Displaying 20 results from an estimated 1000 matches similar to: "ChanIsAvail and SIP"
2004 Jan 16
11
Remote reload Cisco 7960
Does anyone have a working way of having a Cisco 7960 reload its config
remotely. I have tried some of the scripts that I have found on the web,
but to no avail. Thanks for the help.
B. J.
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2004 Dec 09
2
Multiple Instances of Asterisk
I have a quick question for the list. For what reason would you have
multiple instances of asterisk running on a single box? I can maybe see it
if you have multiple IP addresses, but other than that I am drawing a blank.
Thanks,
B. J.
2004 Jun 22
2
Multiple DTMF digits on 7960
Hello all. We have an asterisk system set up, and we are seeing a lot of
multiple DTMF digits being read by asterisk. In digging through the
archives the only answer I have seen is to put in the statement
relaxdtmf=yes in the zapata.conf file. Since we are not using any zapata
devices, I have tried to put that statement in my sip.conf file to no avail.
Any help would be appreciated as my end
2005 Jan 11
1
"o" extension broken?
Hello all. I just found out that I am no longer able to exit out of
voicemail properly by hitting the 0 key, but the * key works. Asterisk
comes back and says "I'm sorry, I did not understand that response" and goes
on in the context. Is this a new "feature" or bug? Is anyone else having
this problem? I am using Asterisk 1.0.3, and have tried it on two separate
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
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/${EXTEN})
exten
2009 Nov 03
3
Problem with ChanIsAvail
Hi all,
I am having a problem with ChanIsAvail. It always returns the same
result, regardless of whether an extension is available or not.
It always returns 0 Unknown Status.
This is my dialplan.
exten => _2XX,1,ChanIsAvail(SIP/winsor_${EXTEN}|s)
exten => _2XX,2,Verbose(0, ${AVAILSTATUS})
exten => _2XX,3,GoToIf($[${AVAILSTATUS} = "1"]?4:5)
exten =>
2005 Mar 01
5
Polycom Auto-Answer
I am having a problem with Polycom auto-answer. I have the auto-answer
working between PhoneA and PhoneB, but when I try to use the intercom
between more then one phone I start having problems. PhoneA dials *3
which calls PhoneB, PhoneC, and PhoneD. All the phones ring, but only
one will pick up, the rest will hang up and I get this error on
Asterisk: Got SIP response 500 "Internal Server
2012 Dec 20
7
asterisk 11 and DAHDI/i4
In 1.4.43 I would see things from "core show channels" like
DAHDI/18/xxxxx
for line 18
in Asterisk 11 its
DAHDI/i4/xxxx
How do I get the line number back?
Jerry
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
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,
2010 Aug 05
1
Can ChanIsAvail return status from sip uri using router ip
hello,
Although my previous posts in this forum have not received satisfying
answers, here is another question from me.
my question is can i use ChanIsAvail function to get the status of a user in
the format SPI/user-id if i provide user in sip uri like this
ChanIsAvail(SIP/user at 153.18.x.x:5062)
calling user with this sip uri works fine.
I once tried but status returned was "unknow
2004 May 31
1
Failover: iconnecthere to voicepulse
Hi all,
I'm working on a setup for a small office. I'd like to use SIP/iconnecthere
most of the time, because they're cheap. But they only allow a single call.
When the single iconnecthere line is in use, I'd like to use IAX2/voicepulse
instead:
exten => _1NXXNXXXXXX,1,Dial,SIP/BYEXTENSION@iconnect
exten => _1NXXNXXXXXX,2,Dial,IAX2/npI42VoD38@voicepulse/${EXTEN}
Well,
2005 Jan 18
4
TE110P as E1
Hello,
I'm having problem with a wildcard TE110P. As soon as I load
the module (wcte11xp for kernel 2.6.10), it spawns a yellow
error with or without an E1 plugged-in.
Any one managed to set it up in France?
Here are my files:
zaptel.conf:
span=1,0,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31
zapata.conf:
[channels]
language=fr
context=default
switchtype=euroisdn
pridialplan=unknown
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
2005 Sep 12
1
Is "ChanIsAvail" thread safe?
Curious whether the ChanIsAvail command is thread safe. By that I mean, if I
use ChanIsAvail to determine which channel to use, can I be sure that it
will still be available when I go to Dial it on the next line? It occurs to
me that there's a possibility the channel could get used by a competing
thread AFTER my thread has determined it is available and BEFORE my thread
gets a chance to
2005 Jan 27
1
ChanIsAvail not working
I'm testing ChanIsAvail context and it is not working for me.
exten => 55,1,ChanIsAvail(SIP/11&SIP/21)
exten => 55,2,Cut(theChannel=AVAILCHAN,,1)
exten => 55,3,Dial(${theChannel},r)
exten => 55,4,Hangup
exten => 55,102,Goto(s,4)
It is not dialing SIP/21 when I'm talking on SIP/11, it execute
Hangup instruction instruction.
According to notes:
The channels are checked
2005 Aug 28
2
Asterisk 1.2.0-beta1 tarball re-released
Due to a packaging error, the tarball released on Friday night did not
have a version number embedded in it, which results in various strange
build errors and other odd behavior.
The tarball on the FTP servers has been updated to correct this
situation. Sorry for the inconvenience :-)
2007 Jan 24
1
ChanIsAvail kills dialplan processing when no Zap available on 1.2.14.
Hi, I'm trying to use ChanIsAvail to build a resilient 'dialout' macro.
The logic is simple; try Zap/g1 (a group of two E1s), and if that
fails, try locating a channel via DUNDi. Here's a massively cut down version
to illustrate the problem I'm having.
macro dialout ( dest ) {
ChanIsAvail(Zap/g1);
noop(Value of AVAILCHAN is ${AVAILCHAN});
2003 Oct 05
1
ChanIsAvail app setting ${AVAILCHAN} to an unusable value.
I sent this earlier under "Editting variable contents" but no-one
has responded. So, the subject is now more to the problem, instead of
the solution I was trying to implement.
ChanIsAvail returns the channel ID plus "-<session>".
How can I edit ${AVAILCHAN} to remove this session ID, so I can use its
contents in a subsequent Dial statement?
Dialing on Zap just gives a