Re: [asterisk-users] Not able to build the chan_sip.c module

2013-05-27 Thread qasimak...@gmail.com
It depends on chan_local see if that is enabled or not.

Regards,
Qasim


On Mon, May 27, 2013 at 11:56 AM, upendra uppi...@gmail.com wrote:

 Hi,

 i am trying to install asterisk newer version on the Elastix machine, but
 while installing the chan_sip,c module is not building while make. when i
 see  in make menuselect options it showing XXX -- extended , please let
 me know how to enable it and make build chan_sip module.



 --
 Upendra


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Re: [asterisk-users] Call Transfer question

2013-05-16 Thread qasimak...@gmail.com
Hi faheem,

You can do this:

ACTION: Redirect
Channel: Channel ID
Context: Context
Exten: Exten
Priority: Priority

Regards,
Qasim


On Thu, May 16, 2013 at 3:13 PM, Muhammad Faheem faheem2...@gmail.comwrote:

 Hi,
 is possible that two sip extensions: user-1 and user-2 are connected and I
 want to transfer the call from user-1 to a third user user-3.
 I know it is possible through feature keys mapping in features.conf, but I
 want to do this through AMI or Asterisk CLI Commands?

 Please suggest if possible?

 Thank you!
 Muhammad Faheem

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Re: [asterisk-users] Sip and the media path

2013-04-27 Thread qasimak...@gmail.com
Hi David,

Direct media should work either way. if your phones are behind NAT you will
also require the NAT option enabled in asterisk, How ever  the tricky part
in all this is that you wont be able to acurately keep track of calls on
these phones. If or any unforeseen reason the phone goes offline or you
dont recieve BYE signal, asterisk wont be able to know that the call has
ended. So if call information is critical for you then byepassmedia is not
recomended for you.

Regards,
Qasim


On Thu, Apr 25, 2013 at 8:48 PM, David Wessell da...@ringfree.biz wrote:

  Kevin,

  Thanks for the info. Clarification. The asterisk server is NOT on the
 same LAN as the phones. The asterisk server is in a datacenter only
 accessible via WAN.

  However, all of the phones are in side of the same LAN. Will directmedia
 still function that way?

  Thanks
 David

   From: Kevin Larsen kevin.lar...@pioneerballoon.com
 Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Date: Thursday, April 25, 2013 9:16 AM

 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] Sip and the media path

  You will want to look at the directmedia option. You will want all the
 phones on the same lan as the Asterisk server to be directmedia=yes and the
 ones on the wan to be directmedia=no. Then, internal calls will send the
 media between themselves without involving Asterisk, but ones outside on
 the wan will be forced to talk directly to the Asterisk server for
 everything. You might also want to look at the nonat option of directmedia.

 Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208



 From:David Wessell da...@ringfree.biz
 To:Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com,
 Date:04/25/2013 07:33 AM
 Subject:[asterisk-users] Sip and the media path
 Sent by:asterisk-users-boun...@lists.digium.com
 --



 We're running asterisk 1.8 in the DC on a public IP address.

 Connecting to it are about 200 phones behind a LAN in a remote location.

 Is there a way to reliably keep asterisk out of the media stream on
 internal calls inside that LAN? All phones are Polycom Soundpoint phones.

 Asterisk would say in the media stream for any calls that traverse from
 LAN to WAN. However it would step out for LAN to LAN calls.

 Thanks
 David
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Re: [asterisk-users] Jitter Buffer in asterisk 1.8.11.0

2013-04-25 Thread qasimak...@gmail.com
Search jitter in sample sip.conf. Everything is well documented there.


Regards,
Qasim


On Tue, Apr 23, 2013 at 3:03 PM, Muhammad Yousuf muyous...@gmail.comwrote:

 I am using asterisk as SIP/GSM  gateway. I have 2 gsm cards installed in
 server. I am having some issue in audio quality. I want to enable jitter
 buffer on asterisk but don't know, how to do. Any one can help me.

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Re: [asterisk-users] Asterisk 1.8 and 11

2013-04-25 Thread qasimak...@gmail.com
Read up on new features and changelog of asterisk 11 you'll find the
changes there.

Regards,
Qasim


On Thu, Apr 25, 2013 at 3:32 PM, bilal ghayyad bilmar...@yahoo.com wrote:

 Hello;

 How I can compare between Asterisk 1.8 and 11 with reference to the
 following points:

 1) SMS.
 2) gtalk and other social media.
 3) GUI.
 4) Any main difference?

 Regards
 Bilal

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Re: [asterisk-users] set time zone in sip debug logs

2013-02-25 Thread qasimak...@gmail.com
Hi Kamlesh,

Asterisk give you very less control over SIP messaging. You can how ever
add/remove/modify SIP headers from initial invite only. To modify a sip
header you can use asterisk function *SIP_HEADER(name)*. If you want to
permanently change date why not change system date/time?

Regards,
-Qasim

On Tue, Feb 26, 2013 at 11:13 AM, Kamlesh Kumar kamlesh_...@hotmail.comwrote:

  Hello,

 Please suggest the way to change the time zone in below sip debug logs.

 INVITE sip:xxx...@xxx.xxx.xxx.xxx:5060 SIP/2.0
 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK7bbd9;rport
 Max-Forwards: 70
 From: xx sip:xxx...@xxx.xxx.xxx.xxx;tag=as23a29r59
 To: sip:xxx...@xxx.xxx.xxx.xxx:5060
 Contact: sip:xxx...@xxx.xxx.xxx.xxx
 Call-ID: 2f17b2103ea4792d571e2dce7e14b...@xxx.xxx.xxx.xxx
 CSeq: 102 INVITE
 User-Agent: Asterisk PBX 1.6.2.9
 *Date: Tue, 26 Feb 2013 04:54:29 GMT*
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
 Supported: replaces, timer
 Content-Type: application/sdp
 Content-Length: 444

 Thanks,
 Kamlesh

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Re: [asterisk-users] Asterisk SIP authenticate using Radius / LDAP

2012-11-11 Thread qasimak...@gmail.com
You can use Radius Agi developed by PortaOne from following link.

http://www.voip-info.org/wiki/view/PortaOne+Radius+auth

Regards,
Qasim


On Mon, Nov 12, 2012 at 11:24 AM, Samira Hosseini samiramhosse...@yahoo.com
 wrote:


 Hi all,
 based on the following link, I am going to authenticate SIP asterisk users
 via Radius client that is installed on my Asterisk then the radius client
 connect to asterisk using the radius and ldap:

 https://who.rocq.inria.fr/Philippe.Sultan/Asterisk/asterisk_sip_external_authentication.html#AEN237

 So I want to know for implementing the mentioned authentication method I
 need to use the patched asterisk as follow :

 https://issues.asterisk.org/jira/browse/ASTERISK-5278?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel

 Thanks.

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Re: [asterisk-users] Can you help me to use SIPML5 with Asterisk ?

2012-11-08 Thread qasimak...@gmail.com
You can also hardcode these values in call.htm find below lines:

i_port = 4062 + (((new Date().getTime()) % 5) * 1000);^M
s_proxy = sipml5.org;^M

and change them to

i_port = port/ws;^M
s_proxy = ws://* server IP:;^M

Change port and * server IP with required values.

Regards,
Qasim


On Wed, Nov 7, 2012 at 7:52 PM, Joshua Colp jc...@digium.com wrote:

 Lionel BEAUDOIN wrote:

 Hello,


 Hola,

  I saw your email in a forum message, can you help me, I try to use
 SIPML5 with an Asterisk 11 server ?

 My Asterisk server is installed on a Debian server.
 I have download all the sources from sipml5.org


 Please ensure you have followed the instructions at
 https://wiki.asterisk.org/**wiki/display/AST/Asterisk+**WebRTC+Supporthttps://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Supportto
  set up the Asterisk side of things for WebSocket.

  I have modifiied call.htm to target the requests on my server.

 - If I use the port 5060, I can register but I cant emet calls
 - If I use the port 8088, I can't register.

 I think it's because I don't use the WS protocol but when I watch the
 request on the 8088 port with tcpdump, I see that transport is UDP.

 How can I define a registring session with WS transport in the call.htm
 file ?


 You don't need to use your own copy of sipml5. Point a suitable browser to
 the following URL:

 http://sipml5.org/call.htm?**svn=9 http://sipml5.org/call.htm?svn=9

 Go into Expert Mode and disable Video support. Use the WebSocket Server
 URL for your server, like below:

 ws://hostname or IP address of Asterisk:8088/ws

 Fill out the rest of the registration details as you normally would.

 Display Name: account name in sip.conf
 Private Identity: account name in sip.conf
 Public Identity: sip:account name in sip.conf@hostname or IP address of
 Asterisk
 Password: password configured in sip.conf
 Realm: hostname or IP address of Asterisk

 In the future please send emails of this type to the asterisk-users
 mailing list so that everyone can see the conversation and learn. I've
 copied my reply to it.

 Cheers,

 --
 Joshua Colp
 Digium, Inc. | Senior Software Developer
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 Check us out at:  www.digium.com   www.asterisk.org

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Re: [asterisk-users] Uprading to Asterisk 11 issues

2012-11-01 Thread qasimak...@gmail.com
SVN Version is always development version. Try downloading a stable tarball
archive from http://www.asterisk.org/downloads.

Regards,
Qasim


On Thu, Nov 1, 2012 at 6:52 PM, Thomas Thomas debussy...@gmail.com wrote:

 Hello,

 I installed Asterisk 11 via the following command
 * svn co http://svn.asterisk.org/svn/asterisk/branches/11*
 (as written in asteriskdocs.org

 http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/Installing_id240883.html
 )

 But it seems that I have a development version instead of a stable release:
 * core show version*
 *Asterisk SVN-branch-11-r375559 built by user @ user-MS-6580 on a i686
 running Linux on 2012-11-01 13:05:50 UTC*

 Did I do something wrong ?

 Secondly, my logs from Verbose() application are no longer shown. I know
 that the logging system has been changed but what shall I do to see my logs
 (through asterisk -rv) ?


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Re: [asterisk-users] asterisk on arm

2012-09-04 Thread qasimak...@gmail.com
How about stripping it down to bare minimum's?

Regards,
Qasim

On Tue, Sep 4, 2012 at 1:40 PM, Stefan at WPF
stefan.at@googlemail.comwrote:

 I had problems on the Raspberry, like stuttering calls (just in between
 the calls), maybe it was because of call recording but I would expect one
 call recording to be not too much. However I used the packages from the
 repo, maybe compiling it yourself and leaving out unnecessary stuff gives
 beter performance.

 2012/9/4 qasimak...@gmail.com qasimak...@gmail.com

 I have tried it on raspberrypi, Although i didn't do any tests but looks
 promising. Should be able to handle calls in figure of two digits easily,
 The final answer always depends on your configuration.

 Regards,
 Qasim

 On Tue, Sep 4, 2012 at 8:52 AM, Sazzad sazzadbinka...@gmail.com wrote:

 has anyone tried asterisk on arm processors?


 Yeah, I used Asterisk on ARM Cortex8 processors for an embedded board,
 Zoom OMAP35x.


 how is the performance?


 Well, it was an experimental setup to create a custom Asterisk channel.
 So I can't tell you about the performance in production deployment.


 have encountered problems in the compilation?


 Yes, occasionally. Since cross compiler was used there were several
 issues, like setting the appropriate environment variables, customizing
 Makefiles and such. But Timesys' Factory 
 https://linuxlink.timesys.comprovided a kind of SDK for embedded system 
 development. It worked well I
 think.

 --
 Sazzad Bin Kamal


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Re: [asterisk-users] asterisk on arm

2012-09-04 Thread qasimak...@gmail.com
Intel Atom has a whole lot of processing power as compared to RaspberryPi.
Asterisk has modular approach do it dosen't make any difference if you
compile only necessary stuff. You can exclude modules in runtime also and
it will serve you the same purpose.

Regards,
Qasim



On Tue, Sep 4, 2012 at 2:45 PM, Stefan at WPF
stefan.at@googlemail.comwrote:

 Guess this is what most people are doing by compiling only necessary
 stuff. Personally I find this is to much fidling and contraproductive. Just
 bought a small Atom System. Hope it works better.

 2012/9/4 qasimak...@gmail.com qasimak...@gmail.com

 How about stripping it down to bare minimum's?

 Regards,
 Qasim


 On Tue, Sep 4, 2012 at 1:40 PM, Stefan at WPF 
 stefan.at@googlemail.com wrote:

 I had problems on the Raspberry, like stuttering calls (just in between
 the calls), maybe it was because of call recording but I would expect one
 call recording to be not too much. However I used the packages from the
 repo, maybe compiling it yourself and leaving out unnecessary stuff gives
 beter performance.

 2012/9/4 qasimak...@gmail.com qasimak...@gmail.com

 I have tried it on raspberrypi, Although i didn't do any tests but looks
 promising. Should be able to handle calls in figure of two digits easily,
 The final answer always depends on your configuration.

 Regards,
 Qasim

 On Tue, Sep 4, 2012 at 8:52 AM, Sazzad sazzadbinka...@gmail.comwrote:

 has anyone tried asterisk on arm processors?


 Yeah, I used Asterisk on ARM Cortex8 processors for an embedded board,
 Zoom OMAP35x.


 how is the performance?


 Well, it was an experimental setup to create a custom Asterisk
 channel. So I can't tell you about the performance in production
 deployment.


 have encountered problems in the compilation?


 Yes, occasionally. Since cross compiler was used there were several
 issues, like setting the appropriate environment variables, customizing
 Makefiles and such. But Timesys' 
 Factoryhttps://linuxlink.timesys.comprovided a kind of SDK for embedded 
 system development. It worked well I
 think.

 --
 Sazzad Bin Kamal


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Re: [asterisk-users] Asterisk 11 WebSockets.

2012-09-04 Thread qasimak...@gmail.com
Thanks :).


Regards,
Qasim

On Wed, Sep 5, 2012 at 1:52 AM, James Mortensen
james.morten...@a-cti.comwrote:

 qasimakhan at gmail.com qasimakhan at gmail.com writes:

 
 
  Hi,I was testing with newly introduced websocket support in asterisk 11.
 I
 have successfully implemented everything except when i try to make a call
 i get
 no audio. I have tried both SipML5 as well as SIP-JS as clients. the call
 get
 connected but i never hear any audio stream. I however get the following
 warning
 
  WARNING[2626][C-]: chan_sip.c:9686 process_sdp: Ignoring video
 stream
 offer because port number is zero
 
 
  When i turn rtp debug on i can see RTP getting through.
 
  CLI Output:http://pastebin.pk/16sip.conf:
 http://pastebin.pk/17http.conf:
 http://pastebin.pk/19extensions.conf:
 http://pastebin.pk/20Regards,Qasim
 
 
  --
  _

 According to the Asterisk developers, this is an issue in the hands of the
 browser developers. Here is the wiki page on the Asterisk 11 SIP over
 WebSockets:
 https://wiki.asterisk.org/wiki/display/~jcolp/Asterisk+WebRTC+Support

 At this time, no media is flowing.

 James


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[asterisk-users] Asterisk 11 WebSockets.

2012-09-03 Thread qasimak...@gmail.com
Hi,

I was testing with newly introduced websocket support in asterisk 11. I
have successfully implemented everything except when i try to make a call i
get no audio. I have tried both SipML5 as well as SIP-JS as clients. the
call get connected but i never hear any audio stream. I however get the
following warning

WARNING[2626][C-]: *chan_sip.c:9686 process_sdp:* Ignoring video
 stream offer because port number is zero


When i turn rtp debug on i can see RTP getting through.

*CLI Output*:http://pastebin.pk/16

*sip.conf*:http://pastebin.pk/17

*http.conf*:   http://pastebin.pk/19

*extensions.conf*: http://pastebin.pk/20

Regards,
Qasim
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Re: [asterisk-users] asterisk on arm

2012-09-03 Thread qasimak...@gmail.com
I have tried it on raspberrypi, Although i didn't do any tests but looks
promising. Should be able to handle calls in figure of two digits easily,
The final answer always depends on your configuration.

Regards,
Qasim

On Tue, Sep 4, 2012 at 8:52 AM, Sazzad sazzadbinka...@gmail.com wrote:

 has anyone tried asterisk on arm processors?


 Yeah, I used Asterisk on ARM Cortex8 processors for an embedded board,
 Zoom OMAP35x.


 how is the performance?


 Well, it was an experimental setup to create a custom Asterisk channel. So
 I can't tell you about the performance in production deployment.


 have encountered problems in the compilation?


 Yes, occasionally. Since cross compiler was used there were several
 issues, like setting the appropriate environment variables, customizing
 Makefiles and such. But Timesys' Factory 
 https://linuxlink.timesys.comprovided a kind of SDK for embedded system 
 development. It worked well I
 think.

 --
 Sazzad Bin Kamal


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