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web app talking to Freeswitch API
by Freddy Andersen 12 Jun '14

12 Jun '14

We are implementing mod_callcenter in our freeswitch and wanted to develop a small web app that could list members in the queue and also agents and agents status in each queue. These are all available through Freeswitch api. I’m wondering what the best module for getting this information and also login-in/out agents would be? I started looking at twisted but it feels like all I need is a simple rest web api? Is this what mod_curl is? -- - Freddy

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configure fails on 1.4 with enable-core-pgsql-support and custom PG location
by Victor Chukalovskiy 11 Jun '14

11 Jun '14

Hi, Getting the following issues with libpq when trying to ./configure --enable-core-pgsql-support: .... checking for pg_config... /opt/PostgreSQL/9.2/bin/pg_config checking for pkg-config... (cached) /usr/bin/pkg-config checking for PostgreSQL libraries via pg_config... -L/opt/PostgreSQL/9.2/lib checking for PQgetvalue in -lpq... no configure: error: no usable libpq; please install PostgreSQL devel package or equivalent This does not help: https://wiki.freeswitch.org/wiki/PostgreSQL_in_the_core#Configure What helps is to put "/opt/PostgreSQL/9.2/lib" into /etc/ld.so.conf.d/pg.conf BUT this creates issues with other libraries for programs other than freeswitch. Any alternative solution? Background: CentOS 6.4; PostgreSQL 9.2 from Enterprise DB; FS 1.4.4 Thanks for your help! -Victor

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effective_caller_id_name doesn't update Caller-Caller-ID-Name
by Geoff Nordli 11 Jun '14

11 Jun '14

Hi. My goal is to set the Caller-Caller-ID-Name for inbound calls going into a conference. I set the effective_caller_id_name before the call gets sent into the conference. Here is a log 2014-06-10 23:12:38.262554 [DEBUG] switch_ivr.c:612 sofia/external/anonymous(a)anonymous.invalid Command Execute export(effective_caller_id_name=test) EXECUTE sofia/external/anonymous(a)anonymous.invalid export(effective_caller_id_name=test) 2014-06-10 23:12:38.262554 [DEBUG] switch_channel.c:1145 EXPORT (export_vars) [effective_caller_id_name]=[test] 2014-06-10 23:12:38.262554 [DEBUG] switch_ivr.c:612 sofia/external/anonymous(a)anonymous.invalid Command Execute set(effective_caller_id_name=test) EXECUTE sofia/external/anonymous(a)anonymous.invalid set(effective_caller_id_name=test) 2014-06-10 23:12:38.262554 [DEBUG] mod_dptools.c:1373 sofia/external/anonymous(a)anonymous.invalid SET [effective_caller_id_name]=[test] 2014-06-10 23:12:38.262554 [DEBUG] switch_ivr.c:612 sofia/external/anonymous(a)anonymous.invalid Command Execute conference(12001@wideband) EXECUTE sofia/external/anonymous(a)anonymous.invalid conference(12001@wideband) I see from the channel variables it is being set: variable_export_vars: effective_caller_id_name variable_effective_caller_id_name: test But the Caller-Caller-ID-Name variable doesn't get updated after the call gets bridged to the conference room. Is there a different way I should be doing this? thanks, Geoff

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ESL problem using WSGI with Django
by Freddy Andersen 11 Jun '14

11 Jun '14

I created a Django app that uses ESL to manage Freeswitch which works fine when I’m running the app using manage.py BUT when I move this app to our production environment using wsgi.py my connection hangs trying to connect to Freeswitch. I narrowed down that its hung here: class ESLconnection: __swig_setmethods__ = {} __setattr__ = lambda self, name, value: _swig_setattr(self, ESLconnection, name, value) __swig_getmethods__ = {} __getattr__ = lambda self, name: _swig_getattr(self, ESLconnection, name) __repr__ = _swig_repr def __init__(self, *args):- print('before this') this = apply(_ESL.new_ESLconnection, args) print('after this') try: self.this.append(this) except: self.this = this __swig_destroy__ = _ESL.delete_ESLconnection __del__ = lambda self : None; I get 'before this' in my log but thats it… I found a few places where people needed to increase threads for Apaches WSGI process but that does not help… This is what I have: WSGIDaemonProcess debug processes=4 threads=16 stack-size=1048576 maximum-requests=10 inactivity-timeout=30 The I get this in my debug logs: [Wed Jun 11 10:44:26 2014] [error] before connection [Wed Jun 11 10:44:26 2014] [error] before this [Wed Jun 11 10:44:51 2014] [info] mod_wsgi (pid=14624): Daemon process inactivity timer expired, stopping process 'debug'. [Wed Jun 11 10:44:51 2014] [info] mod_wsgi (pid=14624): Shutdown requested 'debug'. anything would help at this point… -- Freddy

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Location where FreeSWITCH mod_callcenter read audio files
by Joel White 11 Jun '14

11 Jun '14

I have made some recordings for the callcenter. I have placed them in several locations (sounds, recordings) yet I cannot get the 'announce-sound' option in mod_callcenter to play these files. Can someone help me in locating the directory that the files need to be placed for the system to be able to access them? Thank you in advance

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ClueCon Weekly! June 11 - FreeSWITCH's fs_cli tricks!
by Ken Rice 11 Jun '14

11 Jun '14

Hey Guys Join us today for fs_cli tricks! You know it and you use it all the time! But when¹s the last time you looked at some of the tricks you can do with fs_cli! Join us today at 1PM EST on sip:888@conference.freeswitch.org to see what fun tricks you can do and help save time! Ken -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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ClueCon Speakers Announced!
by Ken Rice 11 Jun '14

11 Jun '14

Hey ClueCon¹ers! If you havent visited ClueCon.com lately Now might be a Good time to do so! We have added a ton of confirmed speakers and you can see them at https://cluecom.com/speakers !! Come for ClueCon Stay for the Training! Space is limited in the training so register today! The course will include these topics: · Quick-start on FreeSWITCH Basics · Setting up and using WebRTC · Security (SRTP and ZRTP) and Certificates · Using FreeSWITCH's "other" configs - FreeSWITCH as a proxy or hybrid · Multi-tenancy with FreeSWITCH · XML/Curl and controlling FreeSWITCH real time · Raw Event Sockets Who should attend: · VoIP providers seeking ³Open Source² platforms to enhance their businesses · Anyone seeking proficiency in FreeSWITCH · Network Consultants and VARs who need a jump start in the technology · Developers who want to use FreeSWITCH to create new telephony applications and appliances · Savvy Tech People who like to learn and build cool stuff! Schedule Details: · The course will begin at 9am a the (PLEASE NAME THE LOCATION) · The course will end at around 4pm with a one-hour slot to play and answer questions where the teacher will remain present (as long as you have questions!) -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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mod_skypopen and mod_sms
by Russell Treleaven 10 Jun '14

10 Jun '14

It is possible but you will need to tweak headers in the chatplan. first step though is <param name="report_incoming_chatmessages" value="true"/> Sorry I can't give more specifics at the moment. On Tue, Jun 10, 2014 at 7:08 AM, Mark Campbell-Smith < mcampbellsmith(a)gmail.com> wrote: > Hi! > > I read in Jira that it should be possible to use mod_sms with chat > messages received from mod_skypopen. > > How do I do this? > > When I send a message to FS, I see the debug messages received from > mod_skypopen, but do not see mod_sms processing the message. > > 2014-06-10 21:02:48.956151 [DEBUG] skypopen_protocol.c:207 > [1.5.13b+git~20140603T193316Z~70b03acf71~64bit ] [DEBUG_SKYPE 207 > ][skype101 ][IDLE,IDLE] READING: |||CHAT > #skype1/$skyp2;f067e42dadeb74b ACTIVITY_TIMESTAMP 1402398167||| > 2014-06-10 21:02:48.956151 [DEBUG] skypopen_protocol.c:207 > [1.5.13b+git~20140603T193316Z~70b03acf71~64bit ] [DEBUG_SKYPE 207 > ][skype101 ][IDLE,IDLE] READING: |||CHATMESSAGE 1453 STATUS > RECEIVED||| > > I can get normal SIP messages to work. > > Any ideas? > > Thanks > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >

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STUN Binding request times out after 200 OK received on client
by Jayesh Nambiar 10 Jun '14

10 Jun '14

Hi, I am developing an iOS calling App with libnice for ICE capabilities. The calls go through freeswitch and since freeswitch supports ICE, I expected it to work out of the box. But I have a problem. The way libnice works is as follows: 1) The client sends INVITE with his candidates. 2) Freeswitch parses the candidates. 3) Immediately after 200 OK is received, I can see FreeSWITCH choosing an ICE candidate and logging "Activating Audio ICE" indicating the IP and Port and it sends media/RTP to. 4) But the libnice agent tries to look at the candidates received in 200 OK and tries to send a STUN binding request to FreeSWITCH IP as that IP is mentioned as ICE candidate in the 200 OK coming from freeswitch. 5) The FreeSWITCH doesn't respond to that STUN binding request and libnice agent times out. 6) Ideally, if libnice gets back a response from the freeswitch, it should initiate a Re-INVITE with the candidate pair selected. Now here, the freeswitch starts sending RTP to client's candidate, but the client gets stuck because the candidate pair is not formed as the libnice agent keeps waiting for a response for the STUN Binding Request. Is there anything that I am missing here. Any help is highly appreciated. Thanks, --- Jayesh

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httapi and getting bridge hangup cause
by Colin Martin 10 Jun '14

10 Jun '14

Hi, I’m new to Freeswitch, and gradually getting to grips with the httapi application. I’m having trouble getting the hangup cause back after a bridge is complete. I’ve tried Googling around the issue, but I can’t find anything that would explain why this doesn’t work. So, the web server initiates the bridge with the following: <?xml version="1.0" encoding="UTF-8" standalone="no"?><document type="text/freeswitch-httapi"><work><execute application="set">call_timeout=30</execute><execute application="bridge">sofia/gateway/pbx/200(a)destination.com</execute></work></document> The call is placed and the bridge works as expected. The called party then hangs up. As expected, the httapi application makes a further request, but only sends a small subset of the query parameters that it would usually send, so I can’t identify the hangup cause. I expect one of my assumptions is wrong and this is expected behaviour. What am I doing wrong, and how should I be getting the hangup cause? Thanks for any help you can provide! Colin Martin

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