- FreeSWITCH-users - FreeSWITCH Mailing Lists

Ask about the integration between xmpp and freeswitch
by atai＠unseen.is 14 Mar '16

14 Mar '16

Hello all, We tried to setup the FS and can call using SIP protocol, using different apps for ex. xLite, it works very well! And we tried to setup the FS with xmpp. If we have many xmpp users, and we want users to use SIP to call, so we tried to install the dingaling module, configured them, but when we create the xmpp users, there is no notification from xmpp to FS. Is there any full instruction of configure XMPP in FS and example? Thank you very much. Tai Duc.

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mod_shout build problem
by Mi Ke 13 Mar '16

13 Mar '16

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Freeswitch IVR: call drop on transfer to busy extensions
by Francesco Piccinin 13 Mar '16

13 Mar '16

Hi all, I'm experiencing an issue with IVR on production Freeswitch cluster. I'm going to explain the problem as clear as possible just to understand if anyone already get it and hopefully fix it. Phone A makes a call that match an IVR services: call is transfered to context services in xml services. Call is then answered, audio menù is prompted: phone A makes a selection (ex dtmf 2) and the call is trasfered to context default xml default in order to ring phone B. If phone B is in busy state the call is dropped. On cli logs you can observe this behaviour: Mod Sofia first hangup the channel with a USER BUSY cause and then it sends a BYE to Phone A (or to a gateway in case of external calls) including Q850 cause=17 user busy. Phone A or Gateway drop the call without busy tone. In this case I'm expecting a SIP 486 (user busy message) instead of a bye, is that correct? I'm using the app "limit" in order to count active calls on extensions and store them in system db (shared between freeswitch servers). Hope you can help me solving the problem. Thanks Regards Francesco

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how to reply here in list to existing message and not open new thread?
by Dmitry Saratsky 13 Mar '16

13 Mar '16

test hope reply is working now... sorry if not...

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how to reply here in list to existing message and not open new thread?
by Dmitry Saratsky 13 Mar '16

13 Mar '16

Hi, Probably it is most stupid question... But It's my 3rd time I'm trying to reply to existing message, but it is new thread opened here in lists... Please help...

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Empty SDP in 200 OK on received re-invite while faxing with T.38
by Dmitry Saratsky 13 Mar '16

13 Mar '16

Yesterday I have updated to latest git to see if issue is still exists. Will update later

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Thursday FreeSWITCH Bug Hunt!
by Ítalo Rossi 10 Mar '16

10 Mar '16

FreeSWITCHers, Join us TODAY 2PM CST for the Thursday FreeSWITCH Bug Hunt! Where? [conference.freeswitch.org/vc/#/?autocall=888](https://conference.frees witch.org/vc/#/?autocall=888 "https://conference.freeswitch.org/vc/#/?autocall=888" ) What? FreeSWITCH Bug Hunt, Jira Reviews, and General FS Support! Help us help you, Join the Bug Hunt! Ítalo Rossi italo(a)freeswitch.org IRC chat.freenode.net #freeswitch #freeswitch-dev Bugs? https://freeswitch.org/jira Docs? https://freeswitch.org/jira Chat? https://hipchat.freeswitch.org/gUdAgy0m6

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FreeSwitch AMQP Question
by SamyGo 10 Mar '16

10 Mar '16

Hi All, I am experimenting with mod_amqp , I've a RabbitMQ server deployed. What I'm trying to find out is if N number of FreeSwitch Servers are connected to the same virtualhost to send and receive events and all FreeSwitch servers are connected to same backend Database. In that particular situation if one server sends an Update status of a SIP endpoint to the MessageBus and rest of the FS receives them, will they also consider that SIP endpoint as Busy/On-Call ? Another example could be change in call_centre agent state change. If one FS sends an event stating the agent is busy, will rest of the FS servers honour this and not assign calls to the agent until a free event is received !! Thanks, Sammy

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FreeTDM Channel Driver for Asterisk
by Ronaldo Borges 10 Mar '16

10 Mar '16

Hi all, I am a new user of FreeSWITCH and I passed the last few days studying the code from FreeTDM module. I am also a new user of Asterisk and I also studied the chan_dahdi code. At the FreeTDM wiki page I read that FreeTDM (FreeSWITCH) is equivalent to chan_dahdi (Asterisk). I considered FreeTDM code much better written than chan_dahdi. I was thinking about doing a "chan_freetdm" (FreeTDM Channel Driver for Asterisk). Anyone could tell if is it possible? If not, why? Is there a rocky road to this work? Best regards, -- Ronaldo.

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Empty SDP in 200 OK on received re-invite while faxing with T.38
by Dmitry Saratsky 10 Mar '16

10 Mar '16

Hi all, I found that some of my faxes failed to be sent/received bcz remote PBX drop the calls after my FreeSWITCH send empty (without SDP) 200 OK message on received re-invite from remote party while sending or receiving fax with T.38. CLIENT > INVITE (G.711) > FS FS > 200 OK > CLIENT CLIENT > INVITE (T.38) > FS FS > 200 OK > CLEINT ... now receiving fax while fax is received, client send me reinvite with the same SDP T.38 details as in initial T.38 INVITE message. usually it happens after 900 or 1800 seconds - probably it kind of keepalive... FS > 200 OK (with no SDP) > CLEINT now call is dropped( freeswitch: Version 1.5.15b git 3d64402 2015-03-21 19:55:21Z 64bit centos 6 x64 any ideas?

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