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  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: 
Context: 
Exten: 
Priority: 

Regards,
Qasim


On Thu, May 16, 2013 at 3:13 PM, Muhammad Faheem wrote:

> 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  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 
> 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 
> 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] 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  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] Jitter Buffer in asterisk 1.8.11.0

2013-04-24 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 wrote:

> 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] 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()*". 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 wrote:

>  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" ;tag=as23a29r59
> To: 
> Contact: 
> 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  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 = "/ws";^M
s_proxy = "ws://<* server IP>:";^M

Change  and <* server IP> with required values.

Regards,
Qasim


On Wed, Nov 7, 2012 at 7:52 PM, Joshua Colp  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+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 
>
> Go into "Expert Mode" and disable Video support. Use the WebSocket Server
> URL for your server, like below:
>
> ws://:8088/ws
>
> Fill out the rest of the registration details as you normally would.
>
> Display Name: 
> Private Identity: 
> Public Identity: sip:@ Asterisk>
> Password: 
> Realm: 
>
> 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  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 11 WebSockets.

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


Regards,
Qasim

On Wed, Sep 5, 2012 at 1:52 AM, James Mortensen
wrote:

> qasimakhan  gmail.com  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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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
wrote:

> 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 
>
> 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 
>>>
>>> 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 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.com>provided a kind of SDK for embedded 
>>>>> system development. It worked well I
>>>>> think.
>>>>>
>>>>> --
>>>>> Sazzad Bin Kamal
>>>>>
>>>>>
>>>>> --
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>>>>>http://www.asterisk.org/hello
>>>>>
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>>>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>>
>>>>
>>>>
>>>> --
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>>>>
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>>>>
>>>
>>>
>>> --
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>>>
>>
>>
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>
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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
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 
>
> 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  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.com>provided 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

2012-09-03 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  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 
> provided a kind of SDK for embedded system 
> development. It worked well I
> think.
>
> --
> Sazzad Bin Kamal
>
>
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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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