hello:Run: ss7 link status under Asterisk CLI, and check with service the first cic.
Best regards,
James.zhuConnect with Sangoma
website: www.hiastar.com
Date: Thu, 20 Jun 2013 10:41:43 +0200
From: amr132@xxxxxxxxx
To: asterisk-ss7 at lists.digium.com
Subject: Re: Asterisk + chan_ss7 2.2.0 no voice problem
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, Sangoma Asterisk cards, SBC, NetBorder VOIP Gateway, LYNC-TDM, Transcoding
website: 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
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