google voice connection going offline (mod_dingaling)
Hi All, I've setup mod_dingaling to connect to my google voice account according to the instructions on the wiki http://wiki.freeswitch.org/wiki/Google_Voice#Setup_FreeSWITCH_-_Dingaling_to... with two modifications: <param name="rtp-ip" value="auto"/> <param name="ext-rtp-ip" value="auto-nat"/> I use the same IP settings for SIP (sofia) with no problem. The setup works fine for the first hour or two, but then I can see from another google account that the connection established by freeswitch goes offline. If I try anyway to make an outbound call, then the other end does not ring (even if I wait for a long time). However, after some minutes the freeswitch account returns online and works again for a couple of hours. I'm behind a nat and firewall. Is there any port which I have to open to keep freeswitch online? As far as I understand only the google XMMP server needs to listen to port 5222 and port 5269. Or, is there any kind of "keep alive" timer available? The google talk client running in the same environment works flawlessly. Thanks for any advise. Fede
I also had problem in making FS mod_dingaling work with google voice: - enable mod_dingaling in modules.conf.xml at first; - copy the following content into disk file [freeswitch/conf/jingle_profiles/xk_client.xml] according to wiki page [http://wiki.freeswitch.org/wiki/Google_Voice]: <include> <profile type="client"> <param name="name" value="gtalk"/> <param name="login" value="GV-ACCOUNT-NAME@gmail.com/talk"/> <param name="password" value="MY-PASSWORD"/> <param name="server" value="talk.google.com"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="message" value="Hello from Freeswitch!"/> <param name="rtp-ip" value="$${bind_server_ip}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="auto-login" value="true"/> <!-- SASL "plain" or "md5? --> <param name="sasl" value="plain"/> <!-- Enable TLS or not --> <param name="tls" value="true"/> <!-- disable to trade async for more calls --> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="none"/> <!-- default extension --> <param name="exten" value="2001"/> <param name="vad" value="both"/> <param name="candidate-acl" value="wan.auto"/> <param name="local-network-acl" value="localnet.auto"/> </profile> </include> - then, since [freeswitch/conf/jingle_profiles/] already contains cleint.xml and server.xml, I made four changes to the above content of xk_client.xml: <profile type="client"> ===> <profile type="xk_client"> <param name="login" value="GV-ACCOUNT-NAME@gmail.com/talk"/> ===> update it with my own gmail account <param name="password" value="MY-PASSWORD"/> ===> update it with my own password to gmail account. <param name="exten" value="2001"/> ===> <param name="exten" value="1006"/> - add a dialplan extension for default context: <extension name="gvoice_out"> <condition field="destination_number" expression="^50(1\d{10})$"> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="dingaling/gtalk/+$1@voice.google.com"/> </condition> </extension> - also, set up my google voice settings as required by FS - finally, dial out "50 1 408 543 xxxx" on a sip client with internal user "1006", the call was never completed. ----- Original Message ----- From: Federico Beffa <beffa@ieee.org> To: freeswitch-users@lists.freeswitch.org Cc: Sent: Wednesday, September 28, 2011 4:14 PM Subject: [Freeswitch-users] google voice connection going offline (mod_dingaling) Hi All, I've setup mod_dingaling to connect to my google voice account according to the instructions on the wiki http://wiki.freeswitch.org/wiki/Google_Voice#Setup_FreeSWITCH_-_Dingaling_to... with two modifications: <param name="rtp-ip" value="auto"/> <param name="ext-rtp-ip" value="auto-nat"/> I use the same IP settings for SIP (sofia) with no problem. The setup works fine for the first hour or two, but then I can see from another google account that the connection established by freeswitch goes offline. If I try anyway to make an outbound call, then the other end does not ring (even if I wait for a long time). However, after some minutes the freeswitch account returns online and works again for a couple of hours. I'm behind a nat and firewall. Is there any port which I have to open to keep freeswitch online? As far as I understand only the google XMMP server needs to listen to port 5222 and port 5269. Or, is there any kind of "keep alive" timer available? The google talk client running in the same environment works flawlessly. Thanks for any advise. Fede 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
Thanks for the suggestion. That is about what I did and calls work in both directions: outbound and inbound. This however only for a couple of hours. Then the google account assigned to FS goes offline and calls stop working. I have tried to track events and noticed the following: *) fs_cli with log to 7 does not report anything when the account goes offline. *) my NAT/Firewall sends the following message "Firewall session time out, sent TCP RST: TCP" to both my FS host and the google server. This happens to the google talk client from google as well (running on another machine on the same network). However, while the latter handles the situation well and is able to keep the account online, it appears that FS fails to act properly. Is it possible to set a "keep alive" kind of timer in FS? Thanks, Fede On Thu, Sep 29, 2011 at 5:11 AM, king2kin <xing2kin@yahoo.com> wrote:
I also had problem in making FS mod_dingaling work with google voice:
- enable mod_dingaling in modules.conf.xml at first;
- copy the following content into disk file [freeswitch/conf/jingle_profiles/xk_client.xml] according to wiki page [http://wiki.freeswitch.org/wiki/Google_Voice]:
<include> <profile type="client"> <param name="name" value="gtalk"/> <param name="login" value="GV-ACCOUNT-NAME@gmail.com/talk"/> <param name="password" value="MY-PASSWORD"/> <param name="server" value="talk.google.com"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="message" value="Hello from Freeswitch!"/> <param name="rtp-ip" value="$${bind_server_ip}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="auto-login" value="true"/> <!-- SASL "plain" or "md5? --> <param name="sasl" value="plain"/> <!-- Enable TLS or not --> <param name="tls" value="true"/> <!-- disable to trade async for more calls --> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="none"/> <!-- default extension --> <param name="exten" value="2001"/> <param name="vad" value="both"/> <param name="candidate-acl" value="wan.auto"/> <param name="local-network-acl" value="localnet.auto"/> </profile> </include> - then, since [freeswitch/conf/jingle_profiles/] already contains cleint.xml and server.xml, I made four changes to the above content of xk_client.xml:
<profile type="client"> ===> <profile type="xk_client">
<param name="login" value="GV-ACCOUNT-NAME@gmail.com/talk"/> ===> update it with my own gmail account <param name="password" value="MY-PASSWORD"/> ===> update it with my own password to gmail account. <param name="exten" value="2001"/> ===> <param name="exten" value="1006"/> - add a dialplan extension for default context:
<extension name="gvoice_out"> <condition field="destination_number" expression="^50(1\d{10})$"> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="dingaling/gtalk/+$1@voice.google.com"/> </condition> </extension> - also, set up my google voice settings as required by FS
- finally, dial out "50 1 408 543 xxxx" on a sip client with internal user "1006", the call was never completed.
----- Original Message ----- From: Federico Beffa <beffa@ieee.org> To: freeswitch-users@lists.freeswitch.org Cc: Sent: Wednesday, September 28, 2011 4:14 PM Subject: [Freeswitch-users] google voice connection going offline (mod_dingaling)
Hi All,
I've setup mod_dingaling to connect to my google voice account according to the instructions on the wiki http://wiki.freeswitch.org/wiki/Google_Voice#Setup_FreeSWITCH_-_Dingaling_to... with two modifications: <param name="rtp-ip" value="auto"/> <param name="ext-rtp-ip" value="auto-nat"/>
I use the same IP settings for SIP (sofia) with no problem.
The setup works fine for the first hour or two, but then I can see from another google account that the connection established by freeswitch goes offline. If I try anyway to make an outbound call, then the other end does not ring (even if I wait for a long time). However, after some minutes the freeswitch account returns online and works again for a couple of hours.
I'm behind a nat and firewall. Is there any port which I have to open to keep freeswitch online? As far as I understand only the google XMMP server needs to listen to port 5222 and port 5269. Or, is there any kind of "keep alive" timer available?
The google talk client running in the same environment works flawlessly.
Thanks for any advise. Fede
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
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 only thing I can say is to update your FS with the latest git commit. ----- FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/google-voice-connection-going-... Sent from the freeswitch-users mailing list archive at Nabble.com.
Just in case someone will hit the same problem... After some tests, I've found that, changing "bind_server_ip" from "auto" to the local IP address of the FS server (192.168.X.X), appears to solve my timeout problem. (The account did not go offline for more than 12 hours.) To summarize my current configuration: I have a fix public IP assigned to my router (NAT/Firewall). NAT is configured to forward ports 5080 and 5081 to the host on which FS is running and UPnP is disabled. I'm running a FS version checked out from git the 20th of August. Regards, Fede On Thu, Sep 29, 2011 at 5:04 PM, mazilo <Nabble_01394@slickdeals.endjunk.com> wrote:
The only thing I can say is to update your FS with the latest git commit.
----- FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/google-voice-connection-going-... Sent from the freeswitch-users mailing list archive at Nabble.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
Unfortunately I was singing victory too early. My problem still persists... The git version I'm running is from the 20th of August. In the mean time have there been changes which could affect mod_dingaling? Looking in the git commit comments I have the impression that the answer is no. Regards, Fede On Tue, Oct 4, 2011 at 9:25 AM, Federico Beffa <beffa@ieee.org> wrote:
Just in case someone will hit the same problem...
After some tests, I've found that, changing "bind_server_ip" from "auto" to the local IP address of the FS server (192.168.X.X), appears to solve my timeout problem. (The account did not go offline for more than 12 hours.)
To summarize my current configuration: I have a fix public IP assigned to my router (NAT/Firewall). NAT is configured to forward ports 5080 and 5081 to the host on which FS is running and UPnP is disabled.
I'm running a FS version checked out from git the 20th of August.
Regards, Fede
On Thu, Sep 29, 2011 at 5:04 PM, mazilo <Nabble_01394@slickdeals.endjunk.com> wrote:
The only thing I can say is to update your FS with the latest git commit.
----- FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/google-voice-connection-going-... Sent from the freeswitch-users mailing list archive at Nabble.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
Changes in the FreeSWITCH core could affect the way how mod_dingaling functions. You might want to give the latest git a shot and see what happens. Personally, updating to the latest git isn't the solution to all problems ;) On Thu, Oct 6, 2011 at 12:18 AM, Federico Beffa <beffa@ieee.org> wrote:
Unfortunately I was singing victory too early. My problem still persists...
The git version I'm running is from the 20th of August. In the mean time have there been changes which could affect mod_dingaling? Looking in the git commit comments I have the impression that the answer is no.
Regards, Fede
On Tue, Oct 4, 2011 at 9:25 AM, Federico Beffa <beffa@ieee.org> wrote:
Just in case someone will hit the same problem...
After some tests, I've found that, changing "bind_server_ip" from "auto" to the local IP address of the FS server (192.168.X.X), appears to solve my timeout problem. (The account did not go offline for more than 12 hours.)
To summarize my current configuration: I have a fix public IP assigned to my router (NAT/Firewall). NAT is configured to forward ports 5080 and 5081 to the host on which FS is running and UPnP is disabled.
I'm running a FS version checked out from git the 20th of August.
Regards, Fede
On Thu, Sep 29, 2011 at 5:04 PM, mazilo <Nabble_01394@slickdeals.endjunk.com> wrote:
The only thing I can say is to update your FS with the latest git commit.
----- FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/google-voice-connection-going-... Sent from the freeswitch-users mailing list archive at Nabble.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
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
There is no keep alive functionality in the core XMPP library we use in the core of libdingaling (IKSEMEL) One option would to have a NAT that doesn't disconnect active connections just because they are idle (which seems bad to me). Otherwise you could work on a patch, request a feature or come up with some cron job maybe to use the chat api command to send a message to your server from cron or something as a workaround. On Thu, Oct 6, 2011 at 2:18 AM, Federico Beffa <beffa@ieee.org> wrote:
Unfortunately I was singing victory too early. My problem still persists...
The git version I'm running is from the 20th of August. In the mean time have there been changes which could affect mod_dingaling? Looking in the git commit comments I have the impression that the answer is no.
Regards, Fede
On Tue, Oct 4, 2011 at 9:25 AM, Federico Beffa <beffa@ieee.org> wrote:
Just in case someone will hit the same problem...
After some tests, I've found that, changing "bind_server_ip" from "auto" to the local IP address of the FS server (192.168.X.X), appears to solve my timeout problem. (The account did not go offline for more than 12 hours.)
To summarize my current configuration: I have a fix public IP assigned to my router (NAT/Firewall). NAT is configured to forward ports 5080 and 5081 to the host on which FS is running and UPnP is disabled.
I'm running a FS version checked out from git the 20th of August.
Regards, Fede
On Thu, Sep 29, 2011 at 5:04 PM, mazilo <Nabble_01394@slickdeals.endjunk.com> wrote:
The only thing I can say is to update your FS with the latest git commit.
----- FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/google-voice-connection-going-... Sent from the freeswitch-users mailing list archive at Nabble.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
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/ 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
Hi Anthony, thanks for the clarification and the suggestion. As an immediate workaround I would like to try the "cron" approach you suggested. I have therefore experimented a little bit with the chat api command. My jingle_profiles/client.xml looks essentially as the one provided with the example configuration: <profile type="client"> <param name="name" value="$${xmpp_client_profile}"/> <param name="login" value="user1@gmail.com/talk"/> <param name="password" value="XXXX"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="message" value="Jingle all the way"/> <param name="rtp-ip" value="$${bind_server_ip}"/> <param name="ext-rtp-ip" value="auto-nat"/> <param name="auto-login" value="true"/> <!-- SASL "plain" or "md5" --> <param name="sasl" value="plain"/> <!-- if the server where the jabber is hosted is not the same as the one in the jid --> <param name="server" value="talk.google.com"/> <!-- Enable TLS or not --> <param name="tls" value="true"/> <!-- disable to trade async for more calls --> <param name="use-rtp-timer" value="true"/> <!-- default extension (if one cannot be determined) --> <param name="exten" value="888"/> <!-- VAD choose one --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!--<param name="vad" value="both"/>--> <!--<param name="avatar" value="/path/to/tiny.jpg"/>--> <!--<param name="candidate-acl" value="wan.auto"/>--> <param name="local-network-acl" value="localnet.auto"/> </profile> When I try to use the chat command at the fs_cli prompt I'm unable to form a full jid and always get an error message from the remote server. Here an example: chat jingle|888@xmppc|user2@gmail.com|test Sent freeswitch@ubuntu-vbox-t5500> 2011-10-08 14:22:09.804596 [NOTICE] libdingaling.c:1373 SecSEND: ------------------------------------------------------------------------------- <message to="user2@gmail.com" type="chat" from="user1@gmail.com"> <body>test</body> </message> 2011-10-08 14:22:09.844698 [INFO] libdingaling.c:1371 SecRECV: ------------------------------------------------------------------------------- <message to="user1@gmail.com" type="error" from="user2@gmail.com"> <body>test</body> <error code="400" type="modify"> <bad-request xmlns="urn:ietf:params:xml:ns:xmpp-stanzas"></bad-request> <text xmlns="urn:ietf:params:xml:ns:xmpp-stanzas">If set, the 'from' attribute must be set to the user's full JID.</text> </error> </message> "xmppc" is the value of the variable "xmpp_client_profile" as in the default config. I get the same error independently from what I set before "@xmppc" in the from_jid. Differently from this, if as the from_jid I set my explicit jid "user1@gmail.com/talk" as in the client.xml file I get the error: 2011-10-08 14:20:50.446696 [ERR] mod_dingaling.c:544 Invalid Profile gmail.com Despite the fact that even a wrong message reaching the remote server could work as a "keep alive" signal, I still would like to understand the proper syntax. I would be grateful if somebody could explain me what is the correct syntax to use with chat and jingle? Thank you! Fede P.S.: By the way, I'm now working on HEAD. On Fri, Oct 7, 2011 at 7:59 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
There is no keep alive functionality in the core XMPP library we use in the core of libdingaling (IKSEMEL)
One option would to have a NAT that doesn't disconnect active connections just because they are idle (which seems bad to me).
Otherwise you could work on a patch, request a feature or come up with some cron job maybe to use the chat api command to send a message to your server from cron or something as a workaround.
On Thu, Oct 6, 2011 at 2:18 AM, Federico Beffa <beffa@ieee.org> wrote:
Unfortunately I was singing victory too early. My problem still persists...
The git version I'm running is from the 20th of August. In the mean time have there been changes which could affect mod_dingaling? Looking in the git commit comments I have the impression that the answer is no.
Regards, Fede
On Tue, Oct 4, 2011 at 9:25 AM, Federico Beffa <beffa@ieee.org> wrote:
Just in case someone will hit the same problem...
After some tests, I've found that, changing "bind_server_ip" from "auto" to the local IP address of the FS server (192.168.X.X), appears to solve my timeout problem. (The account did not go offline for more than 12 hours.)
To summarize my current configuration: I have a fix public IP assigned to my router (NAT/Firewall). NAT is configured to forward ports 5080 and 5081 to the host on which FS is running and UPnP is disabled.
I'm running a FS version checked out from git the 20th of August.
Regards, Fede
On Thu, Sep 29, 2011 at 5:04 PM, mazilo <Nabble_01394@slickdeals.endjunk.com> wrote:
The only thing I can say is to update your FS with the latest git commit.
----- FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/google-voice-connection-going-... Sent from the freeswitch-users mailing list archive at Nabble.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
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/ 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
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
Hi Federico and Anthony, Were you guys able to come up with a solution for this? I had mine working for about a month without any problems. I'm running the git version from October 1, 2011. I just ran into the same problem after changing my router since I changed my Internet connection from DSL to fiber optic. Reading the comments from this post: http://www.personal.psu.edu/wcs131/blogs/psuvoip/2010/10/using_freeswitch_to... _google.html I guess this is an issue with Freeswitch having problems when there are Internet disconnections. I guess one way to solve this would be to change my router, but I don't really want to buy another ONT router, besides I don't even know if my ISP would let me use a different ONT since the ones that they provide are all "customized". It is really hard to tell when it has disconnected. Even when typing "dingaling status" from the fs_cli, it will show AUTHORIZED when it is disconnected. The only way to tell if it's been disconnected is checking whether I can make calls or not. To solve this ,without having to restart freeswitch I would "disconnect" (according to FS) from my gtalk profile typing "dl_logout gtalk" and reconnect with "dl_login profile=gtalk". I also had the idea to use the "cron" approach and thought this was going to be the solution to all my problems. Of course, I had to test what would happen if the cron job would happen to run in the middle of a call. I manually typed the commands on the fs_cli while I was on a call and my call did not die!! Great!! I had the solution to the problem!! I then finished the call and ooohh surprise!! I was suddenly disconnected from the fs_cli, FS would die, close unexpectedly and create a core dump. I would really like to help to solve this so let me know if there's any way I can help out. I would like to try the chat approach to see if that works. Did you figure out what is the right syntax to chat? Greetings, Ricardo
Dear Ricardo, unfortunately the short answer is no, I still have intermittent disconnection problems. Some more details: * Sending a chat message from mod_dingaling to gtalk was problematic as google assigns you an from JID including a random number. However, to avoid having to find out this random number, in communicating with google talk, the from JID can be left blank.The special keywork "auto_from" has recently been added to mod_dingaling to make use of this functionality with the chat application. Please see http://jira.freeswitch.org/browse/FS-3611?page=com.atlassian.jira.plugin.sys... for more details. * To mod_dingaling a "keep alive" functionality has been added. In practice this sends a small packet to google talk at regular intervals. The functionality has been integrated in the git repository, but not the latest patch: http://jira.freeswitch.org/browse/FS-3612?page=com.atlassian.jira.plugin.sys... Not sure why the latest patch has not yet been integrated. Maybe lack of time on the part of the developers. Disappointingly this does not solve my problem. I also tested sending "presence status" messages as keep alive packets, but this does not help either. I've dug a little bit further into the code to find out exactly where the problem occurs, but I've not found a solution. See http://lists.freeswitch.org/pipermail/freeswitch-dev/2011-October/005341.htm... Regards, Fede On Thu, Dec 1, 2011 at 7:05 AM, Ricardo Monroy <monroyricardo@gmail.com> wrote:
Hi Federico and Anthony,
Were you guys able to come up with a solution for this? I had mine working for about a month without any problems. I'm running the git version from October 1, 2011. I just ran into the same problem after changing my router since I changed my Internet connection from DSL to fiber optic. Reading the comments from this post: http://www.personal.psu.edu/wcs131/blogs/psuvoip/2010/10/using_freeswitch_to... _google.html I guess this is an issue with Freeswitch having problems when there are Internet disconnections. I guess one way to solve this would be to change my router, but I don't really want to buy another ONT router, besides I don't even know if my ISP would let me use a different ONT since the ones that they provide are all "customized". It is really hard to tell when it has disconnected. Even when typing "dingaling status" from the fs_cli, it will show AUTHORIZED when it is disconnected. The only way to tell if it's been disconnected is checking whether I can make calls or not. To solve this ,without having to restart freeswitch I would "disconnect" (according to FS) from my gtalk profile typing "dl_logout gtalk" and reconnect with "dl_login profile=gtalk". I also had the idea to use the "cron" approach and thought this was going to be the solution to all my problems. Of course, I had to test what would happen if the cron job would happen to run in the middle of a call. I manually typed the commands on the fs_cli while I was on a call and my call did not die!! Great!! I had the solution to the problem!! I then finished the call and ooohh surprise!! I was suddenly disconnected from the fs_cli, FS would die, close unexpectedly and create a core dump. I would really like to help to solve this so let me know if there's any way I can help out. I would like to try the chat approach to see if that works. Did you figure out what is the right syntax to chat?
Greetings,
Ricardo
_________________________________________________________________________ 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
Federico Beffa <beffa@...> writes:
Hi Anthony,
thanks for the clarification and the suggestion. As an immediate workaround I would like to try the "cron" approach you suggested. I have therefore experimented a little bit with the chat api command.
My jingle_profiles/client.xml looks essentially as the one provided with the example configuration:
<profile type="client"> <param name="name" value="$${xmpp_client_profile}"/> <param name="login" value="user1@.../talk"/> <param name="password" value="XXXX"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="message" value="Jingle all the way"/> <param name="rtp-ip" value="$${bind_server_ip}"/> <param name="ext-rtp-ip" value="auto-nat"/> <param name="auto-login" value="true"/> <!-- SASL "plain" or "md5" --> <param name="sasl" value="plain"/> <!-- if the server where the jabber is hosted is not the same as the one in the jid --> <param name="server" value="talk.google.com"/> <!-- Enable TLS or not --> <param name="tls" value="true"/> <!-- disable to trade async for more calls --> <param name="use-rtp-timer" value="true"/> <!-- default extension (if one cannot be determined) --> <param name="exten" value="888"/> <!-- VAD choose one --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!--<param name="vad" value="both"/>--> <!--<param name="avatar" value="/path/to/tiny.jpg"/>--> <!--<param name="candidate-acl" value="wan.auto"/>--> <param name="local-network-acl" value="localnet.auto"/> </profile>
When I try to use the chat command at the fs_cli prompt I'm unable to form a full jid and always get an error message from the remote server. Here an example:
chat jingle|888 <at> xmppc|user2@...|test
Sent freeswitch <at> ubuntu-vbox-t5500> 2011-10-08 14:22:09.804596 [NOTICE] libdingaling.c:1373 SecSEND: ------------------------------------------------------------------------------
<message to="user2@..." type="chat" from="user1@..."> <body>test</body> </message>
2011-10-08 14:22:09.844698 [INFO] libdingaling.c:1371 SecRECV: ------------------------------------------------------------------------------
- -
<message to="user1@..." type="error" from="user2@..."> <body>test</body> <error code="400" type="modify"> <bad-request xmlns="urn:ietf:params:xml:ns:xmpp-stanzas"></bad-request> <text xmlns="urn:ietf:params:xml:ns:xmpp-stanzas">If set, the 'from' attribute must be set to the user's full JID.</text> </error> </message>
"xmppc" is the value of the variable "xmpp_client_profile" as in the default config. I get the same error independently from what I set before "@xmppc" in the from_jid. Differently from this, if as the from_jid I set my explicit jid "user1@.../talk" as in the client.xml file I get the error:
2011-10-08 14:20:50.446696 [ERR] mod_dingaling.c:544 Invalid Profile gmail.com
Despite the fact that even a wrong message reaching the remote server could work as a "keep alive" signal, I still would like to understand the proper syntax. I would be grateful if somebody could explain me what is the correct syntax to use with chat and jingle?
Thank you! Fede
P.S.: By the way, I'm now working on HEAD.
Hi Federico and Anthony, Were you guys able to come up with a solution for this? I had mine working for about a month without any problems. I'm running the git version from October 1, 2011. I just ran into the same problem after changing my router since I changed my Internet connection from DSL to fiber optic. Reading the comments from this post: http://www.personal.psu.edu/wcs131/blogs/psuvoip/2010/10/using_freeswitch_to... _google.html I guess this is an issue with Freeswitch having problems when there are Internet disconnections. I guess one way to solve this would be to change my router, but I don't really want to buy another ONT router, besides I don't even know if my ISP would let me use a different ONT since the ones that they provide are all "customized". It is really hard to tell when it has disconnected. Even when typing "dingaling status" from the fs_cli, it will show AUTHORIZED when it is disconnected. The only way to tell if it's been disconnected is checking whether I can make calls or not. To solve this ,without having to restart freeswitch I would "disconnect" (according to FS) from my gtalk profile typing "dl_logout gtalk" and reconnect with "dl_login profile=gtalk". I also had the idea to use the "cron" approach and thought this was going to be the solution to all my problems. Of course, I had to test what would happen if the cron job would happen to run in the middle of a call. I manually typed the commands on the fs_cli while I was on a call and my call did not die!! Great!! I had the solution to the problem!! I then finished the call and ooohh surprise!! I was suddenly disconnected from the fs_cli, FS would die, close unexpectedly and create a core dump. I would really like to help to solve this so let me know if there's any way I can help out. I would like to try the chat approach to see if that works. Did you figure out what is the right syntax to chat? Greetings, Ricardo
participants (6)
-
Anthony Minessale
-
curriegrad2004
-
Federico Beffa
-
king2kin
-
mazilo
-
Ricardo Monroy