mod_dingaling and google voice
Now I am able to use GV for outbound dialing. However, I don't hear any ringback or not sure whether the other party has answered the call. is there any config variable to set up so that I will be notified if my call is answered? thanks!
never mind. add this line <action application="ring_ready"/> and restart FS fixed the issue. On Sun, Feb 27, 2011 at 10:30 PM, envelopes envelopes <sunwood360@gmail.com>wrote:
Now I am able to use GV for outbound dialing. However, I don't hear any ringback or not sure whether the other party has answered the call. is there any config variable to set up so that I will be notified if my call is answered?
thanks!
What FS build are you running? I'd like to compare notes with you on your config, as I'm experiencing the RTP port 0 issue with FreeSWITCH Version 1.0.head (git-0d8e945 2011-02-16 08-57-43 -0600). What about inbound? Do you have that set up? Yossi Neiman Cartis Solutions, Inc. - http://www.cartissolutions.com On 02/28/2011 12:54 AM, envelopes envelopes wrote:
|never mind. add this line <action application="ring_ready"/> and restart FS
fixed the issue.
| On Sun, Feb 27, 2011 at 10:30 PM, envelopes envelopes <sunwood360@gmail.com <mailto:sunwood360@gmail.com>> wrote:
Now I am able to use GV for outbound dialing. However, I don't hear any ringback or not sure whether the other party has answered the call. is there any config variable to set up so that I will be notified if my call is answered?
thanks!
_______________________________________________ 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
The solution to the port 0 issue is to run FS with -nonat so it does not get confused by broken upnp setup. On Mon, Feb 28, 2011 at 3:54 PM, Yossi Neiman <freeswitch@cartissolutions.com> wrote:
What FS build are you running? I'd like to compare notes with you on your config, as I'm experiencing the RTP port 0 issue with FreeSWITCH Version 1.0.head (git-0d8e945 2011-02-16 08-57-43 -0600). What about inbound? Do you have that set up?
Yossi Neiman Cartis Solutions, Inc. - http://www.cartissolutions.com
On 02/28/2011 12:54 AM, envelopes envelopes wrote:
never mind. add this line <action application="ring_ready"/> and restart FS
fixed the issue.
On Sun, Feb 27, 2011 at 10:30 PM, envelopes envelopes <sunwood360@gmail.com> wrote:
Now I am able to use GV for outbound dialing. However, I don't hear any ringback or not sure whether the other party has answered the call. is there any config variable to set up so that I will be notified if my call is answered?
thanks!
_______________________________________________ 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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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire 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 googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
I am running with -nonat, the FS is running with a UA on a public IP address, etc, but both Michael Collins and I are still getting the RTP port 0 issue. I figured I'd compare notes with somebody who has a working config before I file this as something other than a PEBKAC issue... Yossi Neiman Cartis Solutions, Inc. - http://www.cartissolutions.com On 02/28/2011 04:07 PM, Anthony Minessale wrote:
The solution to the port 0 issue is to run FS with -nonat so it does not get confused by broken upnp setup.
On Mon, Feb 28, 2011 at 3:54 PM, Yossi Neiman <freeswitch@cartissolutions.com> wrote:
What FS build are you running? I'd like to compare notes with you on your config, as I'm experiencing the RTP port 0 issue with FreeSWITCH Version 1.0.head (git-0d8e945 2011-02-16 08-57-43 -0600). What about inbound? Do you have that set up?
Yossi Neiman Cartis Solutions, Inc. - http://www.cartissolutions.com
On 02/28/2011 12:54 AM, envelopes envelopes wrote:
never mind. add this line<action application="ring_ready"/> and restart FS
fixed the issue.
On Sun, Feb 27, 2011 at 10:30 PM, envelopes envelopes<sunwood360@gmail.com> wrote:
Now I am able to use GV for outbound dialing. However, I don't hear any ringback or not sure whether the other party has answered the call. is there any config variable to set up so that I will be notified if my call is answered?
thanks!
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you might want to backup your configs and start over with the fresh ones just to test and work backwards to see what's different. its probably something to do with having to omit external-rtp-ip On Mon, Feb 28, 2011 at 4:58 PM, Yossi Neiman <freeswitch@cartissolutions.com> wrote:
I am running with -nonat, the FS is running with a UA on a public IP address, etc, but both Michael Collins and I are still getting the RTP port 0 issue. I figured I'd compare notes with somebody who has a working config before I file this as something other than a PEBKAC issue...
Yossi Neiman Cartis Solutions, Inc. - http://www.cartissolutions.com
On 02/28/2011 04:07 PM, Anthony Minessale wrote:
The solution to the port 0 issue is to run FS with -nonat so it does not get confused by broken upnp setup.
On Mon, Feb 28, 2011 at 3:54 PM, Yossi Neiman <freeswitch@cartissolutions.com> wrote:
What FS build are you running? I'd like to compare notes with you on your config, as I'm experiencing the RTP port 0 issue with FreeSWITCH Version 1.0.head (git-0d8e945 2011-02-16 08-57-43 -0600). What about inbound? Do you have that set up?
Yossi Neiman Cartis Solutions, Inc. - http://www.cartissolutions.com
On 02/28/2011 12:54 AM, envelopes envelopes wrote:
never mind. add this line<action application="ring_ready"/> and restart FS
fixed the issue.
On Sun, Feb 27, 2011 at 10:30 PM, envelopes envelopes<sunwood360@gmail.com> wrote:
Now I am able to use GV for outbound dialing. However, I don't hear any ringback or not sure whether the other party has answered the call. is there any config variable to set up so that I will be notified if my call is answered?
thanks!
_______________________________________________ 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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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire 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 googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
This is a completely fresh rebuild of my configurations from the ground up. jingle configs: http://pastebin.freeswitch.org/15510 Inbound call: http://pastebin.freeswitch.org/15512 Outbound call: http://pastebin.freeswitch.org/15513 FreeSWITCH Version 1.0.head (git-0d8e945 2011-02-16 08-57-43 -0600), launched with this: #!/bin/sh ulimit -s 240 exec /usr/local/freeswitch/bin/freeswitch -u SOME_FS_USER -g SOME_FS_USERS_GROUP -conf SOME_FS_USER_HOMEDIR/instance_00/conf -log SOME_FS_USER_HOMEDIR/instance_00/log -run SOME_FS_USER_HOMEDIR/instance_00/run -db SOME_FS_USER_HOMEDIR/instance_00/db -scripts SOME_FS_USER_HOMEDIR/instance_00/scripts -nonat Sofia UA's running on both 99.X.X.X:5060 and 192.X.X.X:5060. Machine FS runs on is an iptables NAT/Firewall/router. SIP calls to/from the outside world work fine otherwise. -Yossi On 02/28/2011 05:02 PM, Anthony Minessale wrote:
you might want to backup your configs and start over with the fresh ones just to test and work backwards to see what's different. its probably something to do with having to omit external-rtp-ip
On Mon, Feb 28, 2011 at 4:58 PM, Yossi Neiman <freeswitch@cartissolutions.com> wrote:
I am running with -nonat, the FS is running with a UA on a public IP address, etc, but both Michael Collins and I are still getting the RTP port 0 issue. I figured I'd compare notes with somebody who has a working config before I file this as something other than a PEBKAC issue...
Yossi Neiman Cartis Solutions, Inc. - http://www.cartissolutions.com
On 02/28/2011 04:07 PM, Anthony Minessale wrote:
The solution to the port 0 issue is to run FS with -nonat so it does not get confused by broken upnp setup.
On Mon, Feb 28, 2011 at 3:54 PM, Yossi Neiman <freeswitch@cartissolutions.com> wrote:
What FS build are you running? I'd like to compare notes with you on your config, as I'm experiencing the RTP port 0 issue with FreeSWITCH Version 1.0.head (git-0d8e945 2011-02-16 08-57-43 -0600). What about inbound? Do you have that set up?
Yossi Neiman Cartis Solutions, Inc. - http://www.cartissolutions.com
On 02/28/2011 12:54 AM, envelopes envelopes wrote:
never mind. add this line<action application="ring_ready"/> and restart FS
fixed the issue.
On Sun, Feb 27, 2011 at 10:30 PM, envelopes envelopes<sunwood360@gmail.com> wrote:
Now I am able to use GV for outbound dialing. However, I don't hear any ringback or not sure whether the other party has answered the call. is there any config variable to set up so that I will be notified if my call is answered?
thanks!
Yossi Neiman wrote:
Sofia UA's running on both 99.X.X.X:5060 and 192.X.X.X:5060. Machine FS runs on is an iptables NAT/Firewall/router. SIP calls to/from the outside world work fine otherwise. I am lost here. You have two FS machines and one on a public IP Address while the other on a private IP Address? Both are having problems with GV incoming calls?
----- FreeSWITCH hosted on a Seagate DockStar with OpenWRT. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-dingaling-and-google-voice... Sent from the freeswitch-users mailing list archive at Nabble.com.
Negative. I am working with a single instance of FS in this scenario. This single instance just happens to have a SIP UA that runs on 99.X.X.X:5060 and one on 192.168.X.X:5060. Updating to: FreeSWITCH Version 1.0.head (git-9a7dbfb 2011-03-01 13-44-53 -0600) Fixed the outbound calling. But inbound is still not dialplan hunting. http://pastebin.freeswitch.org/15525 Tony, if you have a moment I'm curious to know more about the edge case here. I'm using arno's iptables firewall on that box, and using iptables 1.4.7, kernel 2.6.37. Yossi Neiman Cartis Solutions, Inc. - http://www.cartissolutions.com On 03/01/2011 07:48 PM, mazilo wrote:
Yossi Neiman wrote:
Sofia UA's running on both 99.X.X.X:5060 and 192.X.X.X:5060. Machine FS runs on is an iptables NAT/Firewall/router. SIP calls to/from the outside world work fine otherwise. I am lost here. You have two FS machines and one on a public IP Address while the other on a private IP Address? Both are having problems with GV incoming calls?
----- FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
I don't use GV for direct inbound, because it is not working very well. - Get an account from sipgate.com; - let GV forward inbound to your sipgate PSTN # - and your FS register as a sipgate client. On Mon, Feb 28, 2011 at 1:54 PM, Yossi Neiman < freeswitch@cartissolutions.com> wrote:
What FS build are you running? I'd like to compare notes with you on your config, as I'm experiencing the RTP port 0 issue with FreeSWITCH Version 1.0.head (git-0d8e945 2011-02-16 08-57-43 -0600). What about inbound? Do you have that set up?
Yossi Neiman Cartis Solutions, Inc. - http://www.cartissolutions.com
On 02/28/2011 12:54 AM, envelopes envelopes wrote:
never mind. add this line <action application="ring_ready"/> and restart FS
fixed the issue.
On Sun, Feb 27, 2011 at 10:30 PM, envelopes envelopes < sunwood360@gmail.com> wrote:
Now I am able to use GV for outbound dialing. However, I don't hear any ringback or not sure whether the other party has answered the call. is there any config variable to set up so that I will be notified if my call is answered?
thanks!
_______________________________________________ FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.frees...
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Or get a gizmo5 account and then you can just do SIP forwarding to FS, without the pstn loop. You can find them around for a fee bucks. I have a few, not sure if I'm going to use them... -Avi On Mar 1, 2011 4:59 AM, "envelopes envelopes" <sunwood360@gmail.com> wrote:
I don't use GV for direct inbound, because it is not working very well.
- Get an account from sipgate.com; - let GV forward inbound to your sipgate PSTN # - and your FS register as a sipgate client.
On Mon, Feb 28, 2011 at 1:54 PM, Yossi Neiman < freeswitch@cartissolutions.com> wrote:
What FS build are you running? I'd like to compare notes with you on your config, as I'm experiencing the RTP port 0 issue with FreeSWITCH Version 1.0.head (git-0d8e945 2011-02-16 08-57-43 -0600). What about inbound? Do you have that set up?
Yossi Neiman Cartis Solutions, Inc. - http://www.cartissolutions.com
On 02/28/2011 12:54 AM, envelopes envelopes wrote:
never mind. add this line <action application="ring_ready"/> and restart FS
fixed the issue.
On Sun, Feb 27, 2011 at 10:30 PM, envelopes envelopes < sunwood360@gmail.com> wrote:
Now I am able to use GV for outbound dialing. However, I don't hear any ringback or not sure whether the other party has answered the call. is there any config variable to set up so that I will be notified if my call is answered?
thanks!
_______________________________________________ FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp:// lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.frees...
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envelopes envelopes wrote:
I don't use GV for direct inbound, because it is not working very well. Care to explain what problems you have encountered with GV direct inbound?
- Get an account from sipgate.com;
- let GV forward inbound to your sipgate PSTN # - and your FS register as a sipgate client. This works just fine with an additional latency.
----- FreeSWITCH hosted on a Seagate DockStar with OpenWRT. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-dingaling-and-google-voice... Sent from the freeswitch-users mailing list archive at Nabble.com.
participants (5)
-
Anthony Minessale
-
Avi Marcus
-
envelopes envelopes
-
mazilo
-
Yossi Neiman