February 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Broadsoft SCA w/ Cisco SPA-5xx series
by Gabriel Kuri 17 Feb '10

17 Feb '10

I followed Brian's directions from one of the previous threads on configuring the SPA-5xx series phones for Broadsoft SCA and set manage-shared-appearance=true in the internal profile. SCA appears to be working on outgoing calls between two phones, the line key starts flashing red on the second phone when the first phone picks up the receiver to make a call. However on incoming calls, both phones ring (same extension), however when one of the phones picks up the line, the second phone's line key doesn't flash red or show the first phone on that incoming call. Any ideas? Does shared appearance only work on outgoing phone calls? Thanks, Gabe

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Multiple versions of FreeSWITCH on one box
by Adam Ford 17 Feb '10

17 Feb '10

According to the wiki it is possible to run multiple FreeSWITCH instances on one box. Are there any known issues with having those instances be different versions? Unfortunately I don't have a separate box to test the migration from 1.0.4 to 1.0.5, so I am looking to build a separate instance of 1.0.5 on the same box. -Adam

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external_sip_address and external_rtp_address Question
by Jerry Richards 16 Feb '10

16 Feb '10

I only see one example for setting of external_sip_address and external_rtp_address tags. Is it true they are used to specify a SIP provider outside of a LAN (i.e. through a router)? If so, then can these tags be set for each sip_profile? So, if I have multiple external SIP providers that are accessed through NAT, they would each have their own external_sip_address and external_rtp_address? Best Regards, Jerry

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event-socket outbound: Dialplan failover on socket error?
by peely 16 Feb '10

16 Feb '10

Hi, I'd like to implement a dialplan entry to use a secondary event-socket application if the first esl server is down or could not be connected to. Could somebody please tell me what continue_on_fail options I could set so that I will continue ONLY if the event-socket outbound connection was unsuccessful i.e. an "[ERR] mod_event_socket.c:414 Socket Error!"? By default it seems to continue if the outbound bridge is unsuccessful however I don;t want to do that. Alternatively, is there something I can set from the event-socket session to stop the continue_on_fail from failing over? Thanks, Neil. -- View this message in context: http://old.nabble.com/event-socket-outbound%3A-Dialplan-failover-on-socket-… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Version 1.05 release
by Goran Donev 16 Feb '10

16 Feb '10

I really didn't get a definitive answer on when 1.05 is slated to be released. We are running into some issues that I hope that 1.05 fixes. Do we have an eta? Thx

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mod_zeroconf
by Ivan C Myrvold 16 Feb '10

16 Feb '10

How can I build FreeSWITCH with mod_zeroconf? I can't find any mod_zeroconf to uncomment in modules.conf Ivan

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Call Monitoring and Recording with FS?
by Thomas Switch 16 Feb '10

16 Feb '10

Hello FreeSWITCH folks, I was asked to join a project in the VoIP field. Being a newbie to VoIP, I read a couple of books, many web pages and came across FreeSWITCH. Hope you don't mind answering two questions of mine: a) Similar to a call centre application, I'd need to record *all*conversation. Like "For quality assurance, all conversations will be recorded..."... Could I do that with FS or do I need an additional piece of software or hardware? If I cannot record all conversation easily, I might be able to water the requirement down to an "Operator monitoring a call, can record the call on demand" (see also the second question)? What about that? b) As an operator, I need to be able to monitor any call. I understand, that one can get the active connection from FS. Is there a possibility to get into these calls? Or do I need to hack a "standard" call silently into a conference call with the operator? If yes, is it possible to do that without the participants in the call noticing it? Thanks a lot for your time and patience. Daniel

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DTMF problem
by Maxim Tsvetov 16 Feb '10

16 Feb '10

Hello everybody Please help! I'm trying to setup connection between Cisco 2811 and FS (Win2003) using SIP. Everything working correctly and I can make calls both ways. The only problem is when I'm calling from PSTN to FS over Cisco. It doesn't recognize DTMF. I use G711 a-law (PCMA) codec and inband DTMF. Regards, Maxim

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h323 - sip call is not working
by Nikolay Kondratyev 16 Feb '10

16 Feb '10

Hi all, I compiled FreeSWITCH Version 1.0.5-20100209-0400 (16587M) with mod_h323. When I call from h323 (Avaya IPOffice) to local fs extention (x-lite) I hear ring back, but when x-lite picks up, he hears silence, while IPOffice user continues to hear ringback. The log is at the http://pastebin.freeswitch.org/12091 My configuration is almost default, several local extentions added, and h323.conf from http://wiki.freeswitch.org/wiki/Mod_h323 5840 - user at IPOffice 2853 - x-lite registered at FS IPOffice ip address: 172.23.14.2 FS ip address 172.23.22.49 Can anybody please advise how to solve that? Is it a configuration or a software problem? Thanks in advance, Nikolay.

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FreeSWITCH Debian packages - build issues
by Jason White 16 Feb '10

16 Feb '10

Has anyone successfully built the Debian packages recently from the source repository? The problem I'm experiencing is that openzap is specified to be built, but it is never actually compiled. Consequently, the packages can't be created (the process fails due to the missing mod_openzap.so file). I don't need openzap; I can easily comment it out, but I also think the supplied package files should work as is. first step: confirm whether my experience under Debian Sid is shared by others using different versions of Debian or Ubuntu.

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