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 > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-ss7 mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-ss7 > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-ss7 mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-ss7 > > > > > > > -- _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > asterisk-ss7 mailing list To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-ss7 > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-ss7 mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-ss7 > -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-ss7/attachments/20130620/0de36660/attachment-0001.htm>
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