- FreeSWITCH-users - FreeSWITCH Mailing Lists

Make call from Ekiga phone via freeswitch
by Navnath Sonavne 02 May '13

02 May '13

Hi All, I am able to establish audio call from X-Lite(sip protocol) soft phone to Ekiga(h323 protocol) softphone via freeswitch. I can hear voice from both sides. Now this call making is one way i.e. i can make call only from X-Lite softphone because it is registered on freeswitch. Now I want to make call from Ekiga softphone to X-Lite softphone via freeswitch. My first query is How to register Ekiga phone on freeswitch? Once phone is registered on freeswitch How to make call to X-Lite phone? -Navnath.

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freetdm RBS configuration for multi-port cards
by bedgar＠vseinc.com 01 May '13

01 May '13

Have an issue for freetdm experts: 1. I have it working with one port but what is the proper syntax for multiple Sangoma RBS T1 ports in freetdm.conf and freetdm.conf.xml? The installation scripts wancfg_fs and wancfg_dahdi put them in the PRI type notation and cannot find how to rework for multiple RBS. Here is a working stripped down to only one port. This is after going through script and only setting one port as used. Current: freetdm.conf [span zt wp1] trunk_type => EM em-channel => 1-24 freetdm.conf.xml <configuration name="freetdm.conf" description="Freetdm Configuration"> <settings> <param name="debug" value="0"/>    </settings> <analog_em_spans> <span name="wp1"> <param name="dialplan" value="XML"/> <param name="context" value="public"/> </span> </analog_em_spans> </configuration> Thanks for looking and helping if you can! Brian C. Edgar, Jr.

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Erlang training at ClueCon 2013 - would you be interested?
by Michael Collins 01 May '13

01 May '13

Hello all! As you know, ClueCon 2013 is coming up fast. Lots of things are happening behind the scenes, including preparations for possible pre- and post-cluecon training sessions. One topic of consideration is Erlang. We want to gauge the interest level in having a training session on Erlang. If you are definitely interested in attending an Erlang training session at ClueCon 2013 please let us know. Reply off-list to marketing(a)cluecon.com and CC james(a)2600hz.org. Thanks! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

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Re: [Freeswitch-users] BT claims licensing on SIP
by Usama Zaidi 01 May '13

01 May '13

+1 -----Original Message----- From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of freeswitch-users-request(a)lists.freeswitch.org Sent: Wednesday, May 01, 2013 7:44 PM To: freeswitch-users(a)lists.freeswitch.org Subject: FreeSWITCH-users Digest, Vol 83, Issue 4 Send FreeSWITCH-users mailing list submissions to freeswitch-users(a)lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request(a)lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner(a)lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..."

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Retransmission resolves new ip address
by Chris Martineau 01 May '13

01 May '13

Hi, Have a scenario whereby FS sends an invite to another downstream sip server which is late replying and fs then sends a re-transmission, however fs then subsequently gets a reply. The problem I have is that the downstream server is on a dns-srv roundrobin between 2 servers and the re-transmission is resolving a different server. This seems to be causing the replies that have subsequently come in from the original server to just bounce until they timeout. The re-transmission to the other server seems to carry on fine but sometimes generates a 482 merged call error. The main effect is just messy call stats as I get strange 483 errors generated by fs for some reason internally for the first call and also a normal error or cdr for the re-transmitted part of the call. If I set the downstream server to a fixed ip everything works fine. Should the retransmission do a new dns lookup? If it gets a new ip address should fs handle the dialog on the old ip address? I may be just not setting something correctly but cannot find any references to this problem in any blogs, any ideas. Many thanks Chris

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FreeSWITCH Weekly Conference Call
by Michael Collins 01 May '13

01 May '13

We look forward to our call today: http://wiki.freeswitch.org/wiki/FS_weekly_2013_05_01 We have Omar from OrecX who will be discussing call recording systems. Talk to you soon! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

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Early Media with park ()
by David Wilkie 01 May '13

01 May '13

I'm using Adhearsion to bridge calls together which results in running the following command in FreeSwitch: bgapi originate {return_ring_ready=true,origination_uuid=c7f279eb-eec8-458a-b6e4-89ee1018567f,origination_caller_id_number='+14152345678,originate_timeout=30}sofia/gateway/didlogic/85512345678 &park() Leg B's (in this case 85512345678) phone rings but I cant hear any ringback signal. I tried this out in another FreeSwitch dialplan using bridge e.g. <action application="bridge" data=" {return_ring_ready=true,origination_uuid=c7f279eb-eec8-458a-b6e4-89ee1018567f,origination_caller_id_number='+14152345678,originate_timeout=30}sofia/gateway/didlogic/$${test_number}"/> And I can hear the ringback tone, so I'm assuming it has something to to with the park(). Does anyone know how to get the ringback signal to work in the above case like Adhearsion generates?

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User settings in directory for unauthenticated users
by Siri MM 01 May '13

01 May '13

Hi, I have a system where I use Open ACL, and allow users to register without any passwords required (one reason being all are trusted devices withing the closed network) However, I would like to restrict outgoing calls, and was hoping to use the 'toll_allow' variable, and have set the same for the user in the directory. However, in the dialplan, I see that toll_allow is not set for the extension - when I experimented by removing the open acl and forcing authentication, toll_allow is set. My question is, if this is expected behavior, is there any other way I could pass on user specific settings to the dialplan in an OpenACL environment? Thanks!

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BT claims licensing on SIP
by Cal Leeming [Simplicity Media Ltd] 01 May '13

01 May '13

Hello, Thought I'd share this with you all... So there I was, quietly enjoying myself on theregister, when I stumbled upon what can only be described, as a complete and utter fail. http://www.theregister.co.uk/2013/04/30/bt_trolling_sip_in_battle_with_goog… Responses so far? <@SwK> lololol * foxx2 facepalm <coppice> BT come up with a regular stream of patents with no basis in reality <mcrane> they shouldn't be able to patent a standard protocol I'm fairly sure my immediate response would be to print out 50 copies of [1] and send it via recorded delivery. It would be interesting to hear from anyone who has already been contacted by BT about this, and how they plan to deal with it. I've had enough of the internet for one day ;/ Cal [1] http://25.media.tumblr.com/d0efdb79f57815eff2105f65da4f4df7/tumblr_mh0qugfw…

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Not Reachable
by Blake Priddy 30 Apr '13

30 Apr '13

I am still having the random no b-leg audio until 7 sec after call is answered issue and I have seen in wireshark that when someone says they have a delay the capture shows a destination unreachable about 2-4 times on wireshark. I have changed all my QoS settings for my vlans and I am still having an issue. Attached is a screenshot of the log -- *Blakelund Priddy* Network & Systems Engineer Bryant Public School District Bryant, Arkansas 72022 http://www.bryantschools.org p 501-653-5038 f 501-847-5656

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