[Freeswitch-users] bridging two outbound calls

Jun Sun jsun at junsun.net
Mon Apr 1 23:27:34 MSD 2013


Hmm, crazy! After enabling "bypass_media" I can start to hear the 
conversation from both ends.

originate 
{bypass_media=true}sofia/internal/15107079642 at 216.xxx.xxx.11:5060 
&bridge(sofia/internal/15102991921 at 216.xxx.xxx.11:5060)

This is even better for my callback purpose. However, I really thought 
without bypassing media would be easier to get work first. Still feel 
strange why it would not work ...

Cheers.

Jun

On 3/31/2013 5:19 PM, Jun Sun wrote:
>
> Thanks for the pastebin suggestion. I have attached pastebin URL for the
> output from fs_cli console (debug level 7), from initiating the command
> to both parties picking up the phone.
>
> http://pastebin.com/tZewdvri
>
> I did not touch bypass_media in any way. I assume it should be off. (And
> this is a fairly standard system, other than that it is hosted on AWS
> EC2 machine)
>
> Since I"m using the same SIP trunk, even if bypass_media is set to true,
> it probably should still work.
>
> Thanks in advance. I can get a pcap file if that will be helpful.
>
> Jun
>
> On 3/31/2013 12:18 AM, Peter Olsson wrote:
>> Yes, it's there to separate if the second parameter is for en
>> extension (no '&' prefix) or an application (using '&' prefix). Read
>> the full description here:
>> http://wiki.freeswitch.org/wiki/Mod_commands#originate. If you would
>> allow a space in between it would make '&' the second parameter, and
>> the actual string the third - which is not correct. This is just
>> normal syntax rules for how to parse arguments.
>>
>> About the audio issue - make sure to get debug logs, and a pcap when
>> making the call, post to pastebin. This is usually a NAT or codec
>> issue. SInce you said it was working if calling a conference, another
>> thing to check might be if you're settings bypass_media or not.
>>
>> /Peter
>> ________________________________________
>> Från: Jun Sun [jsun at junsun.net]
>> Skickat: den 31 mars 2013 08:30
>> Till: FreeSWITCH Users Help
>> Cc: Peter Olsson
>> Ämne: Re: [Freeswitch-users] bridging two outbound calls
>>
>> Is there any specific reason why "&" must be immediately followed by the
>> application name? I found this restriction pretty annoying (at least for
>> newbies ;0)
>>
>> A simple patch would easily fix this. See one attached. Any takers?
>>
>> I'm still searching for the solution for no sound after bridging. Would
>> appreciate any pointers. (Again, this really has to be one of the
>> simplest cases ... why it has been so hard?)
>>
>> Cheers.
>>
>> Jun
>>
>> On 3/30/2013 8:36 AM, Jun Sun wrote:
>>>
>>> Oh, my god. That is it! After removing the extra space between "&" and
>>> "bridge", the second number now gets dialed.
>>>
>>> However, I cannot hear each other between these two phones. I think the
>>> signaling part is working, because hanging up one end will cause the
>>> other end hung up. However, the media is not flowing through.
>>>
>>> This must be a simple mistake. Any pointers?
>>>
>>> BTW, using conference() app works, i.e., both ends can connect and talk.
>>> So my system should be in general healthy state.
>>>
>>> Cheers.
>>>
>>> Jun
>>>
>>> On 3/29/2013 11:38 PM, Peter Olsson wrote:
>>>> It looks like you have a space between & and bridge? It might be my
>>>> email reader though. Anyway, it must be set like this: &bridge().
>>>>
>>>> Also, I'm not sure about the tel: stuff, if you can set it that way,
>>>> especially since there is a whitespace in between as well.
>>>>
>>>> /Peter
>>>>
>>>> 30 mar 2013 kl. 07:27 skrev "Jun Sun" <jsun at junsun.net>:
>>>>
>>>>>
>>>>> Yes, I tried. I can reverse the positions of those two numbers and
>>>>> always the first number gets called and the second number gets
>>>>> nothing.
>>>>>
>>>>> I start to wonder whether I'm using bridge() application wrongly.
>>>>> Can it
>>>>> dial out directly to a PSTN number via sofia?
>>>>>
>>>>> I was also fumbling with two orignate commands (followed by park()
>>>>> application) and uuid_bridge to connect. No failures on console,
>>>>> but the
>>>>> two lines are not talking.
>>>>>
>>>>> Thanks.
>>>>>
>>>>> Jun
>>>>>
>>>>> On 3/29/2013 6:25 PM, Zoltan Medveczky wrote:
>>>>>> You probably already tried this, but are you able to place a call to
>>>>>> your party B endpoint at all (i.e.
>>>>>> originate(sofia/internal/18005551212
>>>>>> <tel:18005551212>@X.X.X.X:5060)?
>>>>>>
>>>>>> If that's failing, I'd say there's probably something wrong with your
>>>>>> dial string.  Perhaps your SIP UA is not bound to port 5060?
>>>>>>
>>>>>> On Fri, Mar 29, 2013 at 5:45 PM, Jun Sun <jsun at junsun.net
>>>>>> <mailto:jsun at junsun.net>> wrote:
>>>>>>
>>>>>>      I feel really stupid. This has to be one of the simplest
>>>>>> cases in
>>>>>>      freeswitch, but I can't seem to get it work.
>>>>>>
>>>>>>      My goal is to originate two outbound calls and bridge them
>>>>>> together, a
>>>>>>      typical callback use case. I like to to do it from socket
>>>>>> api/fs_cli.
>>>>>>
>>>>>>      Here is what I typed in fs_cli:
>>>>>>
>>>>>>      originate sofia/internal/15102991912
>>>>>> <tel:15102991912>@X.X.X.X:5060  &
>>>>>>      bridge(sofia/internal/18005551212
>>>>>> <tel:18005551212>@X.X.X.X:5060)
>>>>>>
>>>>>>      The first leg is initiated and answered successfully, but the
>>>>>> second leg
>>>>>>      never happens. From the console I don't see any action done by
>>>>>> FS to do
>>>>>>      the bridging part.
>>>>>>
>>>>>>      Any idea? Thanks in advance.
>>>>>>
>>>>>>      Cheers.
>>>>>>
>>>>>>      Jun
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>>
>>>>>>
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>>>>>>
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>>>>>>      
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>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>>
>>>>>>
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>>>>>>
>>>>>> 
>>>>>> 
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>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>>
>>>>>
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>>>>>
>>>>> 
>>>>> 
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>>>>>
>>>>
>>>> _________________________________________________________________________
>>>>
>>>> Professional FreeSWITCH Consulting Services:
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>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
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>>>
>>>
>>
>>
>>
>> !DSPAM:5157d35132761492122807!
>>
>
>




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