one of those SIP devices hangs up, Asterisk receives the hangup notice
and disconnects the call/channel.
However - what does Asterisk do when the network
On 100222 1818, Kevin P. Fleming wrote:
Kirill 'Big K' Katsnelson wrote:
The caveat here is that it is perfectly normal NOT to transmit any RTP
data in case of long silence. This is why the SIP timers were introduced
in the first place: there is no correct way to detect when the client
I have a private trunk with a peer. A call comes in from the trunk, and
Asterisk calls a peer agent. If the agent transfers the call (this is a
blind transfer using REFER), I want Asterisk to send a REFER back to the
trunk, and essentially stay out of the loop.
As set up, Asterisk initiates a
Sorry, totally forgot to mention that this is a SIP trunk.
Asterisk version is 1.6.1.6.
On 091207 1130, Kirill 'Big K' Katsnelson wrote:
I have a private trunk with a peer. A call comes in from the trunk, and
Asterisk calls a peer agent. If the agent transfers the call (this is a
blind
On 091007 0001, Hans Witvliet wrote:
Obviously, it is an outdated document that should be revised.
It stated:
Now sip supports more than asterisk sip supports. SIP RFC requires tcp
support for example, yet its not in trunk yet.
This feature is already in the 1.6-branch, production. You
On 091001 0406, Mindaugas Kezys wrote:
We had many problems with IAX2, changing to SIP solved them all.
Let me paste link to wise-words which clearly illustrates our experience:
http://wiki.kolmisoft.com/index.php/Why_we_do_not_suggest_to_use_IAX2
Thanks for the wise-words. From the
On 091006 1249, Darrin Henshaw wrote:
1. A call comes into our Asterisk system, it's trunked from one office
to another via IAX.
2. Call enters a queue and is picked up by one of the agents.
3. That agent has to transfer the call, could be for a number of
reasons the client wanted someone in
Alan Lord (News) wrote:
On 01/10/09 00:57, Kirill 'Big K' Katsnelson wrote:
Asterisk 1.6 behind a firewall and NAT. We are terminating multiple DID
calls, originating and transferring.
A provider offers both SIP and IAX trunking. Cateris paribus, what is
the preferred solution to choose
Asterisk 1.6 behind a firewall and NAT. We are terminating multiple DID
calls, originating and transferring.
A provider offers both SIP and IAX trunking. Cateris paribus, what is
the preferred solution to choose? What points to consider?
I can name the provider if this is not against this list
Steve Edwards wrote:
Some say the audio quality is better with SIP. My experience has been with
low volume (xx) calls across the internet and high volume (xxx) within
the same cabinet.
My understanding was that IAX encapsulates the same RTP traffic, or, and
the very least, same stream of
Steve Edwards wrote:
My understanding was that IAX encapsulates the same RTP traffic, or, and
the very least, same stream of data encoded by a codec. Is that not true
in case of IAX? How can a transport protocol affect volume--or quality
(lest it is dropping packets)?
My (limited)
Brian Camp wrote:
What unit is dtmftimeout measured in?
In samples, 1/8000 of a second each, or 125 us if you prefer.
The sample configuration is provided below. Does it mean...
; The amount of time a DTMF digit with no 'end' marker should be
; allowed to continue (in 'samples', 1/8000 of
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