hi:make sure MTP2 and MTP3 are active and cic is matched . Best regards, James.zhuVega VOIP gateways, Sangoma Asterisk cards, SBC, NetBorder VOIP Gateway, LYNC-TDM, Transcodingwebsite: www.hiastar.com Date: Tue, 18 Jun 2013 13:02:14 +0200 From: amr132@xxxxxxxxx To: asterisk-ss7 at lists.digium.com Subject: Re: 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 +216 25 423 600 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 +216 25 423 600 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-ss7/attachments/20130620/2983ebbb/attachment.htm>