Hello,
We are trying to make some kind of a gateway between SIP and XMPP domain. We have an environment with FreeSWITCH and OpenFire (FreeSWITCH is registered as a component in OF). Calls from XMPP to SIP are working fine. The problem is with calls from SIP to XMPP. The caller (SIP) has a ringing tone, a the callee (XMPP) has a 'connecting...' message after answer the call. Do you have any experience in making calls between SIP and XMPP users? Is it possible to establish the video calls also?
Best Regards,
Michał Muszalski Senior Specialist
Orange Labs Poland Obrzeżna 7, 02-691 Warsaw
Muszalski Michał - Korpo TP wrote:
Hello,
We are trying to make some kind of a gateway between SIP and XMPP domain. We have an environment with FreeSWITCH and OpenFire (FreeSWITCH is registered as a component in OF). Calls from XMPP to SIP are working fine. The problem is with calls from SIP to XMPP. The caller (SIP) has a ringing tone, a the callee (XMPP) has a 'connecting...' message after answer the call.
I see this problem with a self-built FS git as of 01/26/2011, but not on a FreeSWITCH Version 1.0.head (git-cf253c3 2011-01-11 15-31-40 -0600). ----- don't and stop are the ONLY two 4-letter words considered offensive to men, but not when used together. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/PD-SIP-XMPP-calls-tp5966636p59... Sent from the freeswitch-users mailing list archive at Nabble.com.
Hello! I've installed FreeSWITCH Version 1.0.head (git-cf253c3 2011-01-11 15-31-40 -0600) and still have the same problem :( Any ideas? Regards, Michal -----Wiadomość oryginalna----- Od: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] W imieniu mazilo Wysłano: 27 stycznia 2011 17:25 Do: freeswitch-users@lists.freeswitch.org Temat: Re: [Freeswitch-users] PD: SIP -> XMPP calls Muszalski Michał - Korpo TP wrote:
Hello,
We are trying to make some kind of a gateway between SIP and XMPP domain. We have an environment with FreeSWITCH and OpenFire (FreeSWITCH is registered as a component in OF). Calls from XMPP to SIP are working fine. The problem is with calls from SIP to XMPP. The caller (SIP) has a ringing tone, a the callee (XMPP) has a 'connecting...' message after answer the call.
I see this problem with a self-built FS git as of 01/26/2011, but not on a FreeSWITCH Version 1.0.head (git-cf253c3 2011-01-11 15-31-40 -0600). ----- don't and stop are the ONLY two 4-letter words considered offensive to men, but not when used together. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/PD-SIP-XMPP-calls-tp5966636p59 66876.html 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
Hi, I have the similar problem. I want to make conversation between SIP and XMPP user, I have configured that they can see together, I can initialize and stop the call betweens two these users. This is my context: - Caller: A, a freeSWITCH SIP user - Called: B, a openfire XMPP user. - Client tools: Jitsi 1. A initialize a call to B 2. A's status is connecting 3. B's status is incoming call 4. B accept the call 5. B's status is connected 6. A's status is still connecting... I have the same problem if B initialize call to A I tried to change the openfire xmpp user by google account, there are two results: 1. If A still use jitsi, B use google web, the result are very good. I work both voice and video. 2. If A still user jitsi, B user google talk, the result are the same as openfire user. I tried to configured the codec. FreeSwitch media mode but it still happened. If i make call between two SIP accounts or two XMPP account, it work well with jitsi. Anyone understands the problems? Thanks in advance! My Server: Linux localhost.localdomain 2.6.32-279.el6.i686 #1 SMP Fri Jun 22 10:59:55 UTC 2012 i686 i686 i386 GNU/Linux. My FS version: FreeSWITCH Version 1.3.0+git~20120911T090940Z~a478a5de04 (1.3.0; git at commit a478a5de04 on Tue, 11 Sep 2012 09:09:40 Z) -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/PD-SIP-XMPP-calls-tp5966636p75... Sent from the freeswitch-users mailing list archive at Nabble.com.
Muszalski Michał - Korpo TP <Michal.Muszalski@...> writes:
Hello,
We are trying to make some kind of a gateway between SIP and XMPP domain.
We have an environment with FreeSWITCH and OpenFire (FreeSWITCH is registered as a component in OF). Calls from XMPP to SIP are working fine. The problem is with calls from SIP to XMPP. The caller (SIP) has a ringing tone, a the callee (XMPP) has a 'connecting…' message after answer the call.
Do you have any experience in making calls between SIP and XMPP users? Is
it possible to establish the video calls also?
Best Regards,
Michał Muszalski
Senior Specialist
Orange Labs Poland
Obrzeżna 7, 02-691 Warsaw
Hello Muszalski Michał - Korpo TP I have a project to do in my training. I am trying to make a gateway between SIP and XMPP domain too. I have also an environment with FreeSWITCH and OpenFire (FreeSWITCH is registered as a component in OF). but Calls from XMPP to SIP are not working, can you please help me, here is my configuration: 1)Openfire: Server Settings -> External Components Enable and set port + secret 2)Freeswitch: add the port+secret to the file "server.xml", mod_dingaling is enabled. 3) client : Jitsi : add sip client (1000@vm-cu-asterisk.wechat.jve) to xmpp client (test@wechat.jve) i got this: Authenticated.Come to ClueCon! When i try to call it: i got this error: Failed to create a call to 1000@vm-cu-asterisk.wechat.jve/talk I look at the logs in openfire, I've got nothing, this don't help to found out what's the problem. I hope you can tell me how you can make calls from XMPP to SIP. Thank you
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Muszalski Michał - Korpo TP