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Posted to user@openmeetings.apache.org by "Parker, David" <Da...@nsight.com> on 2015/09/01 15:43:41 UTC
RE: unidirectional audio via sip
Hi Maxim - have you had any thoughts on this?
________________________________
From: Maxim Solodovnik [solomax666@gmail.com]
Sent: Friday, August 28, 2015 1:54 AM
To: Openmeetings user-list
Subject: Re: unidirectional audio via sip
so to make config simplier:
you can not hear flash audio in your phone
I'll try to check it on Monday
please ping me if I'll be too silent :)
On Fri, Aug 28, 2015 at 2:39 AM, Parker, David <Da...@nsight.com>> wrote:
I apologize if this is not the proper forum for this question.
I've set up openmeetings 3 in a ubuntu guest on top of a linux host. I've also got an asterisk server on the vm along with red5sip.
*Everything* works except for one thing: when dialed in to a room through asterisk, no audio reaches the telephone handset through red5sip.
Audio *from* the phone makes it into the conference on both sides (asterisk conf bridge and openmeetings flash audio). And: if more than one phone dials in via sip to the asterisk bridge, those phones have bi-directional audio between each other.
The only problem is that no audio seems to go towards the asterisk bridge from the red5sip daemon.
I've looked through logs and haven't found anything obvious - same with sniffing. I could really use a pointer on this one.
--
WBR
Maxim aka solomax
Re: unidirectional audio via sip
Posted by Maxim Solodovnik <so...@gmail.com>.
Hello David,
it seems I'm unable to test it, sorry
hopefully someone else can help
On Fri, Sep 4, 2015 at 7:34 PM, Maxim Solodovnik <so...@gmail.com>
wrote:
> Sorry :( was unable to test it :(
> Will try to do it on weekends
> Please do not hesitate to ping me on this :)
>
> On Tue, Sep 1, 2015 at 7:43 PM, Parker, David <Da...@nsight.com>
> wrote:
>
>> Hi Maxim - have you had any thoughts on this?
>>
>>
>> ------------------------------
>> *From:* Maxim Solodovnik [solomax666@gmail.com]
>> *Sent:* Friday, August 28, 2015 1:54 AM
>> *To:* Openmeetings user-list
>> *Subject:* Re: unidirectional audio via sip
>>
>> so to make config simplier:
>>
>> you can not hear flash audio in your phone
>>
>> I'll try to check it on Monday
>> please ping me if I'll be too silent :)
>>
>> On Fri, Aug 28, 2015 at 2:39 AM, Parker, David <Da...@nsight.com>
>> wrote:
>>
>>> I apologize if this is not the proper forum for this question.
>>>
>>> I've set up openmeetings 3 in a ubuntu guest on top of a linux host.
>>> I've also got an asterisk server on the vm along with red5sip.
>>>
>>> *Everything* works except for one thing: when dialed in to a room
>>> through asterisk, no audio reaches the telephone handset through red5sip.
>>>
>>> Audio *from* the phone makes it into the conference on both sides
>>> (asterisk conf bridge and openmeetings flash audio). And: if more than one
>>> phone dials in via sip to the asterisk bridge, those phones have
>>> bi-directional audio between each other.
>>>
>>> The only problem is that no audio seems to go towards the asterisk
>>> bridge from the red5sip daemon.
>>>
>>> I've looked through logs and haven't found anything obvious - same with
>>> sniffing. I could really use a pointer on this one.
>>>
>>>
>>>
>>
>>
>> --
>> WBR
>> Maxim aka solomax
>>
>
>
>
> --
> WBR
> Maxim aka solomax
>
--
WBR
Maxim aka solomax
Re: unidirectional audio via sip
Posted by Maxim Solodovnik <so...@gmail.com>.
Sorry :( was unable to test it :(
Will try to do it on weekends
Please do not hesitate to ping me on this :)
On Tue, Sep 1, 2015 at 7:43 PM, Parker, David <Da...@nsight.com>
wrote:
> Hi Maxim - have you had any thoughts on this?
>
>
> ------------------------------
> *From:* Maxim Solodovnik [solomax666@gmail.com]
> *Sent:* Friday, August 28, 2015 1:54 AM
> *To:* Openmeetings user-list
> *Subject:* Re: unidirectional audio via sip
>
> so to make config simplier:
>
> you can not hear flash audio in your phone
>
> I'll try to check it on Monday
> please ping me if I'll be too silent :)
>
> On Fri, Aug 28, 2015 at 2:39 AM, Parker, David <Da...@nsight.com>
> wrote:
>
>> I apologize if this is not the proper forum for this question.
>>
>> I've set up openmeetings 3 in a ubuntu guest on top of a linux host.
>> I've also got an asterisk server on the vm along with red5sip.
>>
>> *Everything* works except for one thing: when dialed in to a room
>> through asterisk, no audio reaches the telephone handset through red5sip.
>>
>> Audio *from* the phone makes it into the conference on both sides
>> (asterisk conf bridge and openmeetings flash audio). And: if more than one
>> phone dials in via sip to the asterisk bridge, those phones have
>> bi-directional audio between each other.
>>
>> The only problem is that no audio seems to go towards the asterisk bridge
>> from the red5sip daemon.
>>
>> I've looked through logs and haven't found anything obvious - same with
>> sniffing. I could really use a pointer on this one.
>>
>>
>>
>
>
> --
> WBR
> Maxim aka solomax
>
--
WBR
Maxim aka solomax