[asterisk-ss7] chan_ss7 | Link established but no voice

[bipin singh]

Hi,
Check first any  ivr calls via ss7 link. After that try to sip to sip. If
all case are ok then go for your setting. may be sip codec issue.

On Wed, Mar 20, 2013 at 12:01 PM, Vinod Dharashive <vdharashive at gmail.com>wrote:

> Hi ,
>
> You can download SVN trunk as below
>
> svn co http://svn.netfors.com/chan_ss7/trunk/
>
> Cheers!
> Vinod Dharashive
>
>
>
> On Wed, Mar 20, 2013 at 10:50 AM, Bharat Lalcheta <
> bharatlalcheta at gmail.com> wrote:
>
>> May be i am wrong, but if its problem with codecs then you should use
>>
>> Donts use allow=all, use only single codec by adding following in sip and
>> iax
>> disallow = all
>> allow=alaw
>> On Wed, Mar 20, 2013 at 3:47 AM, mohamed amine ferhi <
>> ferhi.med.amine at gmail.com> wrote:
>>
>>> Thanks for your responses.
>>>
>>> @Bharat I already made this change : allow=all => No changes
>>>
>>> @VInod the chan_ss7 version that I use is from here
>>> http://www.netfors.com/chan_ss7_free , any other URL suggested?
>>> The use of asterisk 1.6 is not an option for me unfortunatly.
>>>
>>>
>>>  2013/3/19 Vinod Dharashive <vdharashive at gmail.com>
>>>
>>>> Hi
>>>>
>>>> Download the chan_ss7 from trunk and compiles with asterisk. I have
>>>> tested this with asterisk 1.6. It worked for me.
>>>>
>>>>  Cheers!
>>>> Vinod Dharashive
>>>> Sent from BlackBerry® on Airtel
>>>>
>>>> -----Original Message-----
>>>> From: Bharat Lalcheta <bharatlalcheta at gmail.com>
>>>> Sender: asterisk-ss7-bounces at lists.digium.com
>>>> Date: Wed, 20 Mar 2013 00:12:25
>>>> To: <asterisk-ss7 at lists.digium.com>
>>>> Reply-To: asterisk-ss7 at lists.digium.com
>>>> Subject: Re: [asterisk-ss7] chan_ss7 | Link established but no voice
>>>>
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>>>
>>>
>>>
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>>
>>
>>
>> --
>> Bharat Lalcheta
>> --
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>
>
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-- 
BIPIN RAGHUVANSHI
OPERATION HEAD
ASTERISK (DEVELOPMENT AND RESEARCH)
WWW.EHORIZONS.IN
bipinraghuvanshi at gmail.com
bipin.singh at ehorizons.in
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