Asterisk + chan_ss7 2.2.0 no voice problem

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hi james,
can you tell me how to check this.


On 20 June 2013 02:34, James zhu <zhulizhong at live.com> wrote:

> hi:
> make sure MTP2 and MTP3 are active and cic is matched .
>
> Best regards,
> James.zhu
> Vega VOIP gateways<http://www.hiastar.com/index.php/2011-04-19-18-41-26/sangoma-gateways>
> , Sangoma  <http://www.hiastar.com>Asterisk card <http://www.hiastar.com>s<http://www.hiastar.com/index.php/2011-04-19-18-41-26/sangoma-e1>
> , SBC<http://www.hiastar.com/index.php/2011-04-19-18-41-26/sangoma-zte-session-border-controller>
> , NetBorder VOIP Gateway<http://www.hiastar.com/index.php/2011-04-19-18-41-26/netborder-tdm>
> , LYNC-TDM<http://www.hiastar.com/index.php/2011-04-19-18-41-26/sangoma-lync-pstn>
> , Transcoding<http://www.hiastar.com/index.php/2011-04-19-18-41-26/sangoma->
> website: www.hiastar.com
>
>
> ------------------------------
> Date: Tue, 18 Jun 2013 13:02:14 +0200
> From: amr132 at gmail.com
> To: asterisk-ss7 at lists.digium.com
> Subject: Re: [asterisk-ss7] Asterisk + chan_ss7 2.2.0 no voice problem
>
>
> Hi ferhi
>                 Many thanks for your support Can you send me your conf
> files also is there any special configuration in asterisk
>
> and really many thanks for your support also can we speak on sky-be if you
> don't mind because after downgrading its not working also
>
>
>
>
> On 18 June 2013 12:17, Amr Adel <amr132 at gmail.com> wrote:
>
> Can you send me your conf files also is there any special configuration in
> asterisk
> and really many thanks for your support also can we speak on sky-be if you
> don't mind
>
>
> On 18 June 2013 12:10, mohamed amine ferhi <ferhi.med.amine at gmail.com>wrote:
>
> Here is the link for my post about the same problem:
>
> http://lists.digium.com/pipermail/asterisk-ss7/2013-March/005068.html
>
> In my case up then the downgrade was from version 2.1.0 to 2.0.0
>
> *Ferhi Mohamed Amine**
> **Linux Engineer**
> *
> [image: Twitter] <https://twitter.com/FerhiMedAmine> [image: LinkedIn]<https://www.linkedin.com/profile/view?id=96727939>
>  [image: about.me] <http://about.me/ferhi-medamine> [image: WordPress]<http://easylinuxstuff.wordpress.com/>
> +216 25 423 600
> [image: Skype] ferhi.amine
>
>
>
> 2013/6/18 Amr Adel <amr132 at gmail.com>
>
> Thank for your reply,
> but downgrad to what version i tried 2.0.0 and its not working also.
>
>
> On 18 June 2013 11:28, mohamed amine ferhi <ferhi.med.amine at gmail.com>wrote:
>
> HI Amr, I had this problem before.
>
> The solution was to downgrade ss7 version. After that the voice was
> working fine
>
> *Ferhi Mohamed Amine**
> **Linux Engineer**
> *
> [image: Twitter] <https://twitter.com/FerhiMedAmine> [image: LinkedIn]<https://www.linkedin.com/profile/view?id=96727939>
>  [image: about.me] <http://about.me/ferhi-medamine> [image: WordPress]<http://easylinuxstuff.wordpress.com/>
> +216 25 423 600
> [image: Skype] ferhi.amine
>
>
>
> 2013/6/18 Amr Adel <amr132 at gmail.com>
>
>  Hello
>                 Kindly, I need your assist as I have installed chan_ss7
> with asterisk and got no voice on calls although asterisk is working fine
> With sip I am using
> -sangoma A104de
> -dahdi 2.6.6
> -Asterisk 1.6.2
> -chan_ss7 2.2.0
> Also I get *NOTICE[22933]: mtp.c:1993 mtp_thread_main: Full dahdi input
> buffer detected, incoming packets may have been lost on link 'l1'
> (count=65.)*
> At the beginning of an incoming call is it related to my case?
> You can find my config files attached
>
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