Re: [Freeswitch-users] bridging two outbound calls
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@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@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@junsun.net <mailto:jsun@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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_________________________________________________________________________
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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Hmm, crazy! After enabling "bypass_media" I can start to hear the conversation from both ends. originate {bypass_media=true}sofia/internal/15107079642@216.xxx.xxx.11:5060 &bridge(sofia/internal/15102991921@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.
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@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@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@junsun.net <mailto:jsun@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
_________________________________________________________________________
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
_________________________________________________________________________
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
_________________________________________________________________________
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
_________________________________________________________________________
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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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@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@216.xxx.xxx.11:5060 &bridge(sofia/internal/15102991921@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.
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@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@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@junsun.net > <mailto:jsun@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 > > >
> > > Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org <mailto:consulting@freeswitch.org> > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > <mailto:FreeSWITCH-users@lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > >
> > > Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
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FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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http://lists.freeswitch.org/mailman/options/freeswitch-users
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
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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@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@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@216.xxx.xxx.11:5060 &bridge(sofia/internal/15102991921@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.
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@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@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@junsun.net >> <mailto:jsun@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 >> >> >>
>> >> >> Professional FreeSWITCH Consulting Services: >> consulting@freeswitch.org <mailto:consulting@freeswitch.org> >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org >> <mailto:FreeSWITCH-users@lists.freeswitch.org> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >> http://www.freeswitch.org >> >> >> >> >>
>> >> >> Professional FreeSWITCH Consulting Services: >> consulting@freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users@lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >> http://www.freeswitch.org > > >
> > > Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > >
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Mystery solved! I did not open UDP port 16384-32768. I thought I have done that, but apparently took it away (or created a new instance somewhere along the way). Re-opening the ports makes the bridging call happy. Thanks for your help, Steven, Peter. Apparently conferencing outbound calls only generates outbound RTP traffic which can still work even without those ports open. Cheers. Jun On 4/1/2013 2:08 PM, Jun Sun wrote:
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@gmail.com <mailto:steveayre@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@junsun.net <mailto:jsun@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 <tel:15107079642>@216.xxx.xxx.11:5060 &bridge(sofia/internal/15102991921 <tel:15102991921>@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@junsun.net <mailto:jsun@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@junsun.net <mailto:jsun@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> >>>>>> <tel: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@junsun.net <mailto:jsun@junsun.net> >>>>>> <mailto:jsun@junsun.net <mailto:jsun@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> >>>>>> <tel:15102991912 <tel:15102991912>>@X.X.X.X:5060 & >>>>>> bridge(sofia/internal/18005551212 <tel:18005551212> >>>>>> <tel: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 >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> >>>>>> >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting@freeswitch.org <mailto:consulting@freeswitch.org> <mailto:consulting@freeswitch.org <mailto:consulting@freeswitch.org>> >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>> http://www.cudatel.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> >>>>>> <mailto:FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org>> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> >>>>>> >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> >>>>>> >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting@freeswitch.org <mailto:consulting@freeswitch.org> >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>>> http://www.cudatel.com >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> >>>>>> >>>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> >>>>> >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting@freeswitch.org <mailto:consulting@freeswitch.org> >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>> http://www.cudatel.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>> >>>> _________________________________________________________________________ >>>> >>>> Professional FreeSWITCH Consulting Services: >>>> consulting@freeswitch.org <mailto:consulting@freeswitch.org> >>>> http://www.freeswitchsolutions.com >>>> >>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>> http://www.cudatel.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> >>>> http://www.freeswitch.org >>>> >>> >>> >> >> >> >> !DSPAM:5157d35132761492122807! >> > >
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FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
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participants (2)
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Jun Sun
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Steven Ayre