Displaying 20 results from an estimated 7000 matches similar to: "Already on the phone?"
2023 Jul 06
1
Setting codec on originating (calling) channel with chan_pjsip (SIP_CODEC_INBOUND equivalent)
On Thu, Jul 6, 2023 at 1:43 PM Michael Ulitskiy <mulitskiy at acedsl.com>
wrote:
> Hello,
>
> After I have re-read the "PJSIP Advanced Codec negotiation" document, it
> occurred to me that the desired behavior should actually happen
> automatically, just due to the codec negotiation logic, but it looks like
> asterisk doesn't actually follow the described
2023 Jul 06
1
Setting codec on originating (calling) channel with chan_pjsip (SIP_CODEC_INBOUND equivalent)
Oh, that's great. It wasn't clear from that page, at least not for me. :-(
Having it clearly stated on the document would save me (and probably
others) lots of time.
Thanks for clarifying it. Any idea on the timeframe of implementation?
*Michael Ulitskiy*
Ace Innovative Networks, Inc.
Main/SMS: 212-868-2366
Direct/SMS: 212-812-1203
https://www.aceinnovative.com
On 7/6/23 12:47, Joshua
2003 Jul 16
3
Segmentation fault with chan_oh323
Hi,
I'm trying to interconnect sip and h323 endpoints using asterisk
and asterisk crashes with segmentation fault whenever h323
connection needs to be established. It registers with gatekeeper ok though.
Here are the symptoms.
If the call initiated by SIP device, asterisk replies to it "Trying" and then
silently crashes (it launched as asterisk -vvvvcd).
In debug log I can see the
2015 Oct 06
2
PJSIP: how to retrieve underlying SIP Call-ID
Hello,
I've started to play with PJSIP and got stuck at the following problem.
I need to retrieve SIP Call-ID associated with PJSIP channel.
For inbound channel I can use ${PJSIP_HEADER(read,Call-ID)}, but that doesn't work for
outbound channel even in pre-dial or hangup handler. Whatever I do PJSIP_HEADER
seem to be unable to read headers for outbound channel.
Here's what I do:
2023 Jul 06
1
Setting codec on originating (calling) channel with chan_pjsip (SIP_CODEC_INBOUND equivalent)
Hello,
After I have re-read the "PJSIP Advanced Codec negotiation" document, it
occurred to me that the desired behavior should actually happen
automatically, just due to the codec negotiation logic, but it looks
like asterisk doesn't actually follow the described logic which is
likely a bug.
Can you please follow with me through a simple sip call and see if I'm
missing
2003 Jul 22
2
No callerid on outgoing call over chan_h323
Hi,
Has anybody managed to get callerid properly set on a call from
local to asterisk SIP endpoint through h323-pstn gateway to a
regular phone.
I'm using ata186 as SIP endpoint. It has 12125551234 assigned to it.
When I place a call to pstn I'm not receiving 12125551234 as the clid,
but a number assigned to PRI channel by phone company.
It worked with chan_oh323, but there were other
2004 Aug 23
1
using ChanIsAvail
Hi
I am trying to use ChanIsAvail to decide if a particular extension is
available in the sip channel
I am using MySQL to hold my SIP friends.
and wy cvs version shows Asterisk CVS-08/02/04
my intention is, that if the extension is not available in Sip channel, I
will send the call somewhere else
my extensions file contains the following:
exten => _[123]XX,1,ChanIsAvail(sip/${EXTEN})
exten
2005 Mar 15
1
Not ringing phone that are in use
We have a small number of phones, when a call comes in we want all the
phones that aren't in use to ring.
Is there a simple way to test and see what phones are in use then ring
the other phones? I tried some
code like this:
[zap]
exten => s,1,Answer
exten => s,2,ChanIsAvail(${DERRICK})
exten => s,3,SetVar,"EVERYONE=${DERRICK}"
exten => s,4,ChanIsAvail(${DON})
exten
2009 Nov 03
3
Problem with ChanIsAvail
Hi all,
I am having a problem with ChanIsAvail. It always returns the same
result, regardless of whether an extension is available or not.
It always returns 0 Unknown Status.
This is my dialplan.
exten => _2XX,1,ChanIsAvail(SIP/winsor_${EXTEN}|s)
exten => _2XX,2,Verbose(0, ${AVAILSTATUS})
exten => _2XX,3,GoToIf($[${AVAILSTATUS} = "1"]?4:5)
exten =>
2005 Jun 03
3
911 context, is this right?
I have 3 analog trunks zap/1, zap/4 and zap/5. zap/5 is the least used
line. Would the following work for 911 calls?
[e911]
exten => 911,1,ChanIsAvail(Zap/1)
exten => 911,2,Dial(Zap/1/911)
exten => 911,3,Hangup()
exten => 911,102,ChanIsAvail(Zap/4)
exten => 911,103,Dial(Zap/4/911)
exten => 911,104,Hangup()
exten => 911,203,ChanIsAvail(Zap/5)
exten =>
2004 Aug 20
3
Strange problem with Dial
I'm trying to add an emergency dial to my context. However, when I try to
dial it, I get caught in an endless loop.
For debugging, I have pared out nearly all the control flow and just have
ChanIsAvail() and Dial() called. Using two different extensions to call teh
same number, I get two different actions by *.
Here is the vvverbose output:
-- Starting simple switch on
2005 Mar 23
4
Chanisavail and IAX2
Guys.
Anybody doing ChanisAvail on IAX2 channels?
Im trying to do this:
exten => s,7,ChanIsAvail(IAX2/anton:intrudercom@armando-gw)
But I get that the chan is unavailable eventhough I can make calls to that
channel. Is there any chatch?
The channels is defined as peer and Ialso tried doing a register on iax.conf
for that channel. Everything is registering ok and I CAN make the call.
Any
2003 Oct 13
1
AGI solution to Grandstream BT102 call waiting problem
I'm trying to fix a problem with the GrandStream Budgetone 102. I've been reading the source code, mailing lists and other resources. Here's the scenario and the approach I have been pursuing. I'm having some problems with the AGI calls and I hope someone can give me some clarification.
PSTN <---> T1,PRI * <---> Grandstream BT 102 (12)
2004 Apr 08
1
Two operators, 10 rollover lines, Cisco 7960G chanisavail problem
Here's my situation.
I have two receptionists that answer incoming lines. Each has a 7960G with
5 incoming lines each. I'm trying to set this up so each line on each phone
doesn't utilize call waiting. My problem seems to be that
ChanisAvail(Sip/cisco1&Sip/cisco2&Sip/cisco3&Sip/cisco4&Sip/cisco5) always
returns cisco1.
Here are the sip.conf entries: (mind you,
2006 Feb 14
4
ChanIsAvail
Hi,
So I've done my research on Chanisavail, read the wiki, checked the
archive but can't seem to find anything to suit my scenario. I've
played around with it a lot, but I'm still scratching my head on what
I need to do.
What I need is to be able to accept a call by SIP and ring all
telephones that are not in use (which just so happen to be on Zap
interfaces, but might be SIP
2007 Jun 25
1
Problems with ChanIsAvail always return status 0
Hi list:
I'm having the next problem, it appear that the application ChanIsAvail
is not working on Asterisk 1.4.5 always return me 0 in AVAILSTATUS.
I add my dialplan and the output to the cli.
THanks.
In the example i'm dialing from extension SIP/112
My DialPlan Secction:
[macro-callonlyiffree]
exten => s,1,ChanIsAvail(${ARG1}|s)
exten => s,n,NoOp(${AVAILCHAN})
exten
2011 Mar 15
4
[1.4] Asterisk doesn't hang up?
Hello
I'm trying to use ChanIsAvail() to check when the landline is back
to idle after a call, but for some reason, Asterisk doesn't detect
that the callee has hung up after listening to MoH for a few seconds:
========== extensions.conf
;Play MoH for a few seconds, hang up, and
;check ChanIsAvail() able to detect when line idle again
exten => 8888,1,Answer()
exten =>
2007 May 31
2
Net2Phone Multiple SIP Trunk Not Working
Hi All,
As Net2Phone don't permit more than one session per account, I configured
about 10 sip trunks and configure multiple trunk routing but once the first
trunk is used I cannot make additional calls, I also cofigure my dial plan
in other way using the chanisavail command but still not working.
The chanisavail command configuration is correct as I can make calls using
other trunk than
2019 Jan 10
2
Hint and state
Hi,
on an Asterisk 16 with PJSIP I want to know the state of a device (idle,
busy, unavailable, ...) in the dialplan. I tried with ChanIsAvail() but
this one doesn't return the real state (eg a device calling an extension
which is running ChanIsAvail() is marked as idle!)
When I use in a console "core show hints" or "core show hint
<EXTENSION>" I get the right
2003 Nov 03
2
MWI - I know this has been discussed in depth already
Let post this question.. Because I must be real slow... The following is
my config on this...
group=1
context=default
signalling=fxs_ks
channel => 1
context=local
signalling=fxo_ks
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
echocancel=yes
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=1.5
txgain=1.5
immediate=no
musiconhold=default
usecallerid=yes