May 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

XML encoding attribute ?
by Torstein Knutsen 03 May '13

03 May '13

Hi I have a questione regarding XML output in event socket or cli. Example : "+" in the fs_cli when issuing "conference xml_list" returns <caller_id_name>*%2B*4726262626</caller_id_name> When in fact my caller_id_name is "+4726262626". The printout shows no encoding attribute : <?xml version="1.0"?> I'm I correct to assume that if I can get encoding in utf-8, the "+" will be "+" and not "*%2B"* in the listing ? Does anybody know how I can accomplish this ? Im running ubuntu server 10.04 / FreeSWITCH Version 1.2.8+git~20130412T165647Z~3fde0e453e (git 3fde0e4 2013-04-12 16:56:47Z) best regards Torstein

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T38 Re-INVITE: FS responds 100 Trying, nothing further happens
by Örn Arnarson 03 May '13

03 May '13

Hello, I'm trying to get T38 gatewaying to work. I have a call coming in with FAX audio, and an endpoint that needs T38. The endpoint will send a T38 re-invite, to which FS responds 100 trying, and after 40 seconds the endpoint will give up (citing stale 100 trying). I set these variables before briding: <action application="set" data="fax_enable_t38=true"/> <action application="set" data="fax_enable_t38_request=true"/> And then I bridge like this (dialstring is sofia/profile/number@ip:port): <action application="bridge" data="{execute_on_image=t38_gateway self nocng}${dialstring}"/> I've tried all variants I can think of, and always have the same result, except for when fax_enable_t38 = false, in which case FS rejects the re-invite. I found some old discussions on the mailing list alluding to the same problem, but not much in the way of solutions. Can anybody give me a pointer as to what I'm doing wrong? I was running a git clone from 1.2.2, but I upgraded to latest stable (1.2.7) with the same results. Here's what happens in FS. At the very end you can see the Re-INVITE, and then nothing happens post that point. http://pastebin.freeswitch.org/20853 Best regards, Örn

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Copy rtmp extra header to sofia
by Iwan Budi Kusnanto 02 May '13

02 May '13

Hi, I want to copy RTMP extra header to Sofia/SIP. I can do it in dialplan like this <action application="export" data="nolocal:sip_h_X-Flash-Header1=${rtmp_u__X-Flash-Header1}"/> This way, i need to specify each header name in dialplan. Can we make dialplan that support arbitrary header name? I can do it in Lua with this method: - execute uuid_dump - search line that started by variable_rtmp_u__X-Flash - extract header_name and header_value - session:execute("export", "nolocal:sip_h_" .. header_name .. "=" .. header_value) Any better way to do this? Thanks -- Iwan Budi Kusnanto

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make ekiga to flex call
by Navnath Sonavne 02 May '13

02 May '13

Hi All, I am using flex client(at 192.168.9.165) given in freeswitch source to make call to Ekiga(h323 protocol) softphone(at 192.168.8.27) via freeswitch server(at 192.168.8.41). I have two users registered on freeswitch with extension as 1100 and 1101 in default context. I loged in using one of extension in flex client,then I dail another extension. After I dail,call goes to Ekiga softphone at 192.168.8.27 because I have made changes in default.xml accordingly to forward calls to this ip. Here is My default.xml part : <extension name="Local_Extension"> <condition field="destination_number" expression="^(\d{4})$"> <action application="export" data="dialed_extension=$1"/> <action application="set" data="effective_caller_id_number=${dialed_extension}"/> <action application="set" data="PEER_IP=192.168.8.27"/> <action application="set" data="call_timeout=30"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="proxy_media=true"/> <action application="bridge" data="h323/$1@${PEER_IP}"/> </condition> </extension> Ekiga phone can answer the call and there is audio transmission from both end successfully. But this is one way calling i.e. from flex to ekiga. Now I want to call from Ekiga softphone to flex client. How to call flex client from ekiga softphone? Anybody please me on this issue. -Navnath.

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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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