[asterisk-users] Help_video call not run

2011-12-20 Thread Durgesh Mishra 

- Forwarded Message - 
From: "Durgesh Mishra"  
To: "asterisk-users"  
Sent: Wednesday, December 21, 2011 10:36:06 AM GMT +05:30 Chennai, Kolkata, 
Mumbai, New Delhi 
Subject: Help_video call not run 






Hi all 



In sip.conf 

i take as 

[general] 

videosupport=yes 



 

[phone1] 
type=friend 
host=dynamic 
context= employees 
disallow=all 
allow=ilbc 
allow=g729 
allow=gsm 
allow=g723 
allow=ulaw 
allow=alaw 
allow=adpcm 
allow=h263p 
allow=h264 
allow=h263 

[phone2] 
 type=friend 
host=dynamic 
context= employees 
disallow=all 
allow=ilbc 
allow=g729 
allow=gsm 
allow=g723 
allow=ulaw 
allow=alaw 
allow=adpcm 
allow=h263p 
allow=h261 
allow=h263 









in extension.conf 

[employees] 

exten => 101,1,Dial(SIP/phone1,10) 

exten => 102,1,Playback(song2_check) 







in /var/lib/asterisk/sounds/en 

i store song2_check file(which is video file ,which has audio format   MPEG 
Layer 3) 



i check in  /var/lib/asterisk/sounds/en 

its shows as 

-rw-r--r-- 1 root root 89726486 Dec 20 10:18 song_check.mp4 

  

i check codecs of file through VLC player , its shows as its codec is using is 
H264. 



i check through cli  

core show codecs 

   : 

    131072 (1 << 17)    (0x2)  image    png   (PNG 
image) 
 262144 (1 << 18)    (0x4)  video   h261   (H.261 
Video) 
 524288 (1 << 19)    (0x8)  video   h263   (H.263 
Video) 
    1048576 (1 << 20)   (0x10)  video  h263p   (H.263+ 
Video) 
    2097152 (1 << 21)   (0x20)  video   h264   (H.264 
Video) 
    4194304 (1 << 22)   (0x40)  video  mpeg4   (MPEG4 
Video) 

  ::: 











i dial 102 from 101  
 phone 101(xlite)  has following codec support for H623 H623+ 



check log as 

[Dec 20 18:38:01] WARNING[10533] file.c: File song2_check does not exist in any 
format 
[Dec 20 18:38:01] WARNING[10533] file.c: Unable to open song2_check (format 
0x180400 (ilbc|h263|h263p)): No such file or directory 





phone1 goes just hung up. no vedio play 



I want to play video file. Plz tell me ,where i am wrong ,and how i can do it. 



thanks 





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[asterisk-users] OT - Which switch to play with LLDP-MED

2011-12-20 Thread Olivier 
Hi,

I would like to play with LLDP-MED in a lab, and specifically, to test
phone provisionning and auto-configuration (assign phones to VLANs,
...).

Eight 10/100 PoE ports would be enough for me.
Which model would you recommend ?

Regards.

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Re: [asterisk-users] Help_video call not run

2011-12-20 Thread amit anand 
Hi

what is the format of the file you are trying to play with exact codec info.


On Tue, Dec 20, 2011 at 19:17, Durgesh Mishra <
durgesh.mis...@rancoretech.com> wrote:

> Hi all
>
>
>
> In sip.conf
>
> i take as
>
> [general]
>
> videosupport=yes
>
>
>
>; then UDPTL will flow to the remote device
>
> [phone1]
> type=friend
> host=dynamic
> context= employees
> disallow=all
> allow=ilbc
> allow=g729
> allow=gsm
> allow=g723
> allow=ulaw
> allow=alaw
> allow=adpcm
> allow=h263p
> allow=h264
> allow=h263
>
> [phone2]
>  type=friend
> host=dynamic
> context= employees
> disallow=all
> allow=ilbc
> allow=g729
> allow=gsm
> allow=g723
> allow=ulaw
> allow=alaw
> allow=adpcm
> allow=h263p
> allow=h261
> allow=h263
>
>
>
>
>
>
>
>
>
> in extension.conf
>
> [employees]
>
> exten => 101,1,Dial(SIP/phone1,10)
>
> exten => 102,1,Playback(song2_check)
>
>
>
>
>
>
>
> in /var/lib/asterisk/sounds/en
>
> i store song2_check file(which is video file ,which has audio format
> MPEG Layer 3)
>
>
>
> i dial 102 from 101 
>  phone 101(xlite)  has following codec support for H623 H623+
>
>
>
> check log as
>
> [Dec 20 18:38:01] WARNING[10533] file.c: File song2_check does not exist
> in any format
> [Dec 20 18:38:01] WARNING[10533] file.c: Unable to open song2_check
> (format 0x180400 (ilbc|h263|h263p)): No such file or directory
>
>
>
>
>
> phone1 goes just hung up. no vedio play
>
>
>
> I want to play video file. Plz tell me ,where i am wrong ,and how i can do
> it.
>
>
>
> thanks
>
>
>
>
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
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>



-- 

Amit Anand


+91 9818559898
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[asterisk-users] queue not skipping ringing phone

2011-12-20 Thread Matt Hamilton 

I have a queue that distributes calls among 3 phones. When a phone is in use 
(including on hold), queue skips that device and sends the call to the next 
available one as expected. On the other hand, if a call comes in while one of 
the phones is ringing, the queue doesn't seem to recognize that phone as "in 
use" and sends the second call to the ringing phone. If the first call is 
answered, the second call is sent to the next available phone right away.

I'm new to asterisk and wondering if this is normal; I thought the ringing 
phone would be skipped "as in use" as well. Is there a setting on the asterisk 
side that I can use to force the queue to skip the "ringing" phone, or should 
this somehow be done on the phone itself?

Thanks,
Matt

Below is the queues.conf:


[qtemplate]
announce-frequency=0
announce-holdtime=no
announce-position=no
autofill=yes  
eventmemberstatus=no   
eventwhencalled=no 
joinempty=strict  
leavewhenempty=strict  
maxlen=0 
memberdelay=0
penaltymemberslimit=0
periodic-announce-frequency=0
queue-callswaiting=silence/1
Sendqueue-thereare=silence/1
queue-youarenext=silence/1
reportholdtime=no 
ringinuse=no   
servicelevel=60   
strategy=rrmemory
timeout=0 
timeoutpriority=app
timeoutrestart=no
retry=0 
weight=0
wrapuptime=0 
musicclass=default
monitor-type=MixMonitor
monitor-format=wav


[q1000](qtemplate)
member=Local/1001@handle-queue,,,SIP/1001
member=Local/1002@handle-queue,,,SIP/1002
member=Local/1003@handle-queue,,,SIP/1003
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[asterisk-users] Help_video call not run

2011-12-20 Thread Durgesh Mishra 





Hi all 



In sip.conf 

i take as 

[general] 

videosupport=yes 



 

[phone1] 
type=friend 
host=dynamic 
context= employees 
disallow=all 
allow=ilbc 
allow=g729 
allow=gsm 
allow=g723 
allow=ulaw 
allow=alaw 
allow=adpcm 
allow=h263p 
allow=h264 
allow=h263 

[phone2] 
 type=friend 
host=dynamic 
context= employees 
disallow=all 
allow=ilbc 
allow=g729 
allow=gsm 
allow=g723 
allow=ulaw 
allow=alaw 
allow=adpcm 
allow=h263p 
allow=h261 
allow=h263 









in extension.conf 

[employees] 

exten => 101,1,Dial(SIP/phone1,10) 

exten => 102,1,Playback(song2_check) 







in /var/lib/asterisk/sounds/en 

i store song2_check file(which is video file ,which has audio format   MPEG 
Layer 3) 



i dial 102 from 101  
 phone 101(xlite)  has following codec support for H623 H623+ 



check log as 

[Dec 20 18:38:01] WARNING[10533] file.c: File song2_check does not exist in any 
format 
[Dec 20 18:38:01] WARNING[10533] file.c: Unable to open song2_check (format 
0x180400 (ilbc|h263|h263p)): No such file or directory 





phone1 goes just hung up. no vedio play 



I want to play video file. Plz tell me ,where i am wrong ,and how i can do it. 



thanks 





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Re: [asterisk-users] Use different local IP for each SIP trunk

2011-12-20 Thread Anton Kvashenkin 
Externip support per device in sip.conf
http://edvina.net/products/edvx/

2011/12/20 giovanni.v 

> Il 20/12/2011 6.07, Anton Kvashenkin ha scritto:
>
>  you can add exterin= in sip.conf for each trunk
>>
>
> I think this can be used only in [general] section not on peers
> definition; also useful only when asterisk is behind nat. Not?
>
>
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Re: [asterisk-users] sendvoicemail=yes not quite working

2011-12-20 Thread Todd Routhier 
On Tue, Dec 20, 2011 at 7:03 PM, M Maki  wrote:

> I have a system working great with the exception of the sendvoicemail=yes
> voicemail.conf option. I can not figure out what I am missing or have
> configured wrong...
>
>
> While in voicemail after selecting 3 for advanced options, then 5 to leave
> a message I am directed to the correct mailbox. But after hearing the
> mailbox number/name announcement I am immediately taken back to my mailbox.
> No option is given to leave a message. I can forward messages, but can't
> leave a message. All other aspects of the voicemail system I have tested
> work great.This is on Debian Squeeze (Asterisk 1.8.8.0 on a x86_64) with
> about 120 phones.
>
> Here is the vebose/debug output of that part of the call.
>
>
> == Using SIP RTP CoS mark 5
>  -- Executing [8000@LocalSets:1] VoiceMailMain("SIP/3323-0499", "")
> in new stack
>  -- Playing 'vm-login.ulaw' (language 'en')
>  -- Playing 'vm-password.ulaw' (language 'en')
>  -- Playing 'vm-youhave.ulaw' (language 'en')
>  -- Playing 'vm-no.ulaw' (language 'en')
>  -- Playing 'vm-messages.ulaw' (language 'en')
>  -- Playing 'vm-leavemsg.ulaw' (language 'en')
>  -- Playing 'vm-starmain.ulaw' (language 'en')
>  -- Playing 'vm-extension.ulaw' (language 'en')
>  -- Playing '/var/spool/asterisk/voicemail/default/3318/greet.slin'
> (language 'en')
> <-THIS IS WHERE IT GOES BACK TO MY VOICEMAIL BOX >
>  == Using SIP RTP CoS mark 5
>  -- Playing 'vm-opts.ulaw' (language 'en')
>  == Spawn extension (LocalSets, 8000, 1) exited non-zero on
> 'SIP/3323-0499'
>
>
> Thanks!
>
> Mike
>
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> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Pretty sure you need this in your voicemail.conf file:

sendvoicemail=yes
sendvoicemailThis setting takes a *yes* or *no* value. It enables the
"Leave a message" menu option from the Advanced Options menu which allows
the voicemail user to send a message to another voicemail user.
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Re: [asterisk-users] sendvoicemail=yes not quite working

2011-12-20 Thread Todd Routhier 
On Tue, Dec 20, 2011 at 8:53 PM, Todd Routhier  wrote:

>
>
> On Tue, Dec 20, 2011 at 7:03 PM, M Maki  wrote:
>
>> I have a system working great with the exception of the sendvoicemail=yes
>> voicemail.conf option. I can not figure out what I am missing or have
>> configured wrong...
>>
>>
>> While in voicemail after selecting 3 for advanced options, then 5 to
>> leave a message I am directed to the correct mailbox. But after hearing the
>> mailbox number/name announcement I am immediately taken back to my mailbox.
>> No option is given to leave a message. I can forward messages, but can't
>> leave a message. All other aspects of the voicemail system I have tested
>> work great.This is on Debian Squeeze (Asterisk 1.8.8.0 on a x86_64) with
>> about 120 phones.
>>
>> Here is the vebose/debug output of that part of the call.
>>
>>
>> == Using SIP RTP CoS mark 5
>>  -- Executing [8000@LocalSets:1] VoiceMailMain("SIP/3323-0499", "")
>> in new stack
>>  -- Playing 'vm-login.ulaw' (language 'en')
>>  -- Playing 'vm-password.ulaw' (language 'en')
>>  -- Playing 'vm-youhave.ulaw' (language 'en')
>>  -- Playing 'vm-no.ulaw' (language 'en')
>>  -- Playing 'vm-messages.ulaw' (language 'en')
>>  -- Playing 'vm-leavemsg.ulaw' (language 'en')
>>  -- Playing 'vm-starmain.ulaw' (language 'en')
>>  -- Playing 'vm-extension.ulaw' (language 'en')
>>  -- Playing '/var/spool/asterisk/voicemail/default/3318/greet.slin'
>> (language 'en')
>> <-THIS IS WHERE IT GOES BACK TO MY VOICEMAIL BOX >
>>  == Using SIP RTP CoS mark 5
>>  -- Playing 'vm-opts.ulaw' (language 'en')
>>  == Spawn extension (LocalSets, 8000, 1) exited non-zero on
>> 'SIP/3323-0499'
>>
>>
>> Thanks!
>>
>> Mike
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>   http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
> Pretty sure you need this in your voicemail.conf file:
>
> sendvoicemail=yes
> sendvoicemailThis setting takes a *yes* or *no* value. It enables the
> "Leave a message" menu option from the Advanced Options menu which allows
> the voicemail user to send a message to another voicemail user.
>
>
Oh, wow.. Nevermind, you started your original post saying you have that
option set.
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[asterisk-users] sendvoicemail=yes not quite working

2011-12-20 Thread M Maki 
I have a system working great with the exception of the sendvoicemail=yes 
voicemail.conf option. I can not figure out what I am missing or have 
configured wrong... 


While in voicemail after selecting 3 for advanced options, then 5 to leave a 
message I am directed to the correct mailbox. But after hearing the mailbox 
number/name announcement I am immediately taken back to my mailbox. No option 
is given to leave a message. I can forward messages, but can't leave a message. 
All other aspects of the voicemail system I have tested work great.This is on 
Debian Squeeze (Asterisk 1.8.8.0 on a x86_64) with about 120 phones.

Here is the vebose/debug output of that part of the call.


== Using SIP RTP CoS mark 5
 -- Executing [8000@LocalSets:1] VoiceMailMain("SIP/3323-0499", "") in new 
stack
 -- Playing 'vm-login.ulaw' (language 'en')
 -- Playing 'vm-password.ulaw' (language 'en')
 -- Playing 'vm-youhave.ulaw' (language 'en')
 -- Playing 'vm-no.ulaw' (language 'en')
 -- Playing 'vm-messages.ulaw' (language 'en')
 -- Playing 'vm-leavemsg.ulaw' (language 'en')
 -- Playing 'vm-starmain.ulaw' (language 'en')
 -- Playing 'vm-extension.ulaw' (language 'en')
 -- Playing '/var/spool/asterisk/voicemail/default/3318/greet.slin' (language 
'en')
<-THIS IS WHERE IT GOES BACK TO MY VOICEMAIL BOX >
 == Using SIP RTP CoS mark 5
 -- Playing 'vm-opts.ulaw' (language 'en')
 == Spawn extension (LocalSets, 8000, 1) exited non-zero on 'SIP/3323-0499'


Thanks!

Mike

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Re: [asterisk-users] Populate CDR issues

2011-12-20 Thread Harel Cohen 
Hi Mike,
I've tried updating my CDR's via the h exten but with no success. I've tried 
with both endbeforehexten=no and endbeforehexten=yes (in cdr.conf) but the 
value refused to appear in my CDR (even though I see the Set() application 
being executed in the console under the h exten). Thank you for your suggestion 
though...

Any other thoughts are welcome.


Kind Regards,
Harel Cohen

-Original Message-

Date: Mon, 12 Dec 2011 13:41:31 -0700
From: Mike Diehl 
Subject: Re: [asterisk-users] Populate CDR issues
To: asterisk-users@lists.digium.com
Cc: Harel Cohen 
Message-ID: <201112121341.32142.mdi...@diehlnet.com>
Content-Type: Text/Plain;  charset="iso-8859-1"

On Monday 12 December 2011 4:28:17 am Harel Cohen wrote:
> Danny,
> 
> Why would you think this is a "circumvent"? I'm using a nice feature 
> of 1.8 where I can create any CDR field I like and populate it by 
> using the
> CDR() function. While all other fields that I created are 
> populated properly (however before the 'dial' commences) it seems like 
> at this point of the dial plan the CDR is closed for editing even 
> though I configured endbeforehexten=no in my cdr.conf.

I agree, this is a perfectly valid use of the CDR.  I do the same thing, btw.  
I think what you are seeing is that when your call starts, Asterisk creates a 
record, either in memory, or in a db transaction.  When the call is torn down, 
the record is updated and committed to the db.  The down-shot is that any 
changes you make to the db record get clobbered by this last update.

I ended up making some of my updates in the hang-up phase via the "h" 
extension.  See if that will do what you need.



-- 

Take care and have fun,
Mike Diehl.

--

Date: Thu, 1 Dec 2011 12:57:56 +0100
From: Harel Cohen 
Subject: [asterisk-users] Populate CDR issues
To: "asterisk-users@lists.digium.com"

Message-ID:

Content-Type: text/plain; charset="us-ascii"

Hello list,
I'm trying to populate my CDR logs with values which are available after the 
call has started (e.g. signalling IP of remote user, media IP, codec etc.). 
While CHANNEL function give me all I need for the incoming leg (leg A), I can't 
get the relevant values for the outgoing channel. I've tried using the option 
'U' with my dial command (execute subroutine for called channel after called 
channel answered but before the call is bridged). While this throws the correct 
information to the console it does not populate the CDRs accordingly.
Note: Asterisk ver is 1.8.7.1 and CDR's are written to MySQL with adaptive ODBC 
and the table therein contains the relevant fields.

This is the console with 'very-verbose' output for the 'Dial' application where 
office_Admin2, IP 192.168.20.222, is calling office_ServerRoom, IP 
192.168.20.226. My comments added prefixed by ** and on separate line:

** channel here is source channel: SIP/office_Admin2-0015
[Dec  1 12:14:31] -- Executing [316@InternalDP:5] 
Dial("SIP/office_Admin2-0015", "SIP/office_ServerRoom,,FgU(jump2SetVar)") 
in new stack
[Dec  1 12:14:31]   == Using UDPTL CoS mark 5
[Dec  1 12:14:31]   == Using SIP RTP CoS mark 5
[Dec  1 12:14:31] -- Called SIP/office_ServerRoom
[Dec  1 12:14:31] -- SIP/office_ServerRoom-0016 is ringing
[Dec  1 12:14:31] -- SIP/office_ServerRoom-0016 is ringing
[Dec  1 12:14:33] -- SIP/office_ServerRoom-0016 answered 
SIP/office_Admin2-0015
** from here the channel is the destination channel: 
SIP/office_ServerRoom-0016
[Dec  1 12:14:33] -- Executing [s@jump2SetVar:1] 
Gosub("SIP/office_ServerRoom-0016", "SetVar,postdial,1") in new stack
** This is how I obtain channel information:
** exten => 
postdial,1,Set(CDR(chanoutsigip)=${CHANNEL(peerip)}:${SIPPEER(${CHANNEL(peername)},port)})
** same => n,Set(CDR(chanoutmediaip)=${CHANNEL(rtpdest,audio)})
** same => n,Set(CDR(chanoutcodec)=${CHANNEL(audionativeformat)})
[Dec  1 12:14:33] -- Executing [postdial@SetVar:1] 
Set("SIP/office_ServerRoom-0016", "CDR(chanoutsigip)=192.168.20.226:5065") 
in new stack
[Dec  1 12:14:33] -- Executing [postdial@SetVar:2] 
Set("SIP/office_ServerRoom-0016", 
"CDR(chanoutmediaip)=192.168.20.226:23008") in new stack
[Dec  1 12:14:33] -- Executing [postdial@SetVar:3] 
Set("SIP/office_ServerRoom-0016", "CDR(chanoutcodec)=g729") in new stack
[Dec  1 12:14:33] -- Executing [postdial@SetVar:4] 
Goto("SIP/office_ServerRoom-0016", "endsub,1") in new stack
[Dec  1 12:14:33] -- Goto (SetVar,endsub,1)
[Dec  1 12:14:33] -- Executing [endsub@SetVar:1] 
Return("SIP/office_ServerRoom-0016", "") in new stack
[Dec  1 12:14:33] -- Executing [s@jump2SetVar:2] 
Return("SIP/office_ServerRoom-0016", "") in new stack
[Dec  1 12:14:33] -- Executing [s@app_dial_gosub_virtual_context:1] 
NoOp("SIP/office_ServerRoom-0016", "") in new stack
[Dec  1 12:14:33] -- Auto fallthrough, channel 
'SIP/office_ServerRoom-0016' status is 'U

Re: [asterisk-users] Limit # of inbound calls on SIP trunk

2011-12-20 Thread isrlgb 
Well freepbx has that in the gui you should read the tool tips
Read the trunk limit tooltip 



-Original Message-
From: Steve Edwards 
Sender: asterisk-users-boun...@lists.digium.com
Date: Tue, 20 Dec 2011 12:16:48 
To: Asterisk Users Mailing List - Non-Commercial 
Discussion
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion

Subject: Re: [asterisk-users] Limit # of inbound calls on SIP trunk

Un-top-posting...

> On Mon, 19 Dec 2011, Douglas Mortensen wrote:

>> I have a system with FreePBX, and as far as I can tell it does not 
>> provide a means to limit the number of simultaneous inbound calls on a 
>> SIP trunk. Therefore I suspect that I’ll need to do some manual 
>> dialplan manipulation.

> On Mon, 19 Dec 2011, Steve Edwards wrote:

> The GROUP() and GROUP_COUNT() functions and the GOTOIF() application 
> should do the trick.

On Tue, 20 Dec 2011, Douglas Mortensen wrote:

> Excellent. Do you think these functions would enable me to create rules 
> based on both the concurrent # of inbound and/or outbound calls, or only 
> total # of concurrent calls (agnostic to call direction being inbound 
> vs. outbound)?

If you want a call to be a member of multiple groups, you have to play 
with the category parameter.

 exten = *,n,set(GROUP()=incoming)
 exten = *,n,set(GROUP(incoming)=no)
 exten = *,n,set(GROUP(incoming)=yes)
 exten = *,n,set(GROUP()=outgoing)
 exten = *,n,set(GROUP(outgoing)=no)
 exten = *,n,set(GROUP(outgoing)=yes)
 exten = *,n,verbose(incoming count = ${GROUP_COUNT(incoming)})
 exten = *,n,verbose(outgoing count = ${GROUP_COUNT(outgoing)})
 exten = *,n,verbose(incoming category count = 
${GROUP_COUNT(yes@incoming)})
 exten = *,n,verbose(outgoing category count = 
${GROUP_COUNT(yes@outgoing)})
 exten = *,n,verbose(group list is ${GROUP_LIST()})

-- 
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] Call pickup on Asterisk 1.8 and Polycom IP550s?

2011-12-20 Thread Justin Sherrill 
For what it's worth, the phone is getting enough information.  The first call 
works fine - it's the second call that never triggers the pickup screen, though 
it does cause the lamp to blink for that line.  It's like the phone understands 
"ringing" but not "busy+ringing".  I'm tempted to say it's a Polycom firmware 
issue, but I haven't seen an errata items that matches.

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gord Urquhart
Sent: Friday, December 16, 2011 2:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call pickup on Asterisk 1.8 and Polycom IP550s?

It sounds like the phone is not getting enough info to do a directed pickup, 
have you turned on NotifyCID in sip.conf? If that does'nt work try using  the 
extended BLF stuff (described here http://www.excaliburtech.net/archives/147 
and here http://www.voip-info.org/wiki/view/Asterisk+presence)

gordu

On Thu, Dec 15, 2011 at 12:10 PM, Justin Sherrill 
mailto:justin.sherr...@americanrocksalt.com>>
 wrote:
This is one of those "Is anyone else doing this?/Is anyone else seeing this?" 
posts.

We have an Asterisk 1.8.4 system, with Polycom IP550 phones running firmware 
3.2.3.  If someone on the 'buddy list' - the list of other extensions to watch 
- is called, the phone gets a NOTIFY event and displays a screen with the call 
information and a pickup softkey.

However, if someone on that list is already on the phone and they get a second 
incoming call, the NOTIFY event comes in but the phone never displays the 
changed screen with the pickup button.  It'll flash the light next to that 
extension, but that's it.

Is anyone using a similar setup and seeing this?  It's somewhat rare, but I 
have an office location where everyone there likes to pick up other people's 
calls, and they haven't been using a call queue like they oughta.

Justin Sherrill - American Rock Salt
P: 585-991-6825 F: 585-991-6925 C: 
585-298-6826



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Re: [asterisk-users] GOIP GSM to SIP Gateway?

2011-12-20 Thread Jim Dickenson 
I would think it would be better to set a variable for each user and then have 
a single context with something like:

_NXX,1,Dial(SIP/${WhatToUse}/${EXTEN})

Or something like this.
-- 
Jim Dickenson
mailto:dicken...@cfmc.com

CfMC
http://www.cfmc.com/



On Dec 20, 2011, at 1:03 PM, John Kiniston wrote:

> 
> On Tue, Dec 20, 2011 at 12:39 PM, Matt  wrote:
> 
> Is there anyway (short of defining dial an 8 from this phone for this
> trunk to this SIM and a 9 from this phone for a trunk to this SIM) to
> get it to use certain SIM cards when calls are made from certain
> phones?
> 
> You could define multiple contexts with different pattern matches for each 
> GSM connection and and set your phones to use them, phones 1-3 in context1, 
> phones 4-6 in context2, etc.
> 
> [context1] 
> _NXX,1,Dial(SIP/GSM1/${EXTEN})
> 
> [context2]
> _NXX,1,Dial(SIP/GSM2/${EXTEN})
> 
> [context3]
> _NXX,1,Dial(SIP/GSM3/${EXTEN})
> 
> -- 
> A human being should be able to change a diaper, plan an invasion, butcher a 
> hog, conn a ship, design a building, write a sonnet, balance accounts, build 
> a wall, set a bone, comfort the dying, take orders, give orders, cooperate, 
> act alone, solve equations, analyze a new problem, pitch manure, program a 
> computer, cook a tasty meal, fight efficiently, die gallantly. Specialization 
> is for insects.
> ---Heinlein
> --
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Re: [asterisk-users] GOIP GSM to SIP Gateway?

2011-12-20 Thread John Kiniston 
On Tue, Dec 20, 2011 at 12:39 PM, Matt  wrote:

>
> Is there anyway (short of defining dial an 8 from this phone for this
> trunk to this SIM and a 9 from this phone for a trunk to this SIM) to
> get it to use certain SIM cards when calls are made from certain
> phones?
>
> You could define multiple contexts with different pattern matches for each
GSM connection and and set your phones to use them, phones 1-3 in context1,
phones 4-6 in context2, etc.

[context1]
_NXX,1,Dial(SIP/GSM1/${EXTEN})

[context2]
_NXX,1,Dial(SIP/GSM2/${EXTEN})

[context3]
_NXX,1,Dial(SIP/GSM3/${EXTEN})

-- 
A human being should be able to change a diaper, plan an invasion, butcher
a hog, conn a ship, design a building, write a sonnet, balance accounts,
build a wall, set a bone, comfort the dying, take orders, give orders,
cooperate, act alone, solve equations, analyze a new problem, pitch manure,
program a computer, cook a tasty meal, fight efficiently, die gallantly.
Specialization is for insects.
---Heinlein
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Re: [asterisk-users] OOH323 config file

2011-12-20 Thread Paul Belanger 

On 11-12-20 11:21 AM, Carlos Chavez wrote:

Just a warning to people trying to use ooh323 with Asterisk 1.8.7.  The
example config file that comes with asterisk is called chan_ooh323.conf
when it actually should be named ooh323.conf for it to work.  Sent me
into a panic when I was trying to install an H323 link to an Avaya
server and the ooh323 module would not load because it could not find
its configuration file.  The file needs to be renamed.  Should this be
classified as a bug in the bug tracker?

Yes, open an issue on the tracker, if this is the case we can fix it for 
the next release.


--
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] Limit # of inbound calls on SIP trunk

2011-12-20 Thread Steve Edwards 

Un-top-posting...


On Mon, 19 Dec 2011, Douglas Mortensen wrote:


I have a system with FreePBX, and as far as I can tell it does not 
provide a means to limit the number of simultaneous inbound calls on a 
SIP trunk. Therefore I suspect that I’ll need to do some manual 
dialplan manipulation.



On Mon, 19 Dec 2011, Steve Edwards wrote:


The GROUP() and GROUP_COUNT() functions and the GOTOIF() application 
should do the trick.


On Tue, 20 Dec 2011, Douglas Mortensen wrote:

Excellent. Do you think these functions would enable me to create rules 
based on both the concurrent # of inbound and/or outbound calls, or only 
total # of concurrent calls (agnostic to call direction being inbound 
vs. outbound)?


If you want a call to be a member of multiple groups, you have to play 
with the category parameter.


exten = *,n,set(GROUP()=incoming)
exten = *,n,set(GROUP(incoming)=no)
exten = *,n,set(GROUP(incoming)=yes)
exten = *,n,set(GROUP()=outgoing)
exten = *,n,set(GROUP(outgoing)=no)
exten = *,n,set(GROUP(outgoing)=yes)
exten = *,n,verbose(incoming count = ${GROUP_COUNT(incoming)})
exten = *,n,verbose(outgoing count = ${GROUP_COUNT(outgoing)})
exten = *,n,verbose(incoming category count = 
${GROUP_COUNT(yes@incoming)})
exten = *,n,verbose(outgoing category count = 
${GROUP_COUNT(yes@outgoing)})
exten = *,n,verbose(group list is ${GROUP_LIST()})

--
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] GOIP GSM to SIP Gateway?

2011-12-20 Thread Carlos Rojas 
Hello

It is possible but how do you have the dialplan ?
In your dial plan you can do that

Regards
On Dec 20, 2011 2:40 PM, "Matt"  wrote:

> Hi,
> Has anyone here any experiencing with linking an Asterisk PBX to a
> GOIP GSM to SIP Gateway?  We've got inbound calls from the GSM network
> working properly, however, outbound calls seem to randomly choose a
> SIM line to use.
>
> Is there anyway (short of defining dial an 8 from this phone for this
> trunk to this SIM and a 9 from this phone for a trunk to this SIM) to
> get it to use certain SIM cards when calls are made from certain
> phones?
>
> --
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[asterisk-users] GOIP GSM to SIP Gateway?

2011-12-20 Thread Matt 
Hi,
Has anyone here any experiencing with linking an Asterisk PBX to a
GOIP GSM to SIP Gateway?  We've got inbound calls from the GSM network
working properly, however, outbound calls seem to randomly choose a
SIM line to use.

Is there anyway (short of defining dial an 8 from this phone for this
trunk to this SIM and a 9 from this phone for a trunk to this SIM) to
get it to use certain SIM cards when calls are made from certain
phones?

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Re: [asterisk-users] Limit # of inbound calls on SIP trunk

2011-12-20 Thread Douglas Mortensen 
Excellent. Do you think these functions would enable me to create rules based 
on both the concurrent # of inbound and/or outbound calls, or only total # of 
concurrent calls (agnostic to call direction being inbound vs. outbound)?

Thanks,
-
Doug Mortensen
Network Consultant
Impala Networks
P: 505.327.7300


-Original Message-
From: Steve Edwards [mailto:asterisk@sedwards.com] 
Sent: Monday, December 19, 2011 5:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Limit # of inbound calls on SIP trunk

On Mon, 19 Dec 2011, Douglas Mortensen wrote:

> I have a system with FreePBX, and as far as I can tell it does not 
> provide a means to limit the number of simultaneous inbound calls on a 
> SIP trunk. Therefore I suspect that I’ll need to do some manual 
> dialplan manipulation.

The GROUP() and GROUP_COUNT() functions and the GOTOIF() application should do 
the trick.

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

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[asterisk-users] OOH323 config file

2011-12-20 Thread Carlos Chavez 
Just a warning to people trying to use ooh323 with Asterisk 1.8.7.  The
example config file that comes with asterisk is called chan_ooh323.conf
when it actually should be named ooh323.conf for it to work.  Sent me
into a panic when I was trying to install an H323 link to an Avaya
server and the ooh323 module would not load because it could not find
its configuration file.  The file needs to be renamed.  Should this be
classified as a bug in the bug tracker?

-- 
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001


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Re: [asterisk-users] Dialing problem with Polycom phones after SIP update

2011-12-20 Thread Doug Lytle 


Eric Wieling wrote:

Polycom (r) UC Software: Configuration File Conversion Utility\

On the 
pagehttp://support.polycom.com/PolycomService/support/us/support/voice/soundpoint_ip/soundpoint_ip560.html



And for those of us without Windows, this utility appears to work fine 
under wine.


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] PITCH_SHIFT()

2011-12-20 Thread Leif Madsen 

On 20/12/11 01:15 AM, John Jolly wrote:

In Leif Madsen's AstriCon 2010 talk titled "5 Things You Didn't
Know Asterisk Could Do
"
 he
mentions that the PITCH_SHIFT() function is designed to be used
dynamically and can change the pitch of a channel on the fly
using features.conf. Can someone provide me with any information of how
this would be accomplished for dynamic use? I'm familiar with the
dialplan syntax use examples such as:

exten => 1,1,Set(PITCH_SHIFT(tx)=highest); raises pitch an octave
exten => 1,1,Set(PITCH_SHIFT(rx)=higher) ; raises pitch more
exten => 1,1,Set(PITCH_SHIFT(both)=high) ; raises pitch

and so forth, but don't understand how these functions would be called
dynamically from features.conf.


You'd just create the application_map as documented in features.conf and 
then apply the PITCH_SHIFT() function to whichever channel you want. 
Untested, but should look something like:


pitch_up_them => 3*,peer/both,Set(PITCH_SHIFT(tx)=high)

--
Leif Madsen
http://www.oreilly.com/catalog/asterisk

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Re: [asterisk-users] India Telecom regulations

2011-12-20 Thread Nick Khamis 
How can we get thise license? Who do we have to pay.

Nick.

On Tue, Dec 20, 2011 at 9:52 AM, khalid touati  wrote:
> Thank you Raj,
> I hope it will soon require no license as I heard there is a project to
> change this law, for now I believe I will recommend our office in India to
> go for license (to bridge to PSTN).
> Thanks once more for your help!
>
> 2011/12/19 Raj Mathur (राज माथुर) 
>
>> On Tuesday 20 Dec 2011, khalid touati wrote:
>> > Thank you Raj,
>> > so with VOIP license calls can go beyond our pbx to PSTN (india),
>> > right, if so this what i needed to know to call Indian cellphone
>> > from US (or  other countries)
>>
>> If your objective is to originate calls in the US (using whatever
>> technology), route them over SIP and then terminate them to the PSTN in
>> India, then yes: your Indian presence would need a VoIP licence.
>> Similarly for the reverse: originate a call from Indian PSTN to your
>> local office here and route it using VoIP to any destination (whether
>> within India or abroad).  A licence is required in that case too.
>>
>> In general, interconnection of two different entities by bridging Indian
>> PSTN with any other technology requires a licence.  If you're only doing
>> VoIP-VoIP, or PSTN-PSTN, or bridging an Indian VoIP call to PSTN outside
>> India then it's permitted in principle.  This is why, e.g., Skype is
>> permitted: it doesn't connect to the Indian PSTN at any stage.
>>
>> Once again, IANAL and TINLA.  This is purely from my (mostly informed)
>> understanding of the current laws.
>>
>> Regards,
>>
>> -- Raj
>> --
>> Raj Mathur                          || r...@kandalaya.org   || GPG:
>> http://otheronepercent.blogspot.com || http://kandalaya.org || CC68
>> It is the mind that moves           || http://schizoid.in   || D17F
>>
>> --
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>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
>
> --
> Khalid Touati
> Network Administrator at Endosoft, LLC
> CCNA
>
>
>
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Re: [asterisk-users] India Telecom regulations

2011-12-20 Thread khalid touati 
Thank you Raj,
I hope it will soon require no license as I heard there is a project to
change this law, for now I believe I will recommend our office in India to
go for license (to bridge to PSTN).
Thanks once more for your help!

2011/12/19 Raj Mathur (राज माथुर) 

> On Tuesday 20 Dec 2011, khalid touati wrote:
> > Thank you Raj,
> > so with VOIP license calls can go beyond our pbx to PSTN (india),
> > right, if so this what i needed to know to call Indian cellphone
> > from US (or  other countries)
>
> If your objective is to originate calls in the US (using whatever
> technology), route them over SIP and then terminate them to the PSTN in
> India, then yes: your Indian presence would need a VoIP licence.
> Similarly for the reverse: originate a call from Indian PSTN to your
> local office here and route it using VoIP to any destination (whether
> within India or abroad).  A licence is required in that case too.
>
> In general, interconnection of two different entities by bridging Indian
> PSTN with any other technology requires a licence.  If you're only doing
> VoIP-VoIP, or PSTN-PSTN, or bridging an Indian VoIP call to PSTN outside
> India then it's permitted in principle.  This is why, e.g., Skype is
> permitted: it doesn't connect to the Indian PSTN at any stage.
>
> Once again, IANAL and TINLA.  This is purely from my (mostly informed)
> understanding of the current laws.
>
> Regards,
>
> -- Raj
> --
> Raj Mathur  || r...@kandalaya.org   || GPG:
> http://otheronepercent.blogspot.com || http://kandalaya.org || CC68
> It is the mind that moves   || http://schizoid.in   || D17F
>
> --
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>



-- 
Khalid Touati
Network Administrator at Endosoft, LLC
CCNA
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Re: [asterisk-users] Dialing problem with Polycom phones after SIP update

2011-12-20 Thread Eric Wieling 
Polycom (r) UC Software: Configuration File Conversion Utility\

On the page 
http://support.polycom.com/PolycomService/support/us/support/voice/soundpoint_ip/soundpoint_ip560.html

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justin Sherrill
Sent: Tuesday, December 20, 2011 8:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dialing problem with Polycom phones after SIP 
update

Out of curiosity, what is "the Polycom script"?

I obviously haven't moved from 3.2.x firmware yet.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: Friday, December 16, 2011 4:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dialing problem with Polycom phones after SIP 
update

Did you run your old configurations thru the Polycom script to convert them to 
work with 3.3+?

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Marco Mooijekind
Sent: Friday, December 16, 2011 4:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dialing problem with Polycom phones after SIP 
update

Hello Gord,

the line icon is solid black, which should indicate the lines are registered. 

Marco.



On Fri, Dec 16, 2011 at 10:24 PM, Gord Urquhart  wrote:


Does the phone show the line as registered? The little phone icon on 
the display should be solid for a registered line and just a outline for a 
unregistered line. Using wireshark to watch the SIP traffic is a easy way to 
ensure the REGISTER signally is complete.




On Fri, Dec 16, 2011 at 1:02 PM, Marco Mooijekind 
 wrote:


Dear all,

I'm using serveral Polycom 335 and 650 phones on Asterisk 1.8.
All worked well. After applying the new Polycom UC 4.0.1 
software update to the phones I notice the following:

When dialing an extension, either on- or off hook, the phone 
immediately displays "SIP URL:..".
This does not allow me to enter a regular numeric extension.
The Polycom admin manual states that the phone displays the SIP 
URL input message if the phone is not registered.
This is strange since i do see the phones registering 
themselves in the Asterisk verbose logging.

Anyone experiencing this problem , any tips!

Thanks in advance!

Marco Mooijekind.


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Re: [asterisk-users] Dialing problem with Polycom phones after SIP update

2011-12-20 Thread Justin Sherrill 
Out of curiosity, what is "the Polycom script"?

I obviously haven't moved from 3.2.x firmware yet.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
Sent: Friday, December 16, 2011 4:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dialing problem with Polycom phones after SIP 
update

Did you run your old configurations thru the Polycom script to convert them to 
work with 3.3+?

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Marco Mooijekind
Sent: Friday, December 16, 2011 4:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dialing problem with Polycom phones after SIP 
update

Hello Gord,

the line icon is solid black, which should indicate the lines are registered. 

Marco.



On Fri, Dec 16, 2011 at 10:24 PM, Gord Urquhart  wrote:


Does the phone show the line as registered? The little phone icon on 
the display should be solid for a registered line and just a outline for a 
unregistered line. Using wireshark to watch the SIP traffic is a easy way to 
ensure the REGISTER signally is complete.




On Fri, Dec 16, 2011 at 1:02 PM, Marco Mooijekind 
 wrote:


Dear all,

I'm using serveral Polycom 335 and 650 phones on Asterisk 1.8.
All worked well. After applying the new Polycom UC 4.0.1 
software update to the phones I notice the following:

When dialing an extension, either on- or off hook, the phone 
immediately displays "SIP URL:..".
This does not allow me to enter a regular numeric extension.
The Polycom admin manual states that the phone displays the SIP 
URL input message if the phone is not registered.
This is strange since i do see the phones registering 
themselves in the Asterisk verbose logging.

Anyone experiencing this problem , any tips!

Thanks in advance!

Marco Mooijekind.


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[asterisk-users] Help_video call not run

2011-12-20 Thread Durgesh Mishra 


Hi all 



In sip.conf 

i take as 

[general] 

videosupport=yes 



   ; then UDPTL will flow to the remote device 

[phone1] 
type=friend 
host=dynamic 
context= employees 
disallow=all 
allow=ilbc 
allow=g729 
allow=gsm 
allow=g723 
allow=ulaw 
allow=alaw 
allow=adpcm 
allow=h263p 
allow=h264 
allow=h263 

[phone2] 
 type=friend 
host=dynamic 
context= employees 
disallow=all 
allow=ilbc 
allow=g729 
allow=gsm 
allow=g723 
allow=ulaw 
allow=alaw 
allow=adpcm 
allow=h263p 
allow=h261 
allow=h263 









in extension.conf 

[employees] 

exten => 101,1,Dial(SIP/phone1,10) 

exten => 102,1,Playback(song2_check) 

  





in /var/lib/asterisk/sounds/en 

i store song2_check file(which is video file ,which has audio format   MPEG 
Layer 3) 



i dial 102 from 101  
 phone 101(xlite)  has following codec support for H623 H623+ 



check log as 

[Dec 20 18:38:01] WARNING[10533] file.c: File song2_check does not exist in any 
format 
[Dec 20 18:38:01] WARNING[10533] file.c: Unable to open song2_check (format 
0x180400 (ilbc|h263|h263p)): No such file or directory 





phone1 goes just hung up. no vedio play 



I want to play video file. Plz tell me ,where i am wrong ,and how i can do it. 



thanks 



  

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Re: [asterisk-users] File Convert

2011-12-20 Thread Tzafrir Cohen 
On Tue, Dec 20, 2011 at 05:34:46PM +0530, Gopalakrishnan N wrote:
> Hi users,
> 
> I have Asterisk 1.6.2.20 in Ubuntu 10.04. I am trying to convert a gsm file
> to G729 using file convert, but I am facing error as follows,
> 
> file convert /tmp/welcome.gsm /tmp/welcome.g729
> Failed to convert /tmp/welcome.gsm to /tmp/welcome.g729!
> Command 'file convert /tmp/welcome.gsm /tmp/welcome.g729' failed.
> [Dec 20 17:24:18] WARNING[2221]: translate.c:256 ast_translator_build_path:
> No translator path from g723 to alaw
> [Dec 20 17:24:18] WARNING[2221]: file.c:184 ast_writestream: Unable to
> translate to format g729, source format gsm
> 
> Even though I have the module format_g729.so. Do I need to have licensed
> G729 codec for this? or codec_g729.so?

Yes. The g729 codec module requires a per-codec-instance license. In
your case you use a single codec for encoding the audio to G.729.

BTW: if this is a file you recorded, why convert it from gsm and not
from a higher-quality format? If this is from the stanard set of
prompts: any chance it is already available as g729?

-- 
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icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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[asterisk-users] Asterisk Sip Media Call Type

2011-12-20 Thread Faraj Khasib 
Hi all,
I am trying to make a SIP Video and Audio Call, Now when I add at the Asterisk 
the video Support and the right codec whether I make Audio or Video Call from 
my clients the Call will be received as Video Call, so the problem is if I make 
from one client Audio or Video Call it will be recieved as Video Call, Can you 
plz help me try to solve this problem? Where should I change the Call Media 
Type at Asterisks
Regards
Faraj Khasib
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[asterisk-users] File Convert

2011-12-20 Thread Gopalakrishnan N 
Hi users,

I have Asterisk 1.6.2.20 in Ubuntu 10.04. I am trying to convert a gsm file
to G729 using file convert, but I am facing error as follows,

file convert /tmp/welcome.gsm /tmp/welcome.g729
Failed to convert /tmp/welcome.gsm to /tmp/welcome.g729!
Command 'file convert /tmp/welcome.gsm /tmp/welcome.g729' failed.
[Dec 20 17:24:18] WARNING[2221]: translate.c:256 ast_translator_build_path:
No translator path from g723 to alaw
[Dec 20 17:24:18] WARNING[2221]: file.c:184 ast_writestream: Unable to
translate to format g729, source format gsm

Even though I have the module format_g729.so. Do I need to have licensed
G729 codec for this? or codec_g729.so?

Kindly let me know how to convert the file.

Regards
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Re: [asterisk-users] Use different local IP for each SIP trunk

2011-12-20 Thread giovanni.v 

Il 20/12/2011 6.07, Anton Kvashenkin ha scritto:

you can add exterin= in sip.conf for each trunk


I think this can be used only in [general] section not on peers 
definition; also useful only when asterisk is behind nat. Not?


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Re: [asterisk-users] Problem with Atxfer for the calling party [SOLVED]

2011-12-20 Thread Antonio Modesto 
As explained in the posts before, this tread was solved.


Thanks.

On Tue, 2011-12-13 at 17:07 -0200, Antonio Modesto wrote:

> On Tue, 2011-12-13 at 16:35 -0200, Roberto Linck wrote:
> 
> > Hi Antonio,
> > 
> > 
> > I'd never had used extensions.ael but in extensions.conf, using
> > Macro I always set '__TRANSFER_CONTEXT' to the same context of exten
> > and it works well.
> 
> 
> Thanks, it worked, the MACRO_CONTEXT variable was empty, I've set the
> value to my extensions context and it worked fine.
> 
> 
> Thanks.
> 
> > 
> > 2011/12/13 Antonio Modesto 
> > 
> > Hello everybody,
> > 
> > I found that if i write my macro in the extensions.conf
> > (not in ael), the atxfer works well, the problem is that ael
> > uses gosub instead of the Macro() application, which doesn't
> > change the current context. Does anybody know if i can do
> > anything to solve this? I know if i rewrite all my macros in
> > the common way, it will work, but that's a lot of coding for
> > me. 
> > 
> > 
> > 
> > On Mon, 2011-12-12 at 08:57 -0200, Antonio Modesto wrote:
> > 
> > > Nothing?
> > > 
> > > 
> > > On Mon, 2011-11-21 at 16:21 -0200, Antonio Modesto wrote:
> > > 
> > > > 
> > > > 
> > > > 
> > > > 
> > > > 
> > > > Hi There,
> > > > 
> > > > I'm still having this problem, Does somebody  know
> > > > what can be happening?
> > > > 
> > > > 
> > > > Regards.
> > > > 
> > > > On Fri, 2011-11-11 at 09:40 -0200, Antonio Modesto
> > > > wrote:
> > > > 
> > > > > Hello,
> > > > > 
> > > > > The exten is the parameter passed to the macro,
> > > > > which contains the sip device name. I'll change the
> > > > > name to another less confusing.
> > > > > 
> > > > > * Alexandre, também sou brasileiro hehe, notei que
> > > > > você já escreveu um livro sobre asterisk, será que
> > > > > você poderia me ajudar com esse problema? Já tem
> > > > > alguns dias que estou na luta aqui hehe.
> > > > > 
> > > > > On Fri, 2011-11-11 at 08:45 -0200, Alexandre Keller
> > > > > wrote:
> > > > > 
> > > > > > You're using ${exten} inside your macro, you should
> > > > > > use ${EXTEN}.
> > > > > > -- 
> > > > > > Atenciosamente,
> > > > > > 
> > > > > > ALEXANDRE KELLER
> > > > > > 
> > > > > > 
> > > > > > http://twitter.com/alexandrekeller
> > > > > > http://www.facebook.com/alexandre.keller.BR
> > > > > > 
> > > > > > "Dinheiro é a consequência de um trabalho bem
> > > > > > feito e não o motivo para se fazer um bom trabalho."
> > > > > > 
> > > > > > 
> > > > > > P Antes de imprimir pense em seu compromisso com
> > > > > > o Meio Ambiente.
> > > > > > 
> > > > > > On 11/11/2011, at 08:38, Antonio Modesto wrote:
> > > > > > 
> > > > > > 
> > > > > > > On Mon, 2011-11-07 at 09:12 -0600, Danny Nicholas
> > > > > > > wrote:
> > > > > > > 
> > > > > > > > It can have to do with either the telephones
> > > > > > > > dial plan or the context in the Asterisk dial
> > > > > > > > plan combined with your features.conf settings.
> > > > > > > 
> > > > > > > 
> > > > > > > I noticed that my problem occurs when i use a
> > > > > > > macro to dial sip devices, my dialplan is like
> > > > > > > this:
> > > > > > > 
> > > > > > > - Each sip device has its own context
> > > > > > > - This context includes the outgoing call contexts
> > > > > > > that this extension can use for making calls and
> > > > > > > includes a context called "ramais", which has the
> > > > > > > dial plan to call another extensions, it uses a
> > > > > > > macro to do this.
> > > > > > > 
> > > > > > > Here is the configuration for my extension
> > > > > > > "modesto" :
> > > > > > > 
> > > > > > > # sip.conf
> > > > > > > [modesto](default_extension)
> > > > > > > username=modesto
> > > > > > > context=modesto
> > > > > > > callerid="modesto" <106>
> > > > > > > callgroup=4
> > > > > > > pickupgroup=4
> > > > > > > 
> > > > > > > # Default extension template
> > > > > > > type=friend
> > > > > > > dtmfmode=auto
> > > > > > > host=dynamic
> > > > > > > disallow=all
> > > > > > > allow=ulaw
> > > > > > > allow=alaw
> > > > > > > deny=0.0.0.0/0.0.0.0
> > > > > > > permit=192.168.1.0/255.255.255.0
> > > > > > > c

Re: [asterisk-users] Use different local IP for each SIP trunk

2011-12-20 Thread Paulo Santos 

Hello,

Douglas Mortensen wrote:

With that said, then it appears that the only way that I can have
multiple trunks setup with them is to have asterisk use a different
IP for all of the SIP & RTP traffic for each given trunk. Essentially
I would setup multiple IP addresses on my eth0 interface. Is there a
way in asterisk that I could configure it to use one local IP for the
source in all SIP/RTP traffic for 1 SIP trunk & then a different
local IP for the other SIP trunk?


It's not an asterisk configuration but rather a interface configuration.
I need something similar and I use 2 IPs on the same port. In debian,
the configuration goes like this:

auto eth0
iface eth0 inet static
address 
netmask 
network 
broadcast 
gateway 

auto eth0:0
iface eth0:0 inet static
address 
netmask 
up route add -net  netmask  gw 

And you can add more routes for other specific IPs/networks.


José Pablo Méndez Soto wrote:

May I ask why do you need different IP addresses to source calls? I
mean, its not a common practice, would like to understand the idea
behind it.


In my case, the operator installed a gateway with a dedicated line and 
it's connected to the local network, but instead of being 192.168.0.0 
it's on 10.0.0.0. So I use this 2 networks in the same NIC in the 
asterisk machine.


Best regards,
Paulo Santos


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