August 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Using freeswitch as Generic SIP SBC/RtpProxy without predefined endpoints
by Taisto Qvist 14 May '16

14 May '16

Hi, I am an OpenSIPS/Kamailio user that has spent the whole day trying to figure out which software is most suitable to build an SBC that only acts as a full b2bua with rtp-relay, with an undefined number of destinations(entire domains with lots of unknown proxies) on each side. I've been looking at yate, freeswitch, SEMS, but I am overwhelmed by the gazillions of configuration-files that seem to be needed. I have absolutely NO need of pbx/user functionality, just a B2BUA and rtp-proxy capabilities(with full codec-transparency), with the addition that I must be able to add some custom logic to decide the DNS or IP(v6) address of the next-hop sip-node. All examples I've found on freeswitch SBC config, has predefined gateways/trunks/proxies as "predefined next-hops", and I want the SBC to be able to send anywhere, depending on custom application logic. I'd stick with kamailio/rtp-proxy and the standard topology-hiding module, but unfortunately I need the full topology hiding of a b2bua. Is this easily achievable with Freeswitch? Any help with be immensely appreciated! Regards Taisto Qvist

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Mastering FreeSWITCH Book
by Anthony Minessale 09 Apr '16

09 Apr '16

Hi, We've been asked by packt to produce a 3rd book called "Mastering FreeSWITCH" Would anyone be interested in working on the authoring of that book and get their name in print? We need help with creating the content as well as formatting the text into the word templates they require etc. Please contact consulting(a)freeswitch.org if you are interested. -- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>* ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900

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Is MySQL OK to use?
by Alex Lake 06 Apr '16

06 Apr '16

I find these "Error in my_thread_global_end()" messages somewhat annoying in my fs1.2stable on Ubuntu12.04 box. Is there any advice (other than "don't use MySQL") for how to install it better? Might it be something to do with thread-safe libraries (I'm using libmyodbc_r.so at the moment)

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FS, SS7 and Dialogic IMG2020 in the UK
by Gavin Henry 15 Mar '16

15 Mar '16

Hi all, We're looking to get a pair of Dialogic IMG2020 for some BT Ofcom regulated SS7 interconnects: http://www.dialogic.com/en/products/gateways/img/img2020.aspx Any issues known with FS behind these? Thanks, Gavin. -- http://www.surevoip.co.uk

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Mod_callcenter ODBC vs File
by Jai Rangi 11 Feb '16

11 Feb '16

Hello, FS version: FreeSWITCH Version 1.4.18+git~20150312T185523Z~4eed221b69~64bit (git 4eed221 2015-03-12 18:55:23Z 64bit) I found a very strange behavior on mod_call center when used with odbc. This was easy to reproduce when the agents are busy and new calls come in. New member goes in abandoned state immediately and stay in same state forever. Even if previous call is complete and Agent is in Waiting state. Infact some times new members go in abandoned state immediately even if the agents are available. Everything works perfect once I changed the mod_callcenter.conf.xml config to file only mode, new calls go in Waiting state and call is connected to agent as soon as the agent is free. Any idea why the new member goes in abandoned state randomly in place of being in waiting state? Can it be due to too small timeout in call center module as it query the DB every time? Can that be changed? Anyone else had smiler problem, any experience or any recommendations will be much appreciated. Thank you, -Jai

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Random calls failing with WRONG_CALL_STATe when using TLS
by Emrah 26 Jan '16

26 Jan '16

Hi all, I am facing a very frustrating issue. I often have to dial twice when using my Yealink phone with TLS because the first attempt times out. The logs on the Yealink indicate that the first invite is successfully received, to which my FS sends a 100 trying and 407 proxy auth required. It is subsequently when my phone sends back the invite that the connection crashes with the following error: SSL ERROR SYSCALL Is this something common? Why does the SSL connection crashes when the phone attempts to send the second invite? My phone is behind NAT. It is going to be a crazy expedition to collect the logs and Pastebin them, so I am tempting my luck on the list first to see if you have any pointers. As a last piece, my Bria on my iPHone, among other clients, never had this issue. I did experience it from time to time with Blink on Mac OS X. Any help appreciated. Emrah

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Re: [Freeswitch-users] Event socket and Python
by Johan Brannlund 17 Jan '16

17 Jan '16

Sami Montour wrote: > Sorry I misunderstood your post about testing with a closed port. > Anyway, I ran your script (port 8041) but didn't get a connection to > FS, different from what you received. Hmm, that's interesting. I guess either I happened to check out a buggy version of FS, there's something strange with my configuration or there's some ARM-specific issue (I run FS on a Raspberry Pi). I'll have to do some more troubleshooting. Thank you for checking! - Johan

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outbound proxy and port
by Chris B. Ware 05 Jan '16

05 Jan '16

Hi, I need to send registrations and calls from external profile to a si provider, passing through an outbound proxy, port 5080. I I set on sofia: <param name="outbound-proxy" value="10.24.4.151"/> Registrations pass through the OB Proxy, but using, of course, standard 5060 port. I I set on sofia: <param name="outbound-proxy" value="10.24.4.151:5080"/> Param seems to be not understood and registrations go directly to the sip provider. How can I set outbound proxy port? Thanks

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How to enable Screen sharing on Verto Conf?
by Victor Medina 23 Nov '15

23 Nov '15

Hi guys! I been playing around with the verto conference demo on 1.6. How do I enable the screen sharing feature? -- Víctor E. Medina M. Platform Architect / Chief Infrastructure +58424 291 4561 BB #79A8AFA2 @VMCibersys

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Break unaswered bridge
by Alexey Melnichuk 21 Oct '15

21 Oct '15

This is simple dialplan that I try to implement <extension> <condition field="destination_number" expression="^\d+$" break="on-true"> <action application="bind_digit_action" data="conf,*1,exec:execute_extension,xfer"/> <action application="digit_action_set_realm" data="conf"/> <action application="conference" data="100@default"/> </condition> <condition field="destination_number" expression="^xfer$" break="on-true"> <action application="bind_digit_action" data="xfer,*1,?????????"/> <action application="digit_action_set_realm" data="xfer"/> <action application="digit_action_set_realm" data="xfer"/> <action application="read" data="2 11 'tone_stream://%(10000,0,350,440)' target_num 30000 #"/> <action application="bridge" data="user/${target_num}@${domain_name}"/> </condition> </extension> I have two problem with this dialplan 1. in secon `bind_digit_action` I can not use `exec` only `api`. e.g. <action application="bind_digit_action" data="xfer,*1,exec:info"/> does not work. Application executes only when I return from first `exec:execute_extension` 2. I can not cancel bridge before answer I can not use transfer because I do not whant leave conference (e.g. user has `end_conference=true` flag). I try use uuid_kill from Lua like "xfer,*1,lua:kill_legs.lua ${uuid}" -- kill_legs.lua local uuid = argv[1] local sql = ("select uuid from channels where call_uuid='%s' and uuid<>'%s'") :format(uuid, uuid) dbh:query(sql, function(row) local res = api:executeString("uuid_kill " .. row.uuid) end) But this is does not work because it kills only active channels, but for dial-string like `loopback/111/my.domain,[leg_delay_start=20]loopback/222/my.domain` second channels creates only after 20 sec. And because I may have more then one channels I can not use `originate_uuid` In documentation I found `break` application but as I say I can not use it. `uuid_break` api break only play audio.

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