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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-ss7/attachments/20130618/3584dc08/attachment.htm>
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