March 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Wikipbx status
by Traun Leyden 18 Mar '08

18 Mar '08

The biggest incompatibility right now is with mod_voicemail, which was released shortly after wikipbx was released, and I haven't had a chance to go back and make this work. It's a high priority item though. Wikipbx comes with its own voicemail IVR, but its rather simplistic and does not support things like MWI (message waiting indicator). The other thing to be aware of is that it is heavily SIP oriented, so if you are going to be using non-SIP protocols you will probably run into some issues. Hope this helps! > I didn't know about wikipbx until today and it looks like nice from the > screenshots. Is it current with freeswitch in it's features support? > > Brian

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mod_limit
by Brian Snipes 18 Mar '08

18 Mar '08

A little background: I would like our phones to have the ability to roll a call over to a second registration on the phone if the first registration is in use. An example: line1 is registered as 317, line2 as 417. If line1 is in use I would like the call to 317 to go to 417. If 417 is in use then have it rollover to 317 voicemail. Using suggestions on the irc channel I've added extensions for limit and limit_exceeded but when a second call comes into the 317 extension it gets a hangup with the following: 2008-03-17 12:46:33 [INFO] mod_dialplan_xml.c:223 dialplan_hunt() Processing Brian Snipes->limit_exceeded@default 2008-03-17 12:46:33 [NOTICE] switch_ivr.c:982 switch_ivr_session_transfer() Transfer sofia/default/305@192.168.10.2:5060 to enum[limit_exceeded@default] 2008-03-17 12:46:34 [INFO] switch_core_state_machine.c:113 switch_core_standard_on_ring() No Route, Aborting Here is what I have cobbled together through reading and experimenting. I'm not sure if all the set/export statements are needed or not. I used the info application to try and figure out what condition to test for but I'm not doing something right. <extension name="limit" continue="true"> <condition> <action application="limit" data="inbound ${destination_number} 1"/> </condition> </extension> <extension name="limit_exceeded">      <condition field="sip_req_user" expression="^(317)$"> <action application="set" data="destination_number=417"/> <action application="export" data="destination_number=417"/> <action application="set" data="dialed_ext=417"/> <action application="export" data="dialed_ext=417"/> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="${sofia_contact($${domain}/${dialed_ext}@$${domain})}"/> </condition> </extension> Many thanks, Brian

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Re: [Freeswitch-users] IAX and CDR
by Anthony Minessale 18 Mar '08

18 Mar '08

our IAX2 support is limited. We are using our own fork of the iaxclient library from http://iaxclient.wiki.sourceforge.net/ This is the client lib used in most iax soft phones and is not suitable as-is for a server. To avoid causing commotion for the other software that uses this lib we just forked it and added our own mutexes and a few other things. (this is not a hostile fork as I have commit access on the real one. It's just because there are several experimental changes.) Even with our changes, the original lib has no trunking support and still barely qualifies to be used in a server so it's not worth it to add the support. The real solution is to write an entire IAX2 stack from scratch that actually can scale for use on a server. This is within my capability but not prudent because the IAX2 protocol is, in reality, an Asterisk Specific proprietary protocol and it wouldn't be fair for us to have a nicer IAX2 implementation than them. Not to mention, there is nearly no demand for this protocol among our user base. There was an openiax http://www.openiax.org that was underway. Maybe someday if that gets off the ground we can look at supporting it. But all signs point to the fact that we would need major motivation to expand our current iax2 support beyond what it currently is. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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 iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Michael Collins <mcollins(a)fcnetwork.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Monday, March 17, 2008 6:37:37 PM Subject: Re: [Freeswitch-users] IAX and CDR  Not entirely intuitive, but there are two references to IAX2 on this page: http://wiki.freeswitch.org/wiki/Current_Status I believe I heard in passing that FS does not currently support IAX trunking, but I’d wait for confirmation from one of the devs to be sure. -MC From: freeswitch-users-bounces@lists.freeswitch.org[mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Vijay Ramnarine Sent: Monday, March 17, 2008 4:17PM To: freeswitch-users(a)lists.freeswitch.org Subject: [Freeswitch-users] IAXand CDR How much IAX does FS support? does it support IAX trunking? How about CDR ? If there is somewhere I can just go to in order to read up about these, pleasetell me link -----Inline Attachment Follows----- _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

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Wikipbx status
by Brian Snipes 18 Mar '08

18 Mar '08

I didn't know about wikipbx until today and it looks like nice from the screenshots. Is it current with freeswitch in it's features support? Brian

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IAX and CDR
by Vijay Ramnarine 17 Mar '08

17 Mar '08

How much IAX does FS support? does it support IAX trunking? How about CDR ? If there is somewhere I can just go to in order to read up about these, please tell me link

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freeswitch Scenario
by Yogesh Patel 17 Mar '08

17 Mar '08

HI Here what we want to do with the switch. Our scenario is like few wholesale carrier will sendus traffic for different destinations, which we will be sending to thedifferent vendors. So we were using 206.222.8.58 as our live switch to testfreeswitch. So we need to understand howfreeswitch is terming following things and where do it makes them so we cankeep adding more customers and vendors. Once it starts working for us. Scenario 1 customer name “Allianz” will be sending call to us from IP 209.62.77.114. presently we can configure this customer as a SIP but we also want to do h323 there. (we don’t authenticate any customer for user/pass, they only come with IP and prefix) they will send a call to us for destinations like India(91) and Egypt(20) so we have to create appropriate dialplan for them. for India(91) calls should go to indiavendor1 on ip address 122.252.236.10 as a 1st preference and once that is fill it sld go to indiavendor2 on IP address 122.252.236.11 calls for Egypt proper (20) and Egypt Mobile(201) should go to Egyptvendor1. Scenario 2 one SIP user created on freeswiotch which we can use to make test calls to the Gateways(vendors) we add to the switch for the testing purpose. that SIP user should be able to call any existing vendors/gateways for different destinations and also new one we will add in future so I think this one will have different dialplan. Scenario 3 1. where do we have to create SIP users which we can give it to retail customersfor them to cone in any ip, means they have to have user/pass and ipprovided by us to let them register in our system and once done they can make acall out NOTE: where it will dump CDR for all above calls ? Thanks Yogesh.

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Re: [Freeswitch-users] freeswitch Scenario
by Yogesh Patel 17 Mar '08

17 Mar '08

#1 - Is your test server up and running right now? --> tets server is up and running but its on Private IP. #2 - Are you running with the default configs? --> we are running with default configured, configured wikipbx earlier with freeswitch but removed later when we found it can not help us with what we want. #3 - Have you made any changes or additions to the default configs? --> we have made few changes to make things working and did stuff here and there but nothing removed, and almost everything is same except added some more files. #4 - What revision are you on? --> we use SVN so I guess it will be the latest one. Let's get those nailed down first and then we'll take it from there. Yogesh

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TEST TO -users LIST, IGNORE
by Michael Collins 17 Mar '08

17 Mar '08

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Conceptual Question about Freeswitch and SIP
by Kurt Marasco 17 Mar '08

17 Mar '08

It's my understanding that when I call from one SIP address to another that Freeswitch manages the invites and then directly connects the two sip devices, such that freeswitch is no longer involved in the conversation. Is this how it is supposed to work? The reason for my question is that the rtp traffic is going through my freeswitch ip address at all times. It seems that both ends are speaking g711u, so I don't believe that any transcoding is going on. I must be missing something because having all data flow through Freeswitch would not be scalable and I know that one of the big differences that Freeswitch offers is scalability. What am I missing....and I know that I am:) Kurt

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404 Error on incoming calls
by Kurt Marasco 15 Mar '08

15 Mar '08

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