Hi, Anyone with an experience of SIP SIMPLE IM Messaging implemention? I dont see the stuff well documented on http://wiki.freeswitch.org/wiki/Mod_sms <http://wiki.freeswitch.org/wiki/Mod_sms> I followed simple steps to test it with reply back as in default chatplan. <?xml version="1.0" encoding="utf-8"?> <include> <context name="default"> <extension name="demo"> <condition field="to" expression="^(.*)$"> <action application="reply" data="Hello, you said: ${_body}"/> </condition> </extension> </context> </include> I tested it with eyeBeam, Jitsi, it doesnt work, and i do not see reply back. and i see in logs: 2013-03-07 14:58:49.090936 [INFO] mod_sms.c:334 Processing text message 9999->9999 in context default Chatplan: 9999 parsing [default->demo] continue=false Chatplan: 9999@198.84.61.52 Regex (PASS) [demo] to(9999@198.84.61.52) =~ /^(.*)$/ break=on-false Chatplan: 9999@198.84.61.52 Action reply(Hello, you said: ${_body}) -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/IM-messaging-on-FreeSwitch-tp7... Sent from the freeswitch-users mailing list archive at Nabble.com.
Here is the SIP traces of a message "what is this", sent to 9999 to 9999. http://pastebin.pk/sPFX6ZbX <http://pastebin.pk/sPFX6ZbX> -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/IM-messaging-on-FreeSwitch-tp7... Sent from the freeswitch-users mailing list archive at Nabble.com.
I have been using it with csimple clients and works like a charm. There were some params in my profiles mot allowing the messages to reach my leg. Maybe you can post and review your profile for 9999. On Mar 8, 2013 8:55 AM, "mehroz" <mehroz.ashraf85@gmail.com> wrote:
Here is the SIP traces of a message "what is this", sent to 9999 to 9999. http://pastebin.pk/sPFX6ZbX <http://pastebin.pk/sPFX6ZbX>
-- View this message in context:
http://freeswitch-users.2379917.n2.nabble.com/IM-messaging-on-FreeSwitch-tp7...
Sent from the freeswitch-users mailing list archive at Nabble.com.
_________________________________________________________________________ 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
Thanks Daniel. There is an update that i am having Secure signaling TLS. and in oder to deal with NAT issues , i had to put sip-force-contact=NDLB-tls-connectile-dysfunction to force rport and TLS issues. With this configuration chat does not work, BUT i just saw that* it works* when i remove the sip-force-contact parameter. :( Still for your observation, here is my profile Name 198.84.61.52 Domain Name N/A Alias Of internal Auto-NAT false DBName sofia_reg_internal Pres Hosts 198.84.61.52,198.84.61.52 Dialplan XML Context rokacomm Challenge Realm auto_from RTP-IP 198.84.61.52 SIP-IP 198.84.61.52 URL sip:mod_sofia@198.84.61.52:6000 BIND-URL sip:mod_sofia@198.84.61.52:6000 TLS-URL sip:mod_sofia@198.84.61.52:5061 TLS-BIND-URL sips:mod_sofia@198.84.61.52:5061;transport=tls HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN G722,PCMU,PCMA,GSM,AMR,H263,H264,H263-1998,H263-2000 CODECS OUT G722,PCMU,PCMA,GSM,AMR,H263,H264,H263-1998,H263-2000 TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG true PROXY-MEDIA false ZRTP-PASSTHRU true AGGRESSIVENAT false STUN-ENABLED true STUN-AUTO-DISABLE false CALLS-IN 20 FAILED-CALLS-IN 4 CALLS-OUT 10 FAILED-CALLS-OUT 4 REGISTRATIONS 3 and 9999 profile as: Call-ID: c6dfab23429f90be4a57c6c39a3fd3ef@0:0:0:0:0:0:0:0 User: 9999@198.84.61.52 Contact: "9999" <sip:9999@175.110.122.81:6000;transport=udp;registering_acc=198_84_61_52;fs_nat=yes> Agent: Jitsi1.0-build.3967Windows 7 Status: Registered(AUTO-NAT)(unknown) EXP(2013-03-08 15:07:12) EXPSECS(87) Host: rokacomdev1 IP: 175.110.122.81 Port: 6000 Auth-User: 9999 Auth-Realm: 198.84.61.52 MWI-Account: 9999@198.84.61.52 Anything else you need to see? But that is sure , the issue is due to sip_force_contact parameter in directory! -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/IM-messaging-on-FreeSwitch-tp7... Sent from the freeswitch-users mailing list archive at Nabble.com.
Yeah, now that is the the param that gave me headache as well. Contact mr off-list if u need offline message support. I carved a perl and lua config to queue messages if leg is not registered and deliver as soon as it reappears. Didnt find a stock way to do it. On Mar 8, 2013 4:12 PM, "mehroz" <mehroz.ashraf85@gmail.com> wrote:
Thanks Daniel.
There is an update that i am having Secure signaling TLS. and in oder to deal with NAT issues , i had to put sip-force-contact=NDLB-tls-connectile-dysfunction to force rport and TLS issues. With this configuration chat does not work, BUT i just saw that* it works* when i remove the sip-force-contact parameter. :(
Still for your observation, here is my profile
Name 198.84.61.52 Domain Name N/A Alias Of internal Auto-NAT false DBName sofia_reg_internal Pres Hosts 198.84.61.52,198.84.61.52 Dialplan XML Context rokacomm Challenge Realm auto_from RTP-IP 198.84.61.52 SIP-IP 198.84.61.52 URL sip:mod_sofia@198.84.61.52:6000 BIND-URL sip:mod_sofia@198.84.61.52:6000 TLS-URL sip:mod_sofia@198.84.61.52:5061 TLS-BIND-URL sips:mod_sofia@198.84.61.52:5061;transport=tls HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN G722,PCMU,PCMA,GSM,AMR,H263,H264,H263-1998,H263-2000 CODECS OUT G722,PCMU,PCMA,GSM,AMR,H263,H264,H263-1998,H263-2000 TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG true PROXY-MEDIA false ZRTP-PASSTHRU true AGGRESSIVENAT false STUN-ENABLED true STUN-AUTO-DISABLE false CALLS-IN 20 FAILED-CALLS-IN 4 CALLS-OUT 10 FAILED-CALLS-OUT 4 REGISTRATIONS 3
and 9999 profile as: Call-ID: c6dfab23429f90be4a57c6c39a3fd3ef@0:0:0:0:0:0:0:0 User: 9999@198.84.61.52 Contact: "9999" <sip:9999@175.110.122.81:6000 ;transport=udp;registering_acc=198_84_61_52;fs_nat=yes> Agent: Jitsi1.0-build.3967Windows 7 Status: Registered(AUTO-NAT)(unknown) EXP(2013-03-08 15:07:12) EXPSECS(87) Host: rokacomdev1 IP: 175.110.122.81 Port: 6000 Auth-User: 9999 Auth-Realm: 198.84.61.52 MWI-Account: 9999@198.84.61.52
Anything else you need to see?
But that is sure , the issue is due to sip_force_contact parameter in directory!
-- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/IM-messaging-on-FreeSwitch-tp7... Sent from the freeswitch-users mailing list archive at Nabble.com.
_________________________________________________________________________ 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
Daniel Ivanov wrote
Yeah, now that is the the param that gave me headache as well. Contact mr off-list if u need offline message support. I carved a perl and lua config to queue messages if leg is not registered and deliver as soon as it reappears. Didnt find a stock way to do it.
Hi Daniel, I sent you an email directly yesterday since I'm also interested in your perl and lua script for IM handling. I can't seem to find any stock way in FS to handle SIMPLE messaging to an offline user. Ideally I'd like to have a store and forward function so that the message gets delivered when he comes back online, similar to plain old GSM SMS. Thanks for any suggestions -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/IM-messaging-on-FreeSwitch-tp7... Sent from the freeswitch-users mailing list archive at Nabble.com.
Hi, I would like to use the Dialplan configuration to control which gateway is used on a conference dialout. I have a gateway configured in the Dialplan, to match any number. I have tried this on the dialout: conference <name> dial {originate_timeout=30}sofia/external/8005551212 12345 CallerName but I get the errors Cannot locate registered user 8005551212@external mod_conference.c:6775 Cannot create outgoing channel, cause: USER_NOT_REGISTERED I don't want to specify the gateway in the dial request. I want the Dialplan to choose it. What am I doing wrong? Thanks, David
I don't know if this will work - haven't tried it in that scenario, but you might try: loopback/8005551212 -- Nathan On 09/09/2013 11:02 AM, D D wrote:
Hi,
I would like to use the Dialplan configuration to control which gateway is used on a conference dialout.
I have a gateway configured in the Dialplan, to match any number.
I have tried this on the dialout:
conference <name> dial {originate_timeout=30}sofia/external/8005551212 12345 CallerName
but I get the errors Cannot locate registered user 8005551212@external mod_conference.c:6775 Cannot create outgoing channel, cause: USER_NOT_REGISTERED
I don't want to specify the gateway in the dial request. I want the Dialplan to choose it. What am I doing wrong?
Thanks, David
-- ------------------------------------------------------------ Nathan Neulinger nneul@mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect
https://wiki.freeswitch.org/wiki/Sofia-SIP#Dial_out_of_a_gateway On Sep 9, 2013, at 12:02 PM, D D <tru083@yahoo.com> wrote:
Hi,
I would like to use the Dialplan configuration to control which gateway is used on a conference dialout.
I have a gateway configured in the Dialplan, to match any number.
I have tried this on the dialout:
conference <name> dial {originate_timeout=30}sofia/external/8005551212 12345 CallerName
but I get the errors Cannot locate registered user 8005551212@external mod_conference.c:6775 Cannot create outgoing channel, cause: USER_NOT_REGISTERED
I don't want to specify the gateway in the dial request. I want the Dialplan to choose it. What am I doing wrong?
On the 'dial' command, I want to specify only the destination number, not the gateway. I reviewed the documentation at https://wiki.freeswitch.org/wiki/Sofia-SIP#Dial_out_of_a_gateway, but I want the Dialplan configuration to choose the gateway. How can I do that? ________________________________ From: Michael Jerris <mike@jerris.com> To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Sent: Monday, September 9, 2013 11:19 AM Subject: Re: [Freeswitch-users] How can I have the DIalplan choose the gateway for conference dialout? https://wiki.freeswitch.org/wiki/Sofia-SIP#Dial_out_of_a_gateway On Sep 9, 2013, at 12:02 PM, D D <tru083@yahoo.com> wrote: Hi,
I would like to use the Dialplan configuration to control which gateway is used on a conference dialout.
I have a gateway configured in the Dialplan, to match any number.
I have tried this on the dialout:
conference <name> dial {originate_timeout=30}sofia/external/8005551212 12345 CallerName
but I get the errors Cannot locate registered user 8005551212@external
mod_conference.c:6775 Cannot create outgoing channel, cause: USER_NOT_REGISTERED
I don't want to specify the gateway in the dial request. I want the Dialplan to choose it. What am I doing wrong?
_________________________________________________________________________ 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
I believe someone already answered that. Use "loopback/NUMBER" as the dial-out string. This will create a loopback call that whits the dialplan and then follows the usual stuff in there. /Peter 9 sep 2013 kl. 20:06 skrev D D <tru083@yahoo.com>: On the 'dial' command, I want to specify only the destination number, not the gateway. I reviewed the documentation at https://wiki.freeswitch.org/wiki/Sofia-SIP#Dial_out_of_a_gateway, but I want the Dialplan configuration to choose the gateway. How can I do that? ------------------------------ *From:* Michael Jerris <mike@jerris.com> *To:* FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> *Sent:* Monday, September 9, 2013 11:19 AM *Subject:* Re: [Freeswitch-users] How can I have the DIalplan choose the gateway for conference dialout? https://wiki.freeswitch.org/wiki/Sofia-SIP#Dial_out_of_a_gateway On Sep 9, 2013, at 12:02 PM, D D <tru083@yahoo.com> wrote: Hi, I would like to use the Dialplan configuration to control which gateway is used on a conference dialout. I have a gateway configured in the Dialplan, to match any number. I have tried this on the dialout: conference <name> dial {originate_timeout=30}sofia/external/8005551212 12345 CallerName but I get the errors Cannot locate registered user 8005551212@external mod_conference.c:6775 Cannot create outgoing channel, cause: USER_NOT_REGISTERED I don't want to specify the gateway in the dial request. I want the Dialplan to choose it. What am I doing wrong? _________________________________________________________________________ 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 (7)
-
D D
-
Daniel Ivanov
-
mehroz
-
Michael Jerris
-
Nathan Neulinger
-
Peter Olsson
-
privs