bridging two outbound calls
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@X.X.X.X:5060 & bridge(sofia/internal/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
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 @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> 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@X.X.X.X:5060 & bridge(sofia/internal/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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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
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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
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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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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
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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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FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
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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
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_________________________________________________________________________
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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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_________________________________________________________________________
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FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
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By default that parameter is the destination_number to enter the dialplan with (since its an outbound call the channel wouldn't otherwise have one). The & prefix changes that behaviour to tell FS that what follows is directly invoking an application on the channel. Steve On 31 Mar 2013, at 07:30, Jun Sun <jsun@junsun.net> wrote:
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
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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
_________________________________________________________________________
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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_________________________________________________________________________ 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
<junk.patch> _________________________________________________________________________ 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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If your question is why can't FS allow a space, its a) specifically a prefix when evaluating that argument and b) there can be further arguments to the originate command after that one, additional spaces would make it problematic figuring out why arg is which. Steve On 31 Mar 2013, at 07:30, Jun Sun <jsun@junsun.net> wrote:
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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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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<junk.patch> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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I think I understand the logic now. It seems "&" is used as some kind of escape symbol while logically it implies "and consequently ...". I think it is the later logical meaning which lures newbies into using spaces. A simple patch as I attached offers the syntactical sugar without complicating the matter. The patch simply moves all spaces following the "&" to before it. Unless "&" can mean something else in arg list, the patch should be safe and practical. Anyway, I don't think I will forget about the "&" space issue anymore. :) Cheers. Jun On 3/31/2013 10:40 AM, Steven Ayre wrote:
If your question is why can't FS allow a space, its a) specifically a prefix when evaluating that argument and b) there can be further arguments to the originate command after that one, additional spaces would make it problematic figuring out why arg is which.
Steve
On 31 Mar 2013, at 07:30, Jun Sun <jsun@junsun.net> wrote:
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 http://www.freeswitch.org
!DSPAM:5156810f32761697518053!
_________________________________________________________________________ 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
<junk.patch> _________________________________________________________________________ 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
participants (4)
-
Jun Sun
-
Peter Olsson
-
Steven Ayre
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Zoltan Medveczky