Error doesn't occur in 11.2.1
-- Executing [1260@default:1] Answer(SIP/sipuser-0001, ) in new
stack
-- Executing [1260@default:2] Goto(SIP/sipuser-0001,
scottsdale#queues-account,s,1) in new stack
-- Goto (scottsdale#queues-account,s,1)
-- Executing
Where possible you should have a VM to try these things as needed. Where
not, it isn't too difficult to duplicate the contexts and do something like
this
[default]
.
.
Exten = 1260,1,answer
Exten = 1260,n,goto(test-context,s,1)
.
From: asterisk-users-boun...@lists.digium.com
Simplest question first. Does it show up in core show applications or
core show application SetCallerPres? If not, do a make menuselect and see
if something broke in the ability to make the application.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
and Web call
center
And how would you have this working together with Asterisk queueing? I have
seen solutions like this using agent pauses and then making everyithing happen
outside the normal ACD flow, but it's a bit of a hack
l.
2013/1/22 Danny Nicholas da...@debsinc.com
This is how I would see the process working
1. use curl/wget to query Facebook (etc.)
2. determine whether we are to drop a call into the queue or just process a
message
3. determine agent availability through AMI process or asterisk -rx
process.
4. drop the call into the queue or place the
Not the greatest solution, but since you are most likely using a script for the
AMI process, you could do an
Asterisk –rx “core show channels verbose”|grep SIP/testmachine-000d
And get the dialed number from that.
Actually you could issue the AMI command core show channels verbose.
This might have changed but IIRC /etc/asterisk/manager.conf controls what
events you have access to.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Thursday, January 24, 2013 1:14 PM
To: Asterisk Users Mailing List -
Your sounds might be too loud. We use a lot of custom sounds here and when
the volume approaches 0 db (asterisk standard is -3 db) we get fuzz and
clicks.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
I would vote for system() on two accounts. #1 AGI requires more overhead
and protocol #2 you are not expecting a result to return to the dialplan.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Wednesday,
for your answer.
The most important here is that Asterisk continues with the rest of the
dialplan, in case the database-connection fails or hangs or ...
I don't think the System()-command makes this true.
Jonas.
On 01/23/2013 03:27 PM, Danny Nicholas wrote:
I would vote for system() on two
something
Exten = s,n,Dial(SIP/peer3,,10) ; dial peer 3
Exten = s,n,hangup()
The peer MUST be dialed even if the script Jonas.php is still running.
Jonas.
On 01/23/2013 03:44 PM, Danny Nicholas wrote:
Let's assume you're using this snippet
[default]
Exten = s,1,answer()
Exten = s,n
Originate is the answer here. Let’s say your X-lite is SIP/100 and you’re
dialing 555-1212. From the x-lite you dial 555-1212 and Asterisk does a dial
command to execute the call. From the web, we “originate” the call from
SIP/100 to 555-1212. Asterisk makes sure SIP/100 is available then
As I am going to mis-explain this, an Asterisk SIP call originates on port
5060 (incoming or outgoing) then uses two RTP ports for audio in and audio
out. Police and Hackers can tap into the RTP ports to monitor your
conversations (I don't really know if the capabilities stop there) but you
can
You are obviously getting the call connected, so the subnet issue is moot.
What this sounds like (pardon the pun) to me is an rtp skip issue. The
working calls are generating rtp connections in the allowed range; the
other calls have one or more ports outside of your rtp range. Verify that
all
-0006 answered Gtalk/+xx-2310
== Spawn extension (gtalk_incoming, r...@gmail.com, 4) exited
non-zero on 'Gtalk/+xx-2310'
On 1/22/13 11:21 AM, Danny Nicholas wrote:
You are obviously getting the call connected, so the subnet issue is moot.
What this sounds like (pardon
://wiki.asterisk.org/wiki/display/AST/Calling+using+Google
-Original Message-
From: Frank [mailto:fr...@efirehouse.com]
Sent: Tuesday, January 22, 2013 10:51 AM
To: Danny Nicholas
Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Google voice with no voice
If you needed a MITM, nothing would work now. The incoming call is
connecting, but no voice or no connection at all?
-Original Message-
From: Frank [mailto:fr...@efirehouse.com]
Sent: Tuesday, January 22, 2013 11:56 AM
To: Danny Nicholas
Subject: Re: [asterisk-users] Google voice
To: Danny Nicholas
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Google voice with no voice
Hi,
No, it's not even connecting.
On the caller side, I do not see anything showing that the called party
picks up.
On the D70 side, when I pick up, I have
What about jabber show channels?
-Original Message-
From: Frank [mailto:fr...@efirehouse.com]
Sent: Tuesday, January 22, 2013 1:12 PM
To: Danny Nicholas
Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Google voice with no voice
*CLI core show
This is incoming, outgoing or idle (no call)?
-Original Message-
From: Frank [mailto:fr...@efirehouse.com]
Sent: Tuesday, January 22, 2013 1:21 PM
To: Danny Nicholas
Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Google voice with no voice
This sounds like a codec issue. Set your verbose to 10 and retry the
incoming call.
-Original Message-
From: Frank [mailto:fr...@efirehouse.com]
Sent: Tuesday, January 22, 2013 1:26 PM
To: Danny Nicholas
Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re
Theoretically you can do this
Exten = 370,1,Voicemailmain(D70@default)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Frank
Sent: Tuesday, January 22, 2013 2:28 PM
To: Asterisk Users Mailing List -
That one's not in my wheelhouse since I only use Polycom phones. Google
D70 Provisioning
-Original Message-
From: Frank [mailto:fr...@efirehouse.com]
Sent: Tuesday, January 22, 2013 2:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Danny Nicholas
Subject: Re
Not doubting how quickly Nagios can respond, but if the Nagios solution is
going to place a call using Asterisk, wouldn’t it be just as efficient (or more
so) to depend on Asterisk?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
...@nyigc.com]
Sent: Tuesday, January 22, 2013 4:12 PM
To: ch...@acsdi.com; Asterisk Users Mailing List - Non-Commercial Discussion;
Danny Nicholas
Cc: Benny Amorsen
Subject: RE: [asterisk-users] Capture queue agent drop and put caller back in
queue
Using qualify=10 ?
-Original Message-
From
For just the messaging part, you should be able to use wget or curl to
interface and create messages. You might have to go a little higher level
like C or Perl, but it sounds very doable.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
The simplest way to do it would be to use sox to remix your moh file with
the message like this:
Let's say you're using the standard file macroform-cold_day.wav. First you
split it into two minute segments like so
Sox macroform-cold_day.wav seg1.wav trim 0.0 120.0
Sox macroform-cold_day.wav
Since Gosub is technically an application, you should be able to modify this
snippet in features.conf
testfeature = #9,peer,Playback,tt-monkeys ;Allow both the caller and
callee to play
;;tt-monkeys to the opposite
channel
To this
testfeature =
I think this is a phone problem not an asterisk one. In my experience a
SIP (IP) phone takes about 20 seconds to properly negotiate (re)registration
(longer for Polycom 501s). The best work-around I could recommend would be
to have an intermediate interface like kamailio (sp) that handles
As I read it you can do it like this:
From http://www.voip-info.org/wiki/view/Asterisk+sip+rtptimeout
Exten = s,1,noop (set rtptimeout so we can have 2 timeouts on a dial)
Exten = s,n,Set(rtptimeout=60)
Exten = s,n,Dial(SIP/peer1,60)
Exten = s,n,Dial(SIP/peer2,60)
Haven't tested this.
-Commercial Discussion
Subject: Re: [asterisk-users] Issue after upgrade to 1.8.20 - Unable to
connect to remote asterisk message on service asterisk start
On Wed, Jan 16, 2013 at 1:37 PM, Danny Nicholas da...@debsinc.com wrote:
Same issue exists with 11.2
I've created issue 20945 to track
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Yves A.
Sent: Tuesday, January 15, 2013 6:07 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] special conference room
Hi list,
I am in need of a
Same issue exists with 11.2
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bakko
Sent: Wednesday, January 16, 2013 1:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
n priority is a runtime value set by the dialplan. To use it in a
database, you would have to update the database using something like
dialplan show context.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roy
From what I read, neither confbridge or meetme have the whisper feature
built-in; This doesn't matter because the moderator would have to use
meetmeadmin or the confbridge equivalent to control the other functions.
The moderator would either need two phones or a phone and a web interface.
Let's
be a showstopper that i may have not
explained clear enough...:
muting and unmuting a caller should have the effect, that the caller can
talk
to the moderator or not... any caller should NEVER hear what other callers
are talking... may he be muted or not...
yves
Am 16.01.2013 23:01, schrieb Danny
I would suggest this
Exten = _0666XX,1,answer()
Exten = _0666XX,n,playback(tt-monkeys)
Exten = _0666XX,n,hangup()
You could just hangup on them, but playing the screeching monkeys will get
the message to them to leave you alone.
From: asterisk-users-boun...@lists.digium.com
of calls from this number _0666XX and i wants to block it
to call my number 520xx .
2013/1/14 Danny Nicholas da...@debsinc.com
Exten = _0666XX,1,answer()
Exten = _0666XX,n,playback(tt-monkeys)
Exten = _0666XX,n,hangup
Yes. This is referred to in the documentation as ex-girlfriend logic.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
isr...@gmail.com
Sent: Monday, January 14, 2013 10:57 AM
To: Asterisk Users Mailing List -
If you're worth the trouble to change my dialplan for, you should suffer
some torture for calling me.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Alvarez
Sent: Monday, January 14, 2013 11:06 AM
To:
AFAIK, the ${CHANNEL(language)} is what controls each. If you wanted to
answer the phone in English, then do voicemails in different languages, this
should work:
[default]
Exten = s,1,Answer()
Exten = s,n,Set(CHANNEL(language)=en) - redundant since default is en
Exten =
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Friday, January 11, 2013 4:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Which tool to edit custom reports from CDR and
queues
Discussion
Subject: Re: [asterisk-users] Set Language for VoiceMailMain
Thanks you for your answer.
There is no language-parameter that can define the language of mailbox and
VoiceMailMain ?
Jonas.
On 01/11/2013 03:33 PM, Danny Nicholas wrote:
AFAIK, the ${CHANNEL(language)} is what controls
/2013 04:07 PM, Danny Nicholas wrote:
No. It is purposely set from the dialplan. In Asterisk 11.X you have the
[zonemessage] section in voicemail.conf that could probably be tweaked to
change the language without dialplan changes. Also in sip.conf you can set
language by peer so you could have
Subject: Re: [asterisk-users] Set Language for VoiceMailMain
Well, I thought you had tried it and thus could tell it with 100% certainty.
Thanks for your help.
Jonas.
On 01/11/2013 04:16 PM, Danny Nicholas wrote:
Since the peer language sets CHANNEL(language), I can say yes with
reasonable
The general rule seems to be, don't restart it unless there's a problem or
you hear of memory leaks. I had a version of 1.4 that I restarted every
night because I read about memory leaks, but I hear of 1.2 installs that
have been running continuously for 10 years.
From:
I don't presently have 11 in production, but in each case where I've put 11
in on top of 10.X the process has been relatively seamless, so I expect my
10.X boxes will go to 11.X sometime this year.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Grant
Bagdasarian
Sent: Wednesday, January 09, 2013 4:25 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Streaming/Recording audio
Regarding the streaming of audio.
that it is correct for sqlite?
For what i know it should contain select or inser operator in dialplan.
Correct if i am wrong.
On Oct 19, 2012 9:18 AM, Danny Nicholas da...@debsinc.com wrote:
Just use the standard dialplan db commands - Read
Set(TESTOP=${DB(Nightop/ext)}) write Set(DB(Nightop/ext)=107
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of luke devon
Sent: Wednesday, January 09, 2013 9:06 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] IVR platform for a mobile operator
Hi Friends ,
I want to setup a
On Jan 9, 2013 8:38 PM, Danny Nicholas da...@debsinc.com wrote:
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of luke devon
Sent: Wednesday, January 09, 2013 9:06 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] IVR
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Monday, January 07, 2013 6:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] IAX2 support of video
According to this:
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Grant
Bagdasarian
Sent: Tuesday, January 08, 2013 9:24 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Streaming/Recording audio
Hello Users,
I've been searching for
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
Sent: Monday, January 07, 2013 2:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Paging unit suggestions
We currently have an
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Monday, January 07, 2013 3:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] IAX2 support of video
Does
Whether you use .call file or AMI, you should still do the call/page using a
context and that context run the PHP script.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bilal ghayyad
Sent: Monday, January 07,
.
Thanks.
On 3 January 2013 17:21, Danny Nicholas da...@debsinc.com wrote:
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
Cunningham
Sent: Thursday, January 03, 2013 3:13 PM
To: Asterisk Users Mailing List - Non-Commercial
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Magnus
Löfqvist
Sent: Friday, January 04, 2013 4:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Calender and EWS with shared calenders
Hi
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leandro
Dardini
Sent: Thursday, January 03, 2013 2:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DAHDI: How to know since when it is used?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis
Sent: Thursday, January 03, 2013 11:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Moving User Agent To
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason Parker
Sent: Thursday, January 03, 2013 2:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Moving User Agent
Just for grins, run netstat -anp on the call using just Asterisk and then
again with OpenSIPS in the mix. It sounds like OpenSIPS or your RTPproxy is
block the audio channels.
--
_
-- Bandwidth and Colocation Provided by
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
Cunningham
Sent: Thursday, January 03, 2013 3:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] faxdetect on/off on the fly?
Hello,
We
Put the AGI call in a macro context and add M(macro) to your Dial string.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henrik
Westerberg
Sent: Wednesday, January 02, 2013 8:02 AM
To: asterisk-users@lists.digium.com
Subject:
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
Sent: Wednesday, January 02, 2013 7:09 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Users list email totals by year .
So
of it. Is there a way to do this or do I have to use G() and
connect the both ends to AGI separately and then bridging them before
recording the call?
Thanks for help.
Regards,
Henrik
Danny Nicholas
Sent: Wednesday, January 02, 2013 8:18 AM
Put the AGI call in a macro context and add M(macro
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: Wednesday, January 02, 2013 9:54 AM
To: ch...@acsdi.com; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users]
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard Kenner
Sent: Wednesday, January 02, 2013 10:00 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Top Posting
I'm the opposite. I'm
Grow up, follow the rules, have a good day.
JohnM
PS. Did not intend to imply that it was Steve that hijacked the thread, in
case anyone read my comment that way JohnM
Steve has waded through enough of these that he should be a hijacker.
--
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ron Wheeler
Sent: Wednesday, January 02, 2013 2:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk as answering
1.6.2 is a deader soldier than 1.4.X.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henrik
Westerberg
Sent: Wednesday, January 02, 2013 3:20 PM
To: asterisk-users@lists.digium.com
Subject: Re:
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Alvarez
Sent: Wednesday, January 02, 2013 4:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Auto ban IP addresses
On Wed, Jan 2,
***
***
Until Monopolysoft fixes Outlook, I think we should Middle Post - Happy
New Year to New Zealand!
***
***
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live
My assembler may be a little bit rusty, but wouldn't -1 against rule #5 =
+1 for rule #5?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every
Wouldn't the answer to that violate family forum rules (see Charlie Sheen
jokes)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Don Kelly
Sent: Monday, December 31, 2012 11:18 AM
To: 'Asterisk Users Mailing
My best guess is that you are creating the .call file with permissions that
don’t allow Asterisk to delete it when it is finished or retries have been
exhausted.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Necati Demir
Sent:
Shouldn't be difficult. You're just setting up the Cisco box as a SIP
gateway. Here's a link to get you started.
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_6_1/ccmcfg/b06gtw
ay.html
From: asterisk-users-boun...@lists.digium.com
If it is writing to /v/l/m, then it is coming from somewhere else. All
Asterisk messages go to /v/l/asterisk.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL
INDRODIYA
Sent: Thursday, December 27, 2012 3:32 AM
To: Asterisk
I would set up 5 shell files called pray1.sh, pray2.sh, etc and then set up
5 entries in /etc/crontab to run them at the specified time daily. The file
pray1.sh should look something like this:
#!/bin/sh
cp /pray1/*.call /tmp
mv /tmp/*.call /var/spool/asterisk/outgoing
the entry in
The simplest way to address this kind of change is to test it a week
(month) or so in advance on your test machine (we have VM's set up to mock
our live machines). A protection against last minute changes is to have
this kind of thing controlled by variables so you can possibly even avoid
I would say that the database method has the advantage over GotoIfTime in
that it should stay the same between releases. More headache on the front
end, but easier once it is up and running.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
#1 I assume you have spandsp installed
#2 I'm guessing you got some hints from this thread -
https://issues.asterisk.org/jira/browse/ASTERISK-18394 ?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New
Asterisk.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, December 27, 2012 4:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] $100 Bounty
It may depend on the asterisk version, but in theory you should just be able to
set callerid(num) and callerid(name) before doing the IAX2 dial. You should
always specify the Asterisk version you are using as features often change or
are available/not available between different releases.
The Asterisk 11 part is irrelevant. You need to use an AGI or local
call to use the ChanIsAvail function.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Thursday, December 20, 2012 11:03 AM
To: Asterisk Users
IMO the local channel call should be the lowest overhead option available.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Thursday, December 20, 2012 11:42 AM
To: Asterisk Users Mailing List - Non-Commercial
Just for grins, do you have a softphone like xlite that you can try the
outgoing call on? I think it's an outgoing issue, not a polycom one.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent:
In my experience, you should set up two identical queues and configurations.
With a little work, you should be able to let server 1 know the phone is in
use by server 2 and vice versa.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Please post the sip.conf entry with any confidential data xxx'ed out.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Thursday, December 13, 2012 2:09 PM
To: Asterisk Users Mailing List -
The two things I would try are changing type from friend to peer and
sendrpid from no to yes. The no matching peer usually means the device
username isn't matching the sip.conf username=.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
This animal might be like the OBI110 box where you set it up in users.conf
instead of sip.conf.
Something like this:
[5001]
transfer=yes
call-limit=5
registersip=no
host = 1.2.3.4
context=default
hasvoicemail=no
dtmfmode=inband
threewaycalling=no
hasdirectory=no
callwaiting=no
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Thursday, December 13, 2012 3:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Digital accoustics trying to register to
Could be, but I'd check the easier to fix polarity settings.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph
Sent: Tuesday, December 11, 2012 11:27 AM
To: asterisk-users@lists.digium.com
Subject:
On 12/11/12 11:30, Danny Nicholas wrote:
Could be, but I'd check the easier to fix polarity settings.
How do I do that?
Notice, that this channel hang-up/disconnect does not happen all the time,
only once a while could be once a day or once a week.
--
Joseph
You don't state version, but I'm pretty sure this animal doesn't exist.
What I did in 1.4 was to set a variable before the gosub so I could track
it. Something like this
Exten = s,n,Set(from=foo)
Exten = s,n,gosub(showfoo,s,1)
Exten = s,n,Set(from=bar)
Exten = s,n,gosub(showfoo,s,1)
[showfoo]
Put a GLOBAL in extensions.conf with the version and use GOTOIF to run
AGI/DEADAGI dependent on it.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Monday, December 10, 2012 10:01 AM
To:
Sounds like a registration timeout issue. What does the sip.conf entry look
like for these?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Monday, December 10, 2012 10:30 AM
To: Asterisk Users Mailing List -
Does each box show up in the others SIP SHOW PEERS?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ken D'Ambrosio
Sent: Monday, December 10, 2012 2:59 PM
To: asterisk-users@lists.digium.com
Subject:
Not sure about this since I use the 10/11 branches and not 1.8, but I think
you need to use the deprecated call-limit for BLF and the new busylimit for
the other features you need.
http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf
From: asterisk-users-boun...@lists.digium.com
What happens if you reinitialize a phone, then do the update? (I keep a
bottle of Ibuprophen on hand just for Polycom issues).
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson
Sent: Thursday, December
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