January 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Unable to make outbound calls
by Ian McMaster 15 Jan '14

15 Jan '14

Apologies for a simple newbie question. Following examples in the first two FreeSWITCH books, have been able to set up a SIP gateway, register, and receive calls. However I can't make calls. The dialplan is being entered, but 'hangs' at switch_channel.c:1055 New Channel sofia/external/16475551212 a) The provider name is "babytel". b) The dial plan bridge is to data="sofia/gateway/babytel/$1" (as per the books) c) The provider xml file is in conf/sip_profiles/external (as per the books) d) sofia status Name: external::babytel Type: gateway State: REGED Why is the bridge channel called "sofia/external/...."? And not "sofia/babytel/..."? Why does sofia status show "external::<provider>" (the books just show the provider name)? Can anyone explain why the bridge to the gateway does not occur? Thank you, Ian.

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Multiple extensions sharing presence
by Stephen Barry 15 Jan '14

15 Jan '14

Hello I am hoping for some advice on whether something is possible: I have created 3 directory extensions: 3862 (Desk Phone), 38621 (PC Softclient), 38622 (Mobile SoftClient), these all need to ring (at different times) when someone dials 3862, for this I have created a ring group and that is working well. Now I would like them all to share presence as all 3 extensions are effectively the same person. I have this working for inbound calls to 3862 answered on either 38621 or 38622 as I have overridden the presence_id in the dialstring for those extensions. Is there a way somehow to get a call made on extension 38621 to update 3862's presence to 'on the phone' as 3862 is being monitored by several phones using BLFs. I have had a good look around the docs, the only thing I can see is the presence map which may be what I am after, but I cannot find any explanation of what it does and how it works. Perhaps there is no way to do this, or a better way to do this. Any advice would most gratefully received. Thanks Steve

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News, Notes, and ClueCon Weekly! Today 1PM EST
by Ken Rice 15 Jan '14

15 Jan '14

Hey Guys Join us today for ClueCon Weekly at sip:888@conference.freeswitch.org 1PM EST. Today we¹ll have Richard ³Moose² Genthner with Symplicity presenting Skydas a FreeSwitch PHP API and Angular JS Mod_Callcenter Panel. In other news, for you guys running Debian, we¹ve added 2 new debian repo¹s for the 1.4beta branch and the Master branch from GIT. See https://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#Debian_pack ages on how to enable these and start using the packages! (we¹ll some open this up from Centos users with new yum repos as well. Watch for that in the near future!) Have a great week FreeSWITCHers! -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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JOB_UUID
by Ryan Gholam 15 Jan '14

15 Jan '14

Hey all , i have a question that kept me searching for a while how could i filter events concerning JOB-UUID if i have 100 JOBS

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FS verision 1.5.7b - sound is somewhat distorted
by Fredrik Åhlén 15 Jan '14

15 Jan '14

Hi everyone, this is my first exeperience with FreeSwitch. I just downloaded FreeSwitch and installed it on Windows 7 using the MSI-installer. I think it’s verision 1.5.7b, at least according to the output from fs_cli.exe. The installation is pure local on my own home LAN (100 mbit), i.e no VOIP gateways etc. I am playing around with the installation and the FSClient 1.2.3.5 as softphone calling the echo test, calling between extension 1000 – 1001, voice mail 4000 etc. Everything seems fine but the sound is somewhat distorted. The sound is a little/somewhat scratchy. I’ve tried to change codec in FreeSwitch and/or the FSClient but without luck. Since I am completely new to FreeSwitch I thought that I would ask the community about it. What might cause this sound distorsion? I have recorded the sound but I was not sure it I should attatch it or not. Any help is appreciated. Output from fs_cli.exe: “ freeswitch@internal> version FreeSWITCH Version 1.5.7b+git~20131111T224713Z~5456daa010~32bit (git 5456daa 201 3-11-11 22:47:13Z 32bit) “ >From vars.xml: “ <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G722,PCMU,PCMA,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/> “ With Best Regards, Fred

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Re: [Freeswitch-users] CHANNEL_HANGUP_COMPLETE cannot received
by Jeff Leung 15 Jan '14

15 Jan '14

I'd suggest that you cross post this issue on FreeSWITCH-CN too as they can better help you. -------- Original message -------- From: 刘云龙 <lyl(a)juyide.com> Date: 01-14-2014 6:39 PM (GMT-08:00) To: freeswitch-users(a)lists.freeswitch.org Subject: [Freeswitch-users] CHANNEL_HANGUP_COMPLETE cannot received Hi! I'm using mod_erlang_event outbound, the env is FreeSWITCH Version 1.2.14 and Erlang R16B02, My erlang applicaton behaviour normal before the call was hangup, but it cannot receive every CHANNEL_HANGUP_COMPLETE, any one can tell me why?

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Free WindowsEyes Screen Reader for owners of office 2010 or 2013
by Mitch Capper 15 Jan '14

15 Jan '14

Not exactly on topic but I know we have a good number of users who use screen readers and so I figured I would mention the nice deal just announced: http://www.windoweyesforoffice.com/FAQ/ . Free current version of WindowsEyes if you own a legitimate copy of office 2010 or 2013. Also, in case anyone using screen readers is not aware, FSClient( http://wiki.freeswitch.org/wiki/FSClient ) the Windows softphone client based on FreeSWITCH is optimized for screen readers including WindowsEyes, thanks to the help of great supporters like giovanni. ~Mitch

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Problem with call on same network
by peetee23＠hushmail.com 14 Jan '14

14 Jan '14

Hi All, I have an issue between clients which are on same network (WiFi for example). What could I do to solve this issue? Im aware of "bypass media" but this will affect other regular calls behind NAT, right? Thank you for your help PT

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Maintain external caller ID on follow me forward
by Adrian Pinder 14 Jan '14

14 Jan '14

Her's what happening: - I have a follow me rule after 3 rings to my internal extension then forward it to my cell phone - someone calls from the pstn side (external caller) the calls comes in with the correct caller ID to my internal extension - When the calls gets forwarded to my cell I get the caller ID setup on freeswitch for all outbound calls basically I get the (Effective Caller ID Number) How can I maintain the external caller id to show up on my cell instead of the did information i entered in freeswitch for all outbound calls. Thanks in advance

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Interrupt a switch_ivr_originate
by Stephen Wilde 14 Jan '14

14 Jan '14

There is a way to interrupt a switch_ivr_originate? In my code I do: switch_status_t status = switch_ivr_originate(NULL, &session, &cause, dialstring, 60, NULL, NULL, NULL, NULL, NULL, SOF_NO_LIMITS, NULL); that ends when completed (i.e. early media present) or call answered or for timeout (60 s in my case). It's possible inside another thread to force the interruption (abort) of this originate? Any suggestion? Stephen

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