What happend to variable_* in socket_outbound?
Woof! I used to get lots of variable_* lines when using socket_outbound. They have disappeared. Is there something I need to configure to get them back? --Woof!
they are still there, at least if you register to myevents or all events. i use the default settings from fs and get plenty of them. 2008/10/28 Andy Spitzer <woof@nortel.com>:
Woof!
I used to get lots of variable_* lines when using socket_outbound. They have disappeared. Is there something I need to configure to get them back?
--Woof!
_______________________________________________ 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
We are using freeswitch as a SIP proxy, where we are letting people register with freeswitch, and in-turn we do the SIP registration for them to "arbitrary" sip servers (as requested by users) - each user gets his own sip gateway in the freeswitch configuration. Then they can make outgoing calls and calls are routed through their specific SIP gateway. Now the problem is that when a call is received from one of these SIP registrations, it hits the public.xml where I can't seem to figure out how to get the SIP gateway information from which it came in. The SIP gateway name actually contains the information where it should be routed to. Any ideas on how to approach this problem? Question - is it possible to do it in the dialplan (dynamic) or do we have to write an application to do this mapping? -Saurabh _________________________________________________________________ When your life is on the go—take your life with you. http://clk.atdmt.com/MRT/go/115298558/direct/01/
On Wed, Oct 29, 2008 at 2:52 PM, Saurabh Aggarwal < saurabh_aggarwal@hotmail.com> wrote:
We are using freeswitch as a SIP proxy, where we are letting people register with freeswitch, and in-turn we do the SIP registration for them to "arbitrary" sip servers (as requested by users) - each user gets his own sip gateway in the freeswitch configuration. Then they can make outgoing calls and calls are routed through their specific SIP gateway.
Now the problem is that when a call is received from one of these SIP registrations, it hits the public.xml where I can't seem to figure out how to get the SIP gateway information from which it came in. The SIP gateway name actually contains the information where it should be routed to. Any ideas on how to approach this problem?
Question - is it possible to do it in the dialplan (dynamic) or do we have to write an application to do this mapping?
-Saurabh
have you looked at this example http://wiki.freeswitch.org/wiki/Getting_Started_Guide#Receiving_an_inbound_c... ram
Yes, but there is no DID in my system for incoming calls. I have users dynamically registering gateways, and calls coming in to SIP ids that they have used to register. -Saurabh Date: Wed, 29 Oct 2008 15:12:28 +0530From: talk2ram@gmail.comTo: freeswitch-users@lists.freeswitch.orgSubject: Re: [Freeswitch-users] SIP incoming call routing On Wed, Oct 29, 2008 at 2:52 PM, Saurabh Aggarwal <saurabh_aggarwal@hotmail.com> wrote: We are using freeswitch as a SIP proxy, where we are letting people register with freeswitch, and in-turn we do the SIP registration for them to "arbitrary" sip servers (as requested by users) - each user gets his own sip gateway in the freeswitch configuration. Then they can make outgoing calls and calls are routed through their specific SIP gateway. Now the problem is that when a call is received from one of these SIP registrations, it hits the public.xml where I can't seem to figure out how to get the SIP gateway information from which it came in. The SIP gateway name actually contains the information where it should be routed to. Any ideas on how to approach this problem? Question - is it possible to do it in the dialplan (dynamic) or do we have to write an application to do this mapping? -Saurabh have you looked at this example http://wiki.freeswitch.org/wiki/Getting_Started_Guide#Receiving_an_inbound_c... ram _________________________________________________________________ When your life is on the go—take your life with you. http://clk.atdmt.com/MRT/go/115298558/direct/01/
whatever you put in the "extension" param in the gateway should control what destination_number it has in the inbound call. you can also do your regex in your dialplan on any of the info in the sip packet besides destination number if you wish. On Wed, Oct 29, 2008 at 4:52 AM, Saurabh Aggarwal < saurabh_aggarwal@hotmail.com> wrote:
Yes, but there is no DID in my system for incoming calls. I have users dynamically registering gateways, and calls coming in to SIP ids that they have used to register.
-Saurabh
------------------------------
Date: Wed, 29 Oct 2008 15:12:28 +0530 From: talk2ram@gmail.com To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] SIP incoming call routing
On Wed, Oct 29, 2008 at 2:52 PM, Saurabh Aggarwal < saurabh_aggarwal@hotmail.com> wrote:
We are using freeswitch as a SIP proxy, where we are letting people register with freeswitch, and in-turn we do the SIP registration for them to "arbitrary" sip servers (as requested by users) - each user gets his own sip gateway in the freeswitch configuration. Then they can make outgoing calls and calls are routed through their specific SIP gateway.
Now the problem is that when a call is received from one of these SIP registrations, it hits the public.xml where I can't seem to figure out how to get the SIP gateway information from which it came in. The SIP gateway name actually contains the information where it should be routed to. Any ideas on how to approach this problem?
Question - is it possible to do it in the dialplan (dynamic) or do we have to write an application to do this mapping?
-Saurabh
have you looked at this example
http://wiki.freeswitch.org/wiki/Getting_Started_Guide#Receiving_an_inbound_c...
ram
------------------------------ When your life is on the go—take your life with you. Try Windows Mobile(R) today <http://clk.atdmt.com/MRT/go/115298558/direct/01/>
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
Thanks - that does work to an extent. Now the problem is that not all gateways would allow "arbitrary" extensions. E.g. AIM Callout - it *requires* that the extension/caller-id be your aim username. -Saurabh Date: Wed, 29 Oct 2008 12:46:44 -0500From: anthony.minessale@gmail.comTo: freeswitch-users@lists.freeswitch.orgSubject: Re: [Freeswitch-users] SIP incoming call routingwhatever you put in the "extension" param in the gateway should control what destination_number it has in the inbound call. you can also do your regex in your dialplan on any of the info in the sip packet besides destination number if you wish. On Wed, Oct 29, 2008 at 4:52 AM, Saurabh Aggarwal <saurabh_aggarwal@hotmail.com> wrote: Yes, but there is no DID in my system for incoming calls. I have users dynamically registering gateways, and calls coming in to SIP ids that they have used to register. -Saurabh Date: Wed, 29 Oct 2008 15:12:28 +0530From: talk2ram@gmail.comTo: freeswitch-users@lists.freeswitch.orgSubject: Re: [Freeswitch-users] SIP incoming call routing On Wed, Oct 29, 2008 at 2:52 PM, Saurabh Aggarwal <saurabh_aggarwal@hotmail.com> wrote: We are using freeswitch as a SIP proxy, where we are letting people register with freeswitch, and in-turn we do the SIP registration for them to "arbitrary" sip servers (as requested by users) - each user gets his own sip gateway in the freeswitch configuration. Then they can make outgoing calls and calls are routed through their specific SIP gateway. Now the problem is that when a call is received from one of these SIP registrations, it hits the public.xml where I can't seem to figure out how to get the SIP gateway information from which it came in. The SIP gateway name actually contains the information where it should be routed to. Any ideas on how to approach this problem? Question - is it possible to do it in the dialplan (dynamic) or do we have to write an application to do this mapping? -Saurabh have you looked at this example http://wiki.freeswitch.org/wiki/Getting_Started_Guide#Receiving_an_inbound_c... ram When your life is on the go—take your life with you. Try Windows Mobile® today_______________________________________________Freeswitch-users mailing listFreeswitch-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:htt... Anthony Minessale IIFreeSWITCH http://www.freeswitch.org/ClueCon http://www.cluecon.com/AIM: anthmMSN:anthony_minessale@hotmail.comGTALK/JABBER/PAYPAL:anthony.minessale@gmail.comIRC: irc.freenode.net #freeswitchFreeSWITCH Developer Conferencesip:888@conference.freeswitch.orgiax:guest@conference.freeswitch.org/888googletalk:conf+888@conference.freeswitch.orgpstn:213-799-1400 _________________________________________________________________ When your life is on the go—take your life with you. http://clk.atdmt.com/MRT/go/115298558/direct/01/
The extension param only influences the username portion of the contact address?. If they require a certain contact address username they are insane. We also have extra params like from-domain, and caller-id-in-from to compensate for other foolish broken sip services who use pointless unnatural requirements on the contents of your invite for security. If you must set the contact to your username, you would have to match on something else in the invite. It would be up to them to send something significant in the invite that you could match against to tell what number they are calling. destination_number is not the only thing you can regex for, route all calls to the "info" app and call the various did and look for a variable that will tell you the info you need. On Wed, Nov 5, 2008 at 1:36 AM, Saurabh Aggarwal < saurabh_aggarwal@hotmail.com> wrote:
Thanks - that does work to an extent.
Now the problem is that not all gateways would allow "arbitrary" extensions. E.g. AIM Callout - it *requires* that the extension/caller-id be your aim username.
-Saurabh
------------------------------
Date: Wed, 29 Oct 2008 12:46:44 -0500 From: anthony.minessale@gmail.com
To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] SIP incoming call routing
whatever you put in the "extension" param in the gateway should control what destination_number it has in the inbound call. you can also do your regex in your dialplan on any of the info in the sip packet besides destination number if you wish.
On Wed, Oct 29, 2008 at 4:52 AM, Saurabh Aggarwal < saurabh_aggarwal@hotmail.com> wrote:
Yes, but there is no DID in my system for incoming calls. I have users dynamically registering gateways, and calls coming in to SIP ids that they have used to register.
-Saurabh
------------------------------
Date: Wed, 29 Oct 2008 15:12:28 +0530 From: talk2ram@gmail.com To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] SIP incoming call routing
On Wed, Oct 29, 2008 at 2:52 PM, Saurabh Aggarwal < saurabh_aggarwal@hotmail.com> wrote:
We are using freeswitch as a SIP proxy, where we are letting people register with freeswitch, and in-turn we do the SIP registration for them to "arbitrary" sip servers (as requested by users) - each user gets his own sip gateway in the freeswitch configuration. Then they can make outgoing calls and calls are routed through their specific SIP gateway.
Now the problem is that when a call is received from one of these SIP registrations, it hits the public.xml where I can't seem to figure out how to get the SIP gateway information from which it came in. The SIP gateway name actually contains the information where it should be routed to. Any ideas on how to approach this problem?
Question - is it possible to do it in the dialplan (dynamic) or do we have to write an application to do this mapping?
-Saurabh
have you looked at this example
http://wiki.freeswitch.org/wiki/Getting_Started_Guide#Receiving_an_inbound_c...
ram
------------------------------ When your life is on the go—take your life with you. Try Windows Mobile(R) today <http://clk.atdmt.com/MRT/go/115298558/direct/01/>
_______________________________________________ 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
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888<http://conference.freeswitch.org/888> googletalk:conf+888@conference.freeswitch.org<googletalk:conf%2B888@conference.freeswitch.org> pstn:213-799-1400
------------------------------ When your life is on the go—take your life with you. Try Windows Mobile(R) today <http://clk.atdmt.com/MRT/go/115298558/direct/01/>
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
by default only some hand-picked events have all the variables due to people complaining that they had too much info ;) if you want every event to have them you can execute "verbose_events" app in your dialplan right before you call "socket" On Wed, Oct 29, 2008 at 4:03 AM, Dennis <odermann@googlemail.com> wrote:
they are still there, at least if you register to myevents or all events. i use the default settings from fs and get plenty of them.
2008/10/28 Andy Spitzer <woof@nortel.com>:
Woof!
I used to get lots of variable_* lines when using socket_outbound. They have disappeared. Is there something I need to configure to get them back?
--Woof!
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
Woof! On Wed, 29 Oct 2008 09:10:32 -0400, Anthony Minessale <anthony.minessale@gmail.com> wrote:
by default only some hand-picked events have all the variables due to people complaining that they had too much info ;)
Sending the same info over and over does seem counter productive.
if you want every event to have them you can execute "verbose_events" app in your dialplan right before you call "socket"
Thanks. I really only need the verbose stuff on the initial dump after "connect", and from then on the ones that don't change aren't really needed as I suck all them into my own state object. Perhaps a keyword on "connect" like "connect verbose" could initially supply the full list of information, about the call, but not on subsequent events? --Woof!
They should already be on the initial events. Take a look at the raw output, you probably were taking them out of a later event. Mike On Oct 29, 2008, at 2:14 PM, Andy Spitzer wrote:
Woof!
On Wed, 29 Oct 2008 09:10:32 -0400, Anthony Minessale <anthony.minessale@gmail.com> wrote:
by default only some hand-picked events have all the variables due to people complaining that they had too much info ;)
Sending the same info over and over does seem counter productive.
if you want every event to have them you can execute "verbose_events" app in your dialplan right before you call "socket"
Thanks. I really only need the verbose stuff on the initial dump after "connect", and from then on the ones that don't change aren't really needed as I suck all them into my own state object.
Perhaps a keyword on "connect" like "connect verbose" could initially supply the full list of information, about the call, but not on subsequent events?
--Woof!
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Woof! On Wed, 29 Oct 2008 14:18:53 -0400, Michael Jerris <mike@jerris.com> wrote:
They should already be on the initial events. Take a look at the raw output, you probably were taking them out of a later event.
Nope. Initial event. No variable_* are reported. Using netcat: Connection from 47.16.90.233 port 8084 [tcp/*] accepted connect Channel-Username: 207 Channel-Dialplan: XML Channel-Caller-ID-Name: 207 Channel-Caller-ID-Number: 207 Channel-Network-Addr: 47.16.90.233 Channel-Destination-Number: IVR Channel-Unique-ID: 1f6a42fe-ef4d-47f9-897d-ed6387a30cd1 Channel-Source: mod_sofia Channel-Context: default Channel-Channel-Name: sofia/woof.us.nortel.com/207%40woof.us.nortel.com Channel-Profile-Index: 1 Channel-Profile-Created-Time: 1225303798001316 Channel-Channel-Created-Time: 1225303798001316 Channel-Channel-Answered-Time: 0 Channel-Channel-Progress-Time: 1225303798010688 Channel-Channel-Progress-Media-Time: 1225303798010688 Channel-Channel-Hangup-Time: 0 Channel-Channel-Transfer-Time: 0 Channel-Screen-Bit: true Channel-Privacy-Hide-Name: false Channel-Privacy-Hide-Number: false Channel-State: CS_EXECUTE Channel-State-Number: 4 Channel-Name: sofia/woof.us.nortel.com/207%40woof.us.nortel.com Unique-ID: 1f6a42fe-ef4d-47f9-897d-ed6387a30cd1 Call-Direction: inbound Answer-State: early Channel-Read-Codec-Name: PCMU Channel-Read-Codec-Rate: 8000 Channel-Write-Codec-Name: PCMU Channel-Write-Codec-Rate: 8000 Caller-Username: 207 Caller-Dialplan: XML Caller-Caller-ID-Name: 207 Caller-Caller-ID-Number: 207 Caller-Network-Addr: 47.16.90.233 Caller-Destination-Number: IVR Caller-Unique-ID: 1f6a42fe-ef4d-47f9-897d-ed6387a30cd1 Caller-Source: mod_sofia Caller-Context: default Caller-Channel-Name: sofia/woof.us.nortel.com/207%40woof.us.nortel.com Caller-Profile-Index: 1 Caller-Profile-Created-Time: 1225303798001316 Caller-Channel-Created-Time: 1225303798001316 Caller-Channel-Answered-Time: 0 Caller-Channel-Progress-Time: 1225303798010688 Caller-Channel-Progress-Media-Time: 1225303798010688 Caller-Channel-Hangup-Time: 0 Caller-Channel-Transfer-Time: 0 Caller-Screen-Bit: true Caller-Privacy-Hide-Name: false Caller-Privacy-Hide-Number: false Content-Type: command/reply Reply-Text: %2BOK%0A Socket-Mode: async Control: full --Woof! --Woof!
yah like i said, update and try again ;) On Wed, Oct 29, 2008 at 1:24 PM, Andy Spitzer <woof@nortel.com> wrote:
Woof!
On Wed, 29 Oct 2008 14:18:53 -0400, Michael Jerris <mike@jerris.com> wrote:
They should already be on the initial events. Take a look at the raw output, you probably were taking them out of a later event.
Nope. Initial event. No variable_* are reported. Using netcat:
Connection from 47.16.90.233 port 8084 [tcp/*] accepted connect
Channel-Username: 207 Channel-Dialplan: XML Channel-Caller-ID-Name: 207 Channel-Caller-ID-Number: 207 Channel-Network-Addr: 47.16.90.233 Channel-Destination-Number: IVR Channel-Unique-ID: 1f6a42fe-ef4d-47f9-897d-ed6387a30cd1 Channel-Source: mod_sofia Channel-Context: default Channel-Channel-Name: sofia/woof.us.nortel.com/207%40woof.us.nortel.com Channel-Profile-Index: 1 Channel-Profile-Created-Time: 1225303798001316 Channel-Channel-Created-Time: 1225303798001316 Channel-Channel-Answered-Time: 0 Channel-Channel-Progress-Time: 1225303798010688 Channel-Channel-Progress-Media-Time: 1225303798010688 Channel-Channel-Hangup-Time: 0 Channel-Channel-Transfer-Time: 0 Channel-Screen-Bit: true Channel-Privacy-Hide-Name: false Channel-Privacy-Hide-Number: false Channel-State: CS_EXECUTE Channel-State-Number: 4 Channel-Name: sofia/woof.us.nortel.com/207%40woof.us.nortel.com Unique-ID: 1f6a42fe-ef4d-47f9-897d-ed6387a30cd1 Call-Direction: inbound Answer-State: early Channel-Read-Codec-Name: PCMU Channel-Read-Codec-Rate: 8000 Channel-Write-Codec-Name: PCMU Channel-Write-Codec-Rate: 8000 Caller-Username: 207 Caller-Dialplan: XML Caller-Caller-ID-Name: 207 Caller-Caller-ID-Number: 207 Caller-Network-Addr: 47.16.90.233 Caller-Destination-Number: IVR Caller-Unique-ID: 1f6a42fe-ef4d-47f9-897d-ed6387a30cd1 Caller-Source: mod_sofia Caller-Context: default Caller-Channel-Name: sofia/woof.us.nortel.com/207%40woof.us.nortel.com Caller-Profile-Index: 1 Caller-Profile-Created-Time: 1225303798001316 Caller-Channel-Created-Time: 1225303798001316 Caller-Channel-Answered-Time: 0 Caller-Channel-Progress-Time: 1225303798010688 Caller-Channel-Progress-Media-Time: 1225303798010688 Caller-Channel-Hangup-Time: 0 Caller-Channel-Transfer-Time: 0 Caller-Screen-Bit: true Caller-Privacy-Hide-Name: false Caller-Privacy-Hide-Number: false Content-Type: command/reply Reply-Text: %2BOK%0A Socket-Mode: async Control: full
--Woof!
--Woof!
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
Woof! On Wed, 29 Oct 2008 14:31:36 -0400, Anthony Minessale <anthony.minessale@gmail.com> wrote:
update and try again ;)
Anthony, you are just too fast! Mike was saying one thing, and you checked in the change it while I was replying to him! Updating now. --Woof!
i updated event_socket so that first event has the variables, that seems like you would always want that. On Wed, Oct 29, 2008 at 1:14 PM, Andy Spitzer <woof@nortel.com> wrote:
Woof!
On Wed, 29 Oct 2008 09:10:32 -0400, Anthony Minessale <anthony.minessale@gmail.com> wrote:
by default only some hand-picked events have all the variables due to people complaining that they had too much info ;)
Sending the same info over and over does seem counter productive.
if you want every event to have them you can execute "verbose_events" app in your dialplan right before you call "socket"
Thanks. I really only need the verbose stuff on the initial dump after "connect", and from then on the ones that don't change aren't really needed as I suck all them into my own state object.
Perhaps a keyword on "connect" like "connect verbose" could initially supply the full list of information, about the call, but not on subsequent events?
--Woof!
_______________________________________________ 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
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
participants (6)
-
Andy Spitzer
-
Anthony Minessale
-
Dennis
-
Michael Jerris
-
ram
-
Saurabh Aggarwal