[asterisk-users] Variables losing their value????

[Sherwood McGowan]

Alrighty Gents, let's see if any of you have encountered this
one...Variables losing their value...I'm setting a variable with four
underscores (used to be two, had same issue) so it can be inherited by child
channels, and then the next line in the dialplan I use it but it appears to
be empty...I've googled and found nothing stating this kind of weirdness..

Asterisk 1.8.2.2 (upgrading to 1.8.2.3 shortly)

dialplan:

[menu.main]
exten => s,1,Set(recfile=${FILTER(0-9,${UNIQUEID})});
exten => s,n,Set(logfile=${recfile}) ;

The log output:
-- Executing [s...@menu.main:1] Set("SIP/-",
"recfile=12974953060") in new stack
-- Executing [s...@menu.main:2] Set("SIP/-", "logfile=") in
new stack

Anybody have thoughts?

Thanks,
S McGowan
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[asterisk-users] what are QUEUE_MAX_PENALTY and QUEUE_MIN_PENALTY veriables?

[shayne.al...@gmail.com]

Dears;

I am looking for a way to handle callers via queuerules, but am not able to
exactly understand the meaning and affect of this two variables on Queue
Application, and how it change the priority of a caller to be answered
sooner.

QUEUE_MAX_PENALTY
QUEUE_MIN_PENALTY

tnx
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Re: [asterisk-users] Asterisk 1.8.3 BLF stopped working

[Rob Hillis]

On 12/02/11 04:02, Bryant Zimmerman wrote:
I am running 1.8.3 and my BLF lights have stopped working. The hints 
appear to be intact when I use core show hints. But none of the phones 
are getting the BLF updates.  This has happend in the past and I have 
had to restart my server. What could be causing this to occur. It did 
not do this with the 1.6.x builds.


Is there a way to reload the hints or force a refresh without re-starting


Does a restart actually fix the problem?  If not, compare the hint 
context in "core show hints" and the "Subscr.Cont." line in "sip show 
peer xxx", where xxx is one of the extensions attempting to subscribe to 
hints.  Make sure the two match.  I've had this problem before, and that 
was the cause.


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Re: [asterisk-users] sangoma wanpipe install error

[Moises Silva]

As the error suggest, try checking /var/log/messages for possible hints on
what went wrong.

Make sure you configured the device with wancfg_dahdi script first.

Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6
Canada
t. 1 905 474 1990 x128 | e. m...@sangoma.com


On Fri, Feb 11, 2011 at 1:47 AM, Roi Stork  wrote:

> Trying to install wanpipe 3.5.18.
>
> No errors during compile. But when I reach the point where wanpipe and
> dahdi_cfg is started, I encountered an error.
>
> Starting WAN Router...
> Loading WAN drivers: wanpipe done.
> Starting up device: wanpipe1
>
>
>wanconfig: WAN device wanpipe1 driver load failed !!
> : ioctl(wanpipe1,ROUTER_SETUP) failed:
> :  22 - Invalid argument
>
>
>Wanpipe driver did not load properly
>Please check /var/log/wanrouter and
>/var/log/messages for errors
>
> Configuring interfaces: w1g1 w1g1: ERROR while getting interface flags: No
> such device
>
> done.
> /etc/wanpipe/scripts/start: 7: Syntax error: Bad for loop variable
>
> DAHDI_SPANCONFIG failed on span 1: No such device
> or address (6)
>
> Dont know why I still keep getting a 'No such device' error even if the
> device was detected (Sangoma a104de, setup asked to configure/skip the 4
> ports) before the error happened.
>
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Re: [asterisk-users] On-Hold Music

[Steve Edwards]

Un-top-posting, but continuing to abuse the hijacked thread...


> On Sat, 12 Feb 2011, ayodele abejide wrote:
>
> > I am having problems playing files with the playback command, also with
> > the Dial (A()) option this is the output from console:
> >
> > This is the dialplan:
> >
> > exten => 1003,n,Playback(home/abejide/Desktop/a.wav)


On Fri, 11 Feb 2011, Steve Edwards wrote:


> Don't specify the file type.


On Sat, 12 Feb 2011, ayodele abejide wrote:


I tried what you suggested and this is the console output:

[Feb 12 03:18:41] WARNING[2774]: file.c:650 ast_openstream_full: File 
/var/lib/asterisk/sounds/home/abejide/Desktop/a does not exist in any format


Does the following shell snippet yield any clues?

for F in /var/lib/asterisk/sounds/home/abejide/Desktop/a*
do
file $F
done

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Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] On-Hold Music

[ayodele abejide]

I tried what you suggested and this is the console output:
[Feb 12 03:18:41] WARNING[2774]: file.c:650 ast_openstream_full: File 
/var/lib/asterisk/sounds/home/abejide/Desktop/a does not exist in any 
format[Feb 12 03:18:41] WARNING[2774]: file.c:956 ast_streamfile: Unable to 
open /var/lib/asterisk/sounds/home/abejide/Desktop/a (format 0x2 (gsm)): No 
such file or directory[Feb 12 03:18:41] WARNING[2774]: app_playback.c:471 
playback_exec: ast_streamfile failed on SIP/1002-0004 for 
/var/lib/asterisk/sounds/home/abejide/Desktop/a

Best Regards,


ABEJIDE, Ayodele A. (CCNA)
+2348039269311

"Before long, paying for a phone call will be as alien as paying for email"




> Date: Fri, 11 Feb 2011 18:12:19 -0800
> From: asterisk@sedwards.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] On-Hold Music
> 
> On Sat, 12 Feb 2011, ayodele abejide wrote:
> 
> > I am having problems playing files with the playback command, also with 
> > the Dial (A()) option this is the output from console:
> > 
> > This is the dialplan:
> > 
> > exten => 1003,n,Playback(home/abejide/Desktop/a.wav)
> 
> Don't specify the file type. Asterisk will try to find a file encoded and 
> formatted to match the encoding of the channel. Failing a match, Asterisk 
> will try to find a file it can transcode to match the encoding of the 
> channel.
> 
> Since you specified a relative path ('does not start with a slash'), 
> Asterisk will prefix (by default) '/var/lib/asterisk/sounds/' to your 
> path yielding:
> 
>   /var/lib/asterisk/sounds/home/abejide/Desktop/a.*
> 
> is this what you want?
> 
> -- 
> Thanks in advance,
> -
> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
> Newline  Fax: +1-760-731-3000
> 
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Re: [asterisk-users] On-Hold Music

[Steve Edwards]

On Sat, 12 Feb 2011, ayodele abejide wrote:


I am having problems playing files with the playback command...


And don't hijack other people's threads :)

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-
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Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] On-Hold Music

[Steve Edwards]

On Sat, 12 Feb 2011, ayodele abejide wrote:

I am having problems playing files with the playback command, also with 
the Dial (A()) option this is the output from console:


This is the dialplan:

exten => 1003,n,Playback(home/abejide/Desktop/a.wav)


Don't specify the file type. Asterisk will try to find a file encoded and 
formatted to match the encoding of the channel. Failing a match, Asterisk 
will try to find a file it can transcode to match the encoding of the 
channel.


Since you specified a relative path ('does not start with a slash'), 
Asterisk will prefix (by default) '/var/lib/asterisk/sounds/' to your 
path yielding:


/var/lib/asterisk/sounds/home/abejide/Desktop/a.*

is this what you want?

--
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-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] On-Hold Music

[ayodele abejide]

I am having problems playing files with the playback command, also with the 
Dial (A()) option this is the output from console:
[Feb 12 01:58:41] WARNING[1569]: file.c:650 ast_openstream_full: File 
home/abejide/Desktop/a.wav does not exist in any format[Feb 12 01:58:41] 
WARNING[1569]: file.c:956 ast_streamfile: Unable to open 
home/abejide/Desktop/a.wav (format 0x2 (gsm)): No such file or directory[Feb 12 
01:58:41] WARNING[1569]: app_playback.c:471 playback_exec: ast_streamfile 
failed on SIP/1002-0002 for home/abejide/Desktop/a.wav
This is the dialplan:

[general]
[default]exten => 1000,1,Dial(sip/1000,30,A(home/abejide/Music/moh1.wav))exten 
=> 1001,1,Dial(sip/1001,30)exten => 1003,1,Answer()exten => 
1003,n,Playback(home/abejide/Desktop/a.wav)exten => 1003,n,Hangup()exten => 
1002,1,Dial(sip/1002)exten => 101,1,AGI(agi://127.0.0.1/adhearsion)exten => 
1,1,Dial(sip/1000,10)exten => 1,n,Playback(vm-nobodyavail)exten => 
1,n,Hangup()exten => 2,1,Answerexten => 2,n,Record(/tmp/a.gsm,3,30)exten => 
2,n,Wait(2)exten => 2,n,Playback(/tmp/a.gsm)exten => 2,n,Hangup()

Best Regards,


ABEJIDE, Ayodele A. (CCNA)
+2348039269311

"Before long, paying for a phone call will be as alien as paying for email"



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Re: [asterisk-users] On-Hold Music

[Steve Howes]

On 11 Feb 2011, at 22:37, Danny Nicholas wrote:
>   In 500 words or less (if possible), please explain what is a legal 
> music-on-hold file?

Depends on the country, and what licence you posses. Googling ' 
hold music regulations' may help.

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Re: [asterisk-users] Gtalk/Jabber Issue

[Jian Gao]

Port 5222 opened?

Jian

On 11-02-10 09:54 PM, William Stillwell wrote:


I was getting unable to make channel..

So, this is what I am doing..

Service stop asterisk

Purge modules

Make clean

Remove all traces of iskemel

Recompile that. With  , add needed entrée into ldconfig.

Verify iksemel loaded via ldconfig –p | grep semel.

Change to /asterisk source location

Make clean

Now ./configure asterisk

Make menuselect, make sure chan_gtalk, and res_jabber as selected.

Make

Make install

Start asterisk..

Trying inbound..

Same thing, jabber call comes in, doesn’t fire the gtalk extension..

Outbound call , I get:

[Feb 11 00:52:18] ERROR[440]: chan_gtalk.c:934 gtalk_alloc: no gtalk 
capable clients to talk to.


[Feb 11 00:52:18] WARNING[440]: app_dial.c:1759 dial_exec_full: Unable 
to create channel of type 'Gtalk' (cause 0 - Unknown)


??

jabber/jingle/gtalk cmd all exist, and modules loaded.

*From:*Vladimir Mikhelson [mailto:v...@mikhelson.com]
*Sent:* Friday, February 11, 2011 12:40 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Cc:* William Stillwell
*Subject:* Re: [asterisk-users] Gtalk/Jabber Issue

William,

Have you tried outgoing calls?  What happens there?

Have you restarted the Asterisk after you fixed the typo?

-Vladimir



On 2/10/2011 10:44 PM, William Stillwell wrote:

Yeah, that was a typo, but I fixed, still no dice.

The incoming jabber call doesn’t fire the gtalk connection.

*From:*asterisk-users-boun...@lists.digium.com 
 
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Warren 
Selby

*Sent:* Thursday, February 10, 2011 10:16 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Gtalk/Jabber Issue

You've got connection=jp_jabber defined in one file, and [jb_jabber] 
defined in the other.


Thanks,

--Warren Selby, dCAP


On Feb 10, 2011, at 5:55 PM, "William Stillwell" 
mailto:will...@stillwellsoft.com>> wrote:


Sorry, Asterisk Build 1.6.2.7

*From:*asterisk-users-boun...@lists.digium.com

[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of
*William Stillwell
*Sent:* Thursday, February 10, 2011 6:50 PM
*To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
*Subject:* [asterisk-users] Gtalk/Jabber Issue

OK, im pulling my hair out, everything looks configured right,
deleted, and started over, etc, etc. but can’t seem to get this to
work

Gtalk.conf

[general]

context=google-in

allowguest=yes

bindaddr=192.168.xxx.xxx

extenip=96.254.xxx.xxx

[guest]

context=google-in

disallow=all

allow=ulaw

allow=g729

connection=jp_jabber

jabber.conf

[general]

debug=yes

;autoprune=no

autoregister=yes

[jb_jabber]

type=client

serverhost=talk.google.com

username=xx...@gmail.com
/Talk

secret=XXX

port=5222

usetls=yes

usesasl=yes

;status=Available

statusmessage="Connected via Asterisk"

;timeout=100

;keepalive=yes

Extensions.conf

[google-in]

exten => s,1,NoOp(Call from GTalk)

exten => s,n,Set(CallerID(Name)="From GoogleTalk")

exten => s,n,Dial(SIP/1000)

jabber show connected

Jabber Users and their status:

   User: xxx...@gmail.com /Talk -
Connected



   Number of users: 1

 CLI on incoming Call 

bannana*CLI>

JABBER: jb_jabber INCOMING: mailto:+1*@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2>"
to="**@gmail.com/TalkD876FAA0
"
id="jingle:10.218.14.137-17447266:1:03800E94"
type="set">mailto:SIP1007753261@10.218.122.83>"
initiator="+1***@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2
"
xmlns:ses="http://www.google.com/session";>http://www.google.com/session/phone";>http://www.google.com/transport/raw-udp"/>http://www.google.com/transport/p2p"/>

bannana*CLI>

JABBER: jb_jabber INCOMING: mailto:+1@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2>"
to="**@gmail.com/TalkD876FAA0
"
id="jingle:10.218.14.137-17447266:1:03800EB9"
type="set">mailto:SIP1007753261@10.218.122.83>"
initiator="+1***@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2
"
xmlns:ses="http://www.google.com/session";>http://www.google.com/session/phone";>Call
cancelled

bannana*CLI>

it doesn’t even try to fire the google-in context ?

Lastest Version of iksemel Installed, asterisk was rebuild after
installed, asterisk sees both jabber/gtalk commands.

It just will NOT ring my dia

[asterisk-users] On-Hold Music

[Danny Nicholas]

Hi gang,

In 500 words or less (if possible), please explain what is a
legal music-on-hold file?  My boss hates the stuff provided with the
distribution and I figure that I'm asking for trouble if I take my Les Mis
tracks and run them through Audacity and SOX to make new files.

 

Thanks in advance

Danny Nicholas

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Re: [asterisk-users] Asterisk compile option DAHDI SPANS

[Paul Belanger]

On 11-02-11 04:15 PM, satish patel wrote:
> That means 1 card 1 span 2 card 2 span 
> 
Not really, for example 1 card can have 4 spans (Digium TDM410P). You
can usually substitute the word 'port' for 'span'.  EG: My card has 2 T1
ports -> My card has 2 T1 spans.

-- 
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Check us out at: http://digium.com & http://asterisk.org

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Re: [asterisk-users] Asterisk compile option DAHDI SPANS

[satish patel]

Thank you so much

That means 1 card 1 span 2 card 2 span 

-S

> Date: Fri, 11 Feb 2011 16:04:19 -0500
> From: pabelan...@digium.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Asterisk compile option DAHDI SPANS
> 
> On 11-02-11 03:57 PM, satish patel wrote:
> > what does the Compiler Option mean "LOTS_OF_SPANS"  ?
> > The description is: "More than 32 DAHDI spans" 
> > Does this mean, more than 32 DAHDI Channels ?
> > 
> > I have TWO T1 line so do i need to select this option ?
> > 
> No, your card would have 2 spans.
> 
> EG:
> 1 span = (24 channels)
> 
> -- 
> Paul Belanger
> Digium, Inc. | Software Developer
> twitter: pabelanger | IRC: pabelanger (Freenode)
> Check us out at: http://digium.com & http://asterisk.org
> 
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Re: [asterisk-users] Asterisk compile option DAHDI SPANS

[Paul Belanger]

On 11-02-11 03:57 PM, satish patel wrote:
> what does the Compiler Option mean "LOTS_OF_SPANS"  ?
> The description is: "More than 32 DAHDI spans" 
> Does this mean, more than 32 DAHDI Channels ?
> 
> I have TWO T1 line so do i need to select this option ?
> 
No, your card would have 2 spans.

EG:
1 span = (24 channels)

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[asterisk-users] Asterisk compile option DAHDI SPANS

[satish patel]

Hi All

what does the Compiler Option mean "LOTS_OF_SPANS"  ?
The description is: "More than 32 DAHDI spans" 
Does this mean, more than 32 DAHDI Channels ?

I have TWO T1 line so do i need to select this option ?

-S

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Re: [asterisk-users] dialplan announcements

[Steve Edwards]

On Fri, 11 Feb 2011, ERIC HERRON wrote:

I want to have an option off the IVR that plays back the announcement 
for the day. At the end of the message, I want the caller to get kicked 
back to the previous menu.


The conditions are that I want the recorder to dial a feature code that 
prompts him to record the message. He then presses 1 to accept. This 
gets saved as announcement.wav.


The record() and background()/playback() applications should be 
appropriate based on the level of detail you have supplied.


Show us what you have so far.

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Re: [asterisk-users] dialplan announcements

[Doug Lytle]

ERIC HERRON wrote:
The conditions are that I want the recorder to dial a feature code 
that prompts him to record the message. He then presses 1 to accept. 
This gets saved as announcement.wav.


You'll need to supply some of your own audio prompts, I recorded my own, 
here is the code I have:


[recordings]

;
;* Voice Prompt recording menu
;

exten => 50*,1,Set(COUNT=0)
exten => 50*,n,Playback(local/extension-recording-menu)
exten => 50*,n,Playback(local/please-record-msg)
exten => 50*,n,Record(mymessage:gsm|5|60)
exten => 50*,n,Playback(local/you-said)
exten => 50*,n,Playback(mymessage)

;**
;* Press 1 to continue or
;* 2 to change your message
;**

exten => 50*,n,Background(local/press1-or-2)
exten => 50*,n,Set(TIMEOUT(digit)=2)
exten => 50*,n,WaitExten(5)

exten => 1,1,System(/bin/mv /var/lib/asterisk/sounds/mymessage.gsm 
/var/lib/asterisk/sounds/local/`date +%s`.gsm)

exten => 1,2,Playback(local/recording-saved)
exten => 1,3,Background(local/press3-torecord-4tohang)

exten => 2,1,Goto(recordings,50*,4)
exten => 3,1,Goto(recordings,50*,4)

exten => 4,1,Playback(vm-goodbye)
exten => 4,2,Hangup()

exten => t,1,GotoIf($[ ${COUNT} > 2 ]?4,1)
exten => t,n,Playback(local/sorry-didnot-getthat)
exten => t,n,Set(COUNT=$[${COUNT} + 1])
exten => t,n,Goto(recordings,50*,8)

exten => i,1,Playback(local/sorry-invalid-choice)
exten => i,2,Goto(recordings,50*,4)

Doug


--

Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary Safety, 
deserve neither Liberty nor Safety."


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Re: [asterisk-users] Asterisk 1.8.3

[satish patel]

Here is the patch did you apply it ? 

https://issues.asterisk.org/file_download.php?file_id=28206&type=bug

> Date: Fri, 11 Feb 2011 08:46:36 -0200
> From: vinic...@canall.com.br
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Asterisk 1.8.3
> 
> That makes two of us. I tried asking on asterisk-dev but had no reply.
> 
> 
> 
> - Mensagem original - 
> 
> 
> Hi 
> 
> Does anyone have any rough idea how far away 1.8.3 is? 
> 
> We can't deploy 1.8 yet because of this issue 
> 
> https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=18403 
> -- 
> Ishfaq Malik 
> Software Developer 
> PackNet Ltd 
> 
> Office: 0161 660 3062 
> 
> 
> -- 
> _ 
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> 
> 
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Re: [asterisk-users] Asterisk 1.8.3

[satish patel]

I have asterisk 1.8.2 in development and i can blind transfer from A to C 
without any issue.  Or may be i am doing wrong thing?

How do i reproduce this error ? 

-S



> Date: Fri, 11 Feb 2011 11:41:37 -0500
> From: pabelan...@digium.com
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Asterisk 1.8.3
> 
> On 11-02-11 11:14 AM, Danny Nicholas wrote:
> > Isn't that a little "Bleeding edge"? Guess that's why
> > I'm still using the 1.4X set.
> >
> No?  OP asked for asterisk 1.8.3, currently asterisk-1.8.3-rc2 is the
> latest RC.  If no regressions are found with the new patches, it will
> become 1.8.3.
> 
> Basically, if you are waiting for asterisk-1.8.x to be released, start
> testing asterisk-1.8.x-rcx and report feedback.
> 
> -- 
> Paul Belanger
> Digium, Inc. | Software Developer
> twitter: pabelanger | IRC: pabelanger (Freenode)
> Check us out at: http://digium.com & http://asterisk.org
> 
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[asterisk-users] dialplan announcements

[ERIC HERRON]

Hey all,

 

I tried to do some searching but I found snippets and I am having trouble
putting it all together.

 

I want to have an option off the IVR that plays back the announcement for
the day. At the end of the message, I want the caller to get kicked back to
the previous menu.

 

The conditions are that I want the recorder to dial a feature code that
prompts him to record the message. He then presses 1 to accept. This gets
saved as announcement.wav.

 

Can anyone shed some shining light on this matter?

 

Thanks,

--Eric

 

 

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[asterisk-users] AstMail

[Flavio Miranda]

Hello everybody,
  Anybody here knows about Astmail ? I have set up in a server but something is 
going wrong!  i can open its web interface but when I put the extension number 
and its password I receive:Invalid mailbox or password   My asterisk is 1.6 and 
my S.O is debian lenny.
I know this is not asterisk but as the project has not a forum or mail list, I 
am trying help here!
Thanks in advanced!

Att,

 

Flavio Roberto Miranda

MSN:flaviormira...@hotmail.com
Skype: flaviormiranda

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Re: [asterisk-users] Asterisk 1.8.3 BLF stopped working

[Bryant Zimmerman]

I am running 1.8.3 and my BLF lights have stopped working. The hints appear 
to be intact when I use core show hints. But none of the phones are getting 
the BLF updates.  This has happend in the past and I have had to restart my 
server. What could be causing this to occur. It did not do this with the 
1.6.x builds.

Is there a way to reload the hints or force a refresh without re-starting

Thanks

Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003 

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Re: [asterisk-users] Gtalk/Jabber Issue

[Vladimir Mikhelson]

William,

As I understand you have to upgrade your GTalk to Google Voice to be
able to place and receive PSTN calls in Asterisk.

The easiest way is to subscribe to GMail using your GTalk credentials. 
As you are in GMail try calling PSTN from there.

Then go to Google Voice / Settings / Phones and make sure GTalk phone
instance is there.  Make sure GTalk lists your new gmail user name, not
the arbitrary one you may have used to create the GTalk account.

If it does not, go ahead and delete the GTalk phone instance in Phones
and log off.  Then go back to GMail and initiate the call application. 
Log off.  Go back to Google Voice and make sure the GTalk instance lists
your GMail e-mail address.

Next step.  Try to edit the GTalk phone instance in Google Voice
Settings and watch for any "pink" messages at the top of the screen when
you hit "Save."  Try to resolve the issues if any.

If you made any changes to your credentials in the process make sure to
log off from all Google applications and reset your Asterisk.  Also
reset Asterisk even if you did not change credentials but tweaked or
initialized the GTalk phone instance in the Google Voice Settings.

-Vladimir



>  
>
> I Double checked the chat settings.
>
>  
>
> For some reason jabber is not sending any outbound response packets at
> all.. not sure why. Will need to see if I can stuff some more debug
> code into res_jabber.c and figure out whats going on, debug seems limited.
>
>  
>
>  
>
>  
>
> *From:*asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of
> *Vladimir Mikhelson
> *Sent:* Friday, February 11, 2011 1:59 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Gtalk/Jabber Issue
>
>  
>
> William,
>
> Another thing.  Have you tried calling from GMail?  If not please make
> sure you can send/receive calls there.
>
> One more test.  Go to your GV Account Settings / Phones, Edit "Google
> Chat", "Save"   Watch for the pink error messages in the upper portion
> of the screen.
>
> -Vladimir
>
>
>
>
> On 2/11/2011 12:32 AM, William Stillwell wrote:
>
> Still no dice..
>
>  
>
> This make no since.. ive gone over the config a million times now..
>
>  
>
> The windows gtalk /voice client works just fine.  (incoming and
> outgoing calls)
>
>  
>
>  
>
>  
>
> *From:*asterisk-users-boun...@lists.digium.com
> 
> [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of
> *Vladimir Mikhelson
> *Sent:* Friday, February 11, 2011 12:51 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Gtalk/Jabber Issue
>
>  
>
> William,
>
> I have just noticed that you have several configuration statements
> commented out.
>
> I would suggest to un-comment the "status=" in jabber.conf.  I would
> also suggest to un-comment the "timeout=", I am not that concerned of
> the "keepalive=".
>
> You can reload jabber, no need to restart the Asterisk.
>
> -Vladimir
>
>
>
> On 2/10/2011 11:40 PM, Vladimir Mikhelson wrote:
>
> William,
>
> Have you tried outgoing calls?  What happens there?
>
> Have you restarted the Asterisk after you fixed the typo?
>
> -Vladimir
>
>
>
> On 2/10/2011 10:44 PM, William Stillwell wrote:
>
> Yeah, that was a typo, but I fixed, still no dice.
>
>  
>
> The incoming jabber call doesn’t fire the gtalk connection.
>
>  
>
>  
>
> *From:*asterisk-users-boun...@lists.digium.com
> 
> [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Warren
> Selby
> *Sent:* Thursday, February 10, 2011 10:16 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Gtalk/Jabber Issue
>
>  
>
> You've got connection=jp_jabber defined in one file, and [jb_jabber]
> defined in the other. 
>
> Thanks,
>
> --Warren Selby, dCAP
>
>
> On Feb 10, 2011, at 5:55 PM, "William Stillwell"
> mailto:will...@stillwellsoft.com>> wrote:
>
> Sorry, Asterisk Build 1.6.2.7
>
>  
>
> *From:*asterisk-users-boun...@lists.digium.com
> 
> [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of
> *William Stillwell
> *Sent:* Thursday, February 10, 2011 6:50 PM
> *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
> *Subject:* [asterisk-users] Gtalk/Jabber Issue
>
>  
>
> OK, im pulling my hair out, everything looks configured right,
> deleted, and started over, etc, etc. but can’t seem to get this to
> work
>
>  
>
>  
>
> Gtalk.conf
>
>  
>
> [general]
>
> context=google-in
>
> allowguest=yes
>
> bindaddr=192.168.xxx.xxx
>
> extenip=96.254.xxx.xxx
>
>  
>
> [guest]
>
> context=google-in
>
> disallow=all
>
> allow=ulaw
>
> allow=g729
>
> connection=jp_jabber
>
>  
>
> jabber.conf
>
>  
>
> [general]

Re: [asterisk-users] Asterisk 1.8.3

[Paul Belanger]

On 11-02-11 11:14 AM, Danny Nicholas wrote:
> Isn't that a little "Bleeding edge"? Guess that's why
> I'm still using the 1.4X set.
>
No?  OP asked for asterisk 1.8.3, currently asterisk-1.8.3-rc2 is the
latest RC.  If no regressions are found with the new patches, it will
become 1.8.3.

Basically, if you are waiting for asterisk-1.8.x to be released, start
testing asterisk-1.8.x-rcx and report feedback.

-- 
Paul Belanger
Digium, Inc. | Software Developer
twitter: pabelanger | IRC: pabelanger (Freenode)
Check us out at: http://digium.com & http://asterisk.org

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Re: [asterisk-users] Asterisk 1.8.3

[Jonathan Thurman]

On Fri, Feb 11, 2011 at 7:59 AM, satish patel  wrote:
>
> I thought it has been resolved in 1.8.2 version

Issue 18403 was not resolved in 1.8.2, but in 1.8.3-rc1.  Release
1.8.3-rc2 was cut on 1/20/2011, so hopefully the full release will be
out soon.

You can see where the issue was merged here:
  http://svn.asterisk.org/svn/asterisk/tags/1.8.3-rc1/ChangeLog

-Jonathan

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Re: [asterisk-users] Asterisk 1.8.3

[Danny Nicholas]

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger
Sent: Friday, February 11, 2011 10:11 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 1.8.3

On 11-02-11 04:37 AM, Ishfaq Malik wrote:
> Does anyone have any rough idea how far away 1.8.3 is?
> 
If you are hard up for a release, you can use the latest 1.8.3 RC[1].

[1]
http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.8.3-rc2.tar.
gz

-- 
Paul Belanger

Isn't that a little "Bleeding edge"?  Guess that's why I'm still using the
1.4X set.


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Re: [asterisk-users] Asterisk 1.8.3

[Paul Belanger]

On 11-02-11 04:37 AM, Ishfaq Malik wrote:
> Does anyone have any rough idea how far away 1.8.3 is?
> 
If you are hard up for a release, you can use the latest 1.8.3 RC[1].

[1]
http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.8.3-rc2.tar.gz

-- 
Paul Belanger
Digium, Inc. | Software Developer
twitter: pabelanger | IRC: pabelanger (Freenode)
Check us out at: http://digium.com & http://asterisk.org

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Re: [asterisk-users] Asterisk 1.8.3

[Ishfaq Malik]

So did I but I'm having the same problem in Asterisk 1.8.2.2 and the
asterisk issue number I pasted the link of has Target Version 1.8.3

On Fri, 2011-02-11 at 15:59 +, satish patel wrote:
> 
> I thought it has been resolved in 1.8.2 version 
> 
> 
> Thanks,
> Satish 
> 
> > Date: Fri, 11 Feb 2011 08:46:36 -0200
> > From: vinic...@canall.com.br
> > To: asterisk-users@lists.digium.com
> > Subject: Re: [asterisk-users] Asterisk 1.8.3
> > 
> > That makes two of us. I tried asking on asterisk-dev but had no
> reply.
> > 
> > 
> > 
> > - Mensagem original - 
> > 
> > 
> > Hi 
> > 
> > Does anyone have any rough idea how far away 1.8.3 is? 
> > 
> > We can't deploy 1.8 yet because of this issue 
> > 
> > https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=18403 

-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062


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Re: [asterisk-users] Asterisk 1.8.3

[satish patel]

I thought it has been resolved in 1.8.2 version 


Thanks,
Satish 

> Date: Fri, 11 Feb 2011 08:46:36 -0200
> From: vinic...@canall.com.br
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Asterisk 1.8.3
> 
> That makes two of us. I tried asking on asterisk-dev but had no reply.
> 
> 
> 
> - Mensagem original - 
> 
> 
> Hi 
> 
> Does anyone have any rough idea how far away 1.8.3 is? 
> 
> We can't deploy 1.8 yet because of this issue 
> 
> https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=18403 
> -- 
> Ishfaq Malik 
> Software Developer 
> PackNet Ltd 
> 
> Office: 0161 660 3062 
> 
> 
> -- 
> _ 
> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 
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> http://www.asterisk.org/hello 
> 
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> To UNSUBSCRIBE or update options visit: 
> http://lists.digium.com/mailman/listinfo/asterisk-users 
> 
> 
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Re: [asterisk-users] Realtime queues not playing prompts

[Jonas Kellens]

Hello,

only the field "periodic_announce" is played, but not according to the 
value in "periodic_announce_frequency".


If I set "periodic_announce_frequency" to 10, than still this announce 
is played every 25 seconds.


Where does this 25 seconds come from ?
Why is "queue_thankyou" played, even if the field is set to NULL ?
Why is "queue_thereare" and "queue_callswaiting" not playing ??


Kind regards,
Jonas.


On 02/11/2011 12:48 PM, Jonas Kellens wrote:

Hello list,

I'm using realtime queues and noticing that prompts are not played as 
expected.


Database :

announce =
queue_youarenext = queue_youarenext
queue_thereare = queue_thereare
queue_callswaiting = queue_callswaiting
queue_holdtime =
queue_thankyou =
queue_reporthold =
announce_frequency= 10
announce_holdtime =
announce-position = yes
periodic_announce =
periodic_announce_frequency =




The only thing that happens is this :

[Feb 11 12:40:01] -- Started music on hold, class 'default', on 
SIP/testcorp6-
[Feb 11 12:40:28] -- Told SIP/testcorp6- in voipq3 their 
queue position (which was 1)
[Feb 11 12:40:28] --  Playing 
'queue-thankyou.alaw' (language 'nl')
[Feb 11 12:40:29] -- Started music on hold, class 'default', on 
SIP/testcorp6-



But I do not hear the position announcement, only the prompt "thank 
you" which I don't want by the way.



Am I missing something in my configuration here ? Prompts like 
"queue_thereare" are not played, but prompts like "queue_thankyou" ARE 
played (which I don't want).

Why ?


Kind regards,

JOnas.


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Re: [asterisk-users] meetme conference & playback of random sound file

[John Kiniston]

Check out the Random Application and the RAND function, Here is a
quick untested example for either.

exten => s,1,Answer
exten => s,n,Background(privacy-please-stay-on-line-to-be-connected)
exten => s,n,Random(33:${CONTEXT},s,FILE1)  ; 33% Num1
exten => s,n,Random(33:${CONTEXT},s,FILE2)  ; 33% Num2
exten => s,n,Random(34:${CONTEXT},s,FILE3)  ; 34% Num3
exten => s,n(FILE1),Background(tt-monkeys)
exten => s,n,Goto(Connect)
exten => s,n(FILE2),Background(tt-weasels)
exten => s,n,Goto(Connect)
exten => s,n(FILE3),Background(gambling-drunk)
exten => s,n,Goto(Connect)
exten => s,n(CONNECT),NoOp
exten => s,n,Meetme(options)


Or using RAND if your prompts are all numbered as prompt0 to prompt100

exten => s,1,Answer
exten => s,n,Background(privacy-please-stay-on-line-to-be-connected)
exten => s,n,Set(promptnum=${RAND(1,100)})
exten => s,n,Background(prompt${promptnum})
exten => s,n,Meetme(options)

On Thu, Feb 10, 2011 at 5:58 PM, John Jolly  wrote:
>
> i am trying to configure the meetme conference (asterisk 1.8) to play a
> random sound file from a specific directory prior to it dropping the caller
> into the conference itself. i am able to successfully get it to play a
> specific file prior to entering the conference unsure how to implement this
> sort of randomization.
>
> Is this possible? Any help will be greatly appreciated.
>
> john jolly jgjolly[at]gmail.com
>

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Re: [asterisk-users] sangoma wanpipe install error

[Gopalakrishnan A.N]

Try to check your kernel version as per the wiki.sangoma.com website. I hope
that will solve your issue.

On Fri, Feb 11, 2011 at 12:17 PM, Roi Stork  wrote:

> Trying to install wanpipe 3.5.18.
>
> No errors during compile. But when I reach the point where wanpipe and
> dahdi_cfg is started, I encountered an error.
>
> Starting WAN Router...
> Loading WAN drivers: wanpipe done.
> Starting up device: wanpipe1
>
>
>wanconfig: WAN device wanpipe1 driver load failed !!
> : ioctl(wanpipe1,ROUTER_SETUP) failed:
> :  22 - Invalid argument
>
>
>Wanpipe driver did not load properly
>Please check /var/log/wanrouter and
>/var/log/messages for errors
>
> Configuring interfaces: w1g1 w1g1: ERROR while getting interface flags: No
> such device
>
> done.
> /etc/wanpipe/scripts/start: 7: Syntax error: Bad for loop variable
>
> DAHDI_SPANCONFIG failed on span 1: No such device
> or address (6)
>
> Dont know why I still keep getting a 'No such device' error even if the
> device was detected (Sangoma a104de, setup asked to configure/skip the 4
> ports) before the error happened.
>
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Gopalakrishnan A.N.
VoIP call - sip:sai...@gtalk2voip.com
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[asterisk-users] digium te220

[Albert]

hi,

can anymore drop me a asterisl's config for digium te220b (with ec) or
at least some good tutorial of configuratin e1 line with that card ?

thanks in advance,
robert

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Re: [asterisk-users] Early audio SIP sequence order question

[Kevin P. Fleming]

On 02/11/2011 02:11 AM, Benoit Panizzon wrote:

Hi Kevin

I just found something interresting:

http://www.faqs.org/rfc/rfc3960.txt

   1. Unless a 180 (Ringing) response is received, never generate
  local ringing.

   2. If a 180 (Ringing) has been received but there are no incoming
  media packets, generate local ringing.

   3. If a 180 (Ringing) has been received and there are incoming
  media packets, play them and do not generate local ringing.

So, yes, this most probably is an asterisk bug, if it's not a config issue I
haven't figured out yet. Shall I submit a bug report?


It is possible that the 'progressinband' configuration option is related 
to this, but I'm not sure that it is (since I believe it only reacts to 
183 messages, not 180). Given that, it would be best to open a bug 
report to get it in the queue for someone to look at.


--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us out at www.digium.com & www.asterisk.org

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[asterisk-users] Realtime queues not playing prompts

[Jonas Kellens]

Hello list,

I'm using realtime queues and noticing that prompts are not played as 
expected.


Database :

announce =
queue_youarenext = queue_youarenext
queue_thereare = queue_thereare
queue_callswaiting = queue_callswaiting
queue_holdtime =
queue_thankyou =
queue_reporthold =
announce_frequency= 10
announce_holdtime =
announce-position = yes
periodic_announce =
periodic_announce_frequency =




The only thing that happens is this :

[Feb 11 12:40:01] -- Started music on hold, class 'default', on 
SIP/testcorp6-
[Feb 11 12:40:28] -- Told SIP/testcorp6- in voipq3 their 
queue position (which was 1)
[Feb 11 12:40:28] --  Playing 
'queue-thankyou.alaw' (language 'nl')
[Feb 11 12:40:29] -- Started music on hold, class 'default', on 
SIP/testcorp6-



But I do not hear the position announcement, only the prompt "thank you" 
which I don't want by the way.



Am I missing something in my configuration here ? Prompts like 
"queue_thereare" are not played, but prompts like "queue_thankyou" ARE 
played (which I don't want).

Why ?


Kind regards,

JOnas.
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Re: [asterisk-users] CDR with unix time.

[Rodrigo Lang]

2011/2/10 Tilghman Lesher 

> On Thursday 10 February 2011 06:13:38 Rodrigo Lang wrote:
> > I wonder if it is possible, without touching the source code, to
> > Asterisk save the cdr with date in unix time instead of the default
> > date. It's possible?
>
> The answer is, it depends upon the backend version you're using.  With
> cdr_pgsql and cdr_mysql from 1.6.2 forward, if the column type is integer
> or float, then the unix timestamp will be used.
>

Hi. I tested in the version 1.6.0 and works fine.

Thanks a lot.


Best regards,
-- 
Rodrigo Lang
Opening your mind - Just another Open Source
site
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Re: [asterisk-users] Asterisk 1.8.3

[Vinícius Fontes]

That makes two of us. I tried asking on asterisk-dev but had no reply.



- Mensagem original - 


Hi 

Does anyone have any rough idea how far away 1.8.3 is? 

We can't deploy 1.8 yet because of this issue 

https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=18403 
-- 
Ishfaq Malik 
Software Developer 
PackNet Ltd 

Office: 0161 660 3062 


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Re: [asterisk-users] Asterisk 1.8.3

[Захаров Антон]

On 11.02.2011 12:37, Ishfaq Malik wrote:

Hi

Does anyone have any rough idea how far away 1.8.3 is?

We can't deploy 1.8 yet because of this issue

https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=18403

Have you tried issue18403.patch ?

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[asterisk-users] Asterisk 1.8.3

[Ishfaq Malik]

Hi

Does anyone have any rough idea how far away 1.8.3 is?

We can't deploy 1.8 yet because of this issue

https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=18403
-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062


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Re: [asterisk-users] Ported Asterisk in Android

[Nikhil]

Thanks for reply. Any other suggestions .

On 12/20/2010 05:52 PM, Service clients - VDI CENTER wrote:

i believe there is a way to do it using asterisk and flashphoner
++

2010/12/20 Gilles mailto:codecompl...@free.fr>>

On Fri, 17 Dec 2010 15:51:33 +0530, Nikhil
mailto:d.nik...@cem-solutions.net>>
wrote:
> Does anyone ported Asterisk to Android OS .please give details

www.servalproject.org 


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--
Cordialement
Gabriel
09 79 94 71 13


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Re: [asterisk-users] Early audio SIP sequence order question

[Benoit Panizzon]

Hi Kevin

I just found something interresting:

http://www.faqs.org/rfc/rfc3960.txt

  1. Unless a 180 (Ringing) response is received, never generate
 local ringing.

  2. If a 180 (Ringing) has been received but there are no incoming
 media packets, generate local ringing.

  3. If a 180 (Ringing) has been received and there are incoming
 media packets, play them and do not generate local ringing.

So, yes, this most probably is an asterisk bug, if it's not a config issue I 
haven't figured out yet. Shall I submit a bug report?

And indeed, if you dial more than one endpoint and more than one is sending 
early audio, which one do you forward? I think nobody tought about that 
issue. Well as long as one is being forwarded that would be ok for our 
case :-)

Kind regards

Benoit Panizzon
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I m p r o W a r e   A G-
__

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CH-4133 PrattelnFax  +41 61 826 93 02
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Re: [asterisk-users] Gtalk/Jabber Issue

[William Stillwell]

1:1 nat, I even turned off iptables.. same issue.

 

Guess I will try install wireshark when I get back next week, im done farting 
with this tonight, when I get back from fort Lauderdale next week I will play 
with it some more.

 

 

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vladimir Mikhelson
Sent: Friday, February 11, 2011 2:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Gtalk/Jabber Issue

 

William,

Another thing to exclude is networking.  Can you verify that nothing blocks the 
specific traffic on your network?  Any chance of taking the packet trace on 
your gateway?

-Vladimir




On 2/11/2011 1:18 AM, William Stillwell wrote: 

I don’t’ appear to have an jabber [] OUTGOING packets?

 

I get just 1 incoming packet, and it just sits there, until it rings to 
voicemail.

 

 

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vladimir Mikhelson
Sent: Friday, February 11, 2011 1:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Gtalk/Jabber Issue

 

William,

I have gone through the similar frustration recently.  Everything works as of 
early morning yesterday. The big difference, I am on 1.8.2.3.

Have you seen this ticket on the tracker 
https://issues.asterisk.org/view.php?id=10512 ?   Anything applicable to your 
case?  The messages are identical to yours on the outgoing call.

-Vladimir




On 2/11/2011 12:32 AM, William Stillwell wrote: 

Still no dice..

 

This make no since.. ive gone over the config a million times now..

 

The windows gtalk /voice client works just fine.  (incoming and outgoing calls)

 

 

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vladimir Mikhelson
Sent: Friday, February 11, 2011 12:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Gtalk/Jabber Issue

 

William,

I have just noticed that you have several configuration statements commented 
out.

I would suggest to un-comment the "status=" in jabber.conf.  I would also 
suggest to un-comment the "timeout=", I am not that concerned of the 
"keepalive=".

You can reload jabber, no need to restart the Asterisk.

-Vladimir



On 2/10/2011 11:40 PM, Vladimir Mikhelson wrote: 

William,

Have you tried outgoing calls?  What happens there?

Have you restarted the Asterisk after you fixed the typo?

-Vladimir



On 2/10/2011 10:44 PM, William Stillwell wrote: 

Yeah, that was a typo, but I fixed, still no dice.

 

The incoming jabber call doesn’t fire the gtalk connection.

 

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby
Sent: Thursday, February 10, 2011 10:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Gtalk/Jabber Issue

 

You've got connection=jp_jabber defined in one file, and [jb_jabber] defined in 
the other. 

Thanks,

--Warren Selby, dCAP


On Feb 10, 2011, at 5:55 PM, "William Stillwell"  
wrote:

Sorry, Asterisk Build 1.6.2.7

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of William Stillwell
Sent: Thursday, February 10, 2011 6:50 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Gtalk/Jabber Issue

 

OK, im pulling my hair out, everything looks configured right, deleted, and 
started over, etc, etc. but can’t seem to get this to work

 

 

Gtalk.conf

 

[general]

context=google-in

allowguest=yes

bindaddr=192.168.xxx.xxx

extenip=96.254.xxx.xxx

 

[guest]

context=google-in

disallow=all

allow=ulaw

allow=g729

connection=jp_jabber

 

jabber.conf

 

[general]

debug=yes

;autoprune=no

autoregister=yes

 

 

[jb_jabber]

type=client

serverhost=talk.google.com

username=xx...@gmail.com/Talk

secret=XXX

port=5222

usetls=yes

usesasl=yes

;status=Available

statusmessage="Connected via Asterisk"

;timeout=100

;keepalive=yes

 

 

Extensions.conf

 

[google-in]

exten => s,1,NoOp(Call from GTalk)

exten => s,n,Set(CallerID(Name)="From GoogleTalk")

exten => s,n,Dial(SIP/1000)

 

jabber show connected 

 

Jabber Users and their status:

   User: xxx...@gmail.com/Talk - Connected



   Number of users: 1

 

 

 CLI on incoming Call 

 

bannana*CLI> 

JABBER: jb_jabber INCOMING: http://www.google.com/session";>http://www.google.com/session/phone";>http://www.google.com/transport/raw-udp"/>http://www.google.com/transport/p2p"/>

bannana*CLI> 

JABBER: jb_jabber INCOMING: http://www.google.com/session";>http://www.google.com/session/phone";>Call 
cancelled

bannana*CLI>

 

 

it doesn’t even try to fire the google-in context ?

 

Lastest Version of iksemel Installed, asterisk was rebuild after installed,