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
>>>>>
>>>>> --
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>>>>
>>>>
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>>>
>>
>
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