[asterisk-ss7] Asterisk + chan_ss7 2.2.0 no voice problem

[Aashiq bhatti]

Dear Amr,

In ss7.conf you have configured two ss7 links on link-l1 and link-l2.
for l3 and l4 if there is no signaling you need to use all channels like
1-31.
firstcic for l2 , l3 and l4 should 33, 65 and 97 respectively.


[link-l2]
linkset => siuc
channels => 1-15,17-31
schannel => 16
firstcic => 33
sltm => no
sls => 0

[link-l3]
linkset => siuc
channels => 1-31
schannel =>
firstcic => 65
sltm => no


[link-l4]
linkset => siuc
channels => 1-31
schannel =>
firstcic => 97
sltm => no


Regards,
-Aashiqbhatti

On Tue, Jun 18, 2013 at 12:46 PM, Amr Adel <amr132 at gmail.com> wrote:

> 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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