[Freeswitch-users] bridging two outbound calls
Jun Sun
jsun at junsun.net
Mon Apr 1 04:19:21 MSD 2013
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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>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>>
>>>>>
>>>>>
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>>>>
>>>>
>>>> _________________________________________________________________________
>>>>
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>>
>>>>
>>>>
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>>>>
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>>>>
>>>>
>>>
>>> _________________________________________________________________________
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>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>>
>>>
>>>
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>>>
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>>
>>
>
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