[Freeswitch-users] bridging two outbound calls
Jun Sun
jsun at junsun.net
Tue Apr 2 01:08:10 MSD 2013
Thanks, Steven. I think it's a very reasonable suspect, although it is
puzzling to see conferencing works with two outbound calls.
I've set freeswitch/ec2 up according to the wiki page,
http://wiki.freeswitch.org/wiki/Amazon_ec2. I will double check.
Cheers.
Jun
On Apr 1, 2013 12:54 PM, "Steven Ayre" <steveayre at gmail.com> wrote:
> AFAIK on AWS EC2 the server runs behind NAT. My guess is your FS install
> isn't configured correctly to be behind NAT, so the server wasn't able to
> send media to FS correctly. But it can send correctly to itself
> (bypass_media).
>
>
> On 1 April 2013 20:27, Jun Sun <jsun at junsun.net> wrote:
>
>>
>> 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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>> >>
>> >>
>> >>
>> >> !DSPAM:5157d35132761492122807!
>> >>
>> >
>> >
>>
>>
>> _________________________________________________________________________
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>>
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>
>
> _________________________________________________________________________
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> http://www.freeswitchsolutions.com
>
>
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