October 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

test
by Peter Hartmann 09 Oct '13

09 Oct '13

Heloooooo! Peter Hartmann Hartmann Computer Consulting http://blog.hartmanncomputer.com

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[freeswitch-codec]
by Dzexolokpli AMOUZOU 09 Oct '13

09 Oct '13

Hello, I have installed freeswitch and kamailio on debian 7.0 wheezy referencing to http://nil.uniza.sk/sip/kamailio-33-and-freeswitch-122-interconnection-voic…. I notice that the quality of the voice using kamailio is better than it using freeswitch. searching the problem source, I find that codec G722 is not include in freeswich. How can I include this codec in freeswwitch modules. thanks you in advance.

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Prevent A leg from hangup after bridge with inbound ESL socket
by Alex Massover 09 Oct '13

09 Oct '13

Hi, I have a very simple dialplan that just do park for incoming calls. All rest of leg management is done via ESL inbound socket. I'm trying to do the same behavior like in this dialplan example, but from ESL inbound socket: <action application="set" data="hangup_after_bridge=false"/> <action application="bridge" data="sofia/internal/1001@$${domain}"/> <action application="bridge" data="sofia/internal/1002@$${domain}"/> The problem is with bridge API, if B leg doesn't answer (e.g. 404, or busy), A leg disconnects. But I'm trying to prevent A leg from disconnecting in order to do bridge to other place. Looks like hangup_after_bridge=false, park_after_bridge=true, transfer_after_bridge etc don't have any effect when bridge done from inbound socket. A leg disconnects always. Is there any way to keep A leg after bridge with inbound socket? I'm aware of originate, but prefer to user bridge. -- Best Regards, Alex Massover

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Re: [Freeswitch-users] Multiple contexts per sip profile
by Lloyd Aloysius 09 Oct '13

09 Oct '13

Please have a look into here http://wiki.freeswitch.org/wiki/Multi-tenant Lloyd * <http://www.sunteltech.ca/blog/>* On Tue, Oct 8, 2013 at 10:51 AM, Grant Bagdasarian <GB(a)cm.nl> wrote: > Hello,**** > > ** ** > > I’m trying to configure FreeSWITCH to have multiple contexts per sip > profile like below.**** > > ** ** > > But this does not seem to work.**** > > The reason I want this, is to have each incoming called number to have its > own context in which extensions can be created. **** > > The problem with using extensions and only one context is I can’t defined > more than one extension with the same name. FS seems to overwrite or ignore > ones that do.**** > > ** ** > > Is this even possible? If so, what do I need to change? Is there perhaps a > better way for doing this?**** > > ** ** > > *sip_profile/inbound.xml:* > > ** ** > > <profile name="inbound">**** > > ** ** > > <settings>**** > > ** ** > > <param name="context" value="inbound"/>**** > > <param name="dialplan" value="XML"/>**** > > ** ** > > </settings>**** > > </profile>**** > > ** ** > > *dialplan/inbound.xml:* > > ** ** > > <?xml version="1.0" encoding="utf-8"?>**** > > <include>**** > > <context name="inbound">**** > > <X-PRE-PROCESS cmd="include" data="inbound/*.xml"/>**** > > ** ** > > <extension name="route_to_context">**** > > <condition field="destination_number" expression="^(\d+)$">**** > > <action application="transfer" data="start XML $1"/>**** > > </condition>**** > > </extension>**** > > **** > > </context>**** > > </include>**** > > ** ** > > *dialplan/inbound/1000.xml:* > > ** ** > > <include>**** > > ** ** > > <context name="1000">**** > > ** ** > > <extension name="start">**** > > <condition field="destination_number" expression="^(\d+)">**** > > <action application="answer"/>**** > > <action application="transfer" data="playback"/>**** > > </condition>**** > > </extension>**** > > ** ** > > ** ** > > <extension name="playback">**** > > <condition>**** > > <action application="playback" > data="/usr/src/freeswitch/sounds/fs-hello-world-for-1000.wav"/>**** > > </condition>**** > > </extension>**** > > ** ** > > </context>**** > > </include>**** > > ** ** > > *dialplan/inbound/1001.xml:* > > ** ** > > <include>**** > > ** ** > > <context name="1001">**** > > ** ** > > <extension name="start">**** > > <condition field="destination_number" expression="^(\d+)">**** > > <action application="answer"/>**** > > <action application="transfer" data="playback"/>**** > > </condition>**** > > </extension>**** > > ** ** > > ** ** > > <extension name="playback">**** > > <condition>**** > > <action application="playback" > data="/usr/src/freeswitch/sounds/fs-hello-world-for-1001.wav"/>**** > > </condition>**** > > </extension>**** > > ** ** > > </context>**** > > </include>**** > > ** ** > > ** ** > > ** ** > > ** ** > > _________________________________________________________________________ > 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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esl dialer
by Marcin S 09 Oct '13

09 Oct '13

Hello, I'm trying to switch from dialogic api to freeswitch/esl. Dealing with inbound calls is easy (using esl outbound socket), however i have some problems with outbound calls. My application is written in C. I create a handle to esl (esl_connect), then issue "bgapi originate ... &socket(127.0.0.1:8084 async full)" in order to handle this connection just like inbound call. So far so good. Here come the problems: 1. esl creates new thread for my call - thats ok, it was expected - but I do not receive any events on this new handle... 2. esl_events fails - here is some strace output: [pid 31588] send(5, "filter unique-id a88f1268-2c28-11e3-befa-311d1641c437\n\n", 55, 0) = 55 [pid 31588] recv(5, "Content-Type: command/reply\nReply-Text: +OK filter added. [unique-id]=[a88f1268-2c28-11e3-befa-311d1641c437]\n\n", 65535, 0) = 110 [pid 31588] send(5, "event plain ALL\n\n", 17, 0) = 17 [pid 31588] recv(5, "Content-Type: command/reply\nReply-Text: -ERR command not found\n\n", 65535, 0) = 64 [pid 31588] send(5, "linger", 6, 0) = 6 [pid 31588] send(5, "\n\n", 2, 0) = 2 [pid 31588] recv(5, "Content-Type: command/reply\nReply-Text: -ERR command not found\n\n", 65535, 0) = 64 3. how can i distinguish, whether this call is successfull or not? What am I doing wrong?

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Re: [Freeswitch-users] esl dialer
by Jeff Leung 09 Oct '13

09 Oct '13

Sorry for that empty reply, didn't lock the phone before i placed it in my pocket. Jeff Leung <jleung(a)v10networks.ca> wrote: Guillermo Ruiz Camauer <grcamauer(a)gmail.com> wrote: I handle all ESL communications from just one thread which receives and processes or generates all events and commands. I do have a timer for each call which I use for certain timeouts (user inputs and ring time). Guillermo On Tue, Oct 8, 2013 at 4:20 AM, Marcin S <ms4esl(a)gmail.com> wrote: > Ok, I will try this approach. Am I right, that in order to handle oubound > calls similiar to inbound, I will have to spawn new thread for each call? > > > 2013/10/5 Guillermo Ruiz Camauer <grcamauer(a)gmail.com> > >> I have a Dialer that uses ESL from C, but I make an inbound connection to >> Freeswitch. It is a socket which I keep open permanently. >> I recommend reading chapter 4 of the FreeSwitch Cookbook. >> >> Guillermo >> >> >> On Fri, Oct 4, 2013 at 5:09 AM, Marcin S <ms4esl(a)gmail.com> wrote: >> >>> Hello, >>> >>> I'm trying to switch from dialogic api to freeswitch/esl. Dealing with >>> inbound calls is easy (using esl outbound socket), however i have some >>> problems with outbound calls. >>> My application is written in C. I create a handle to esl (esl_connect), >>> then issue "bgapi originate ... &socket(127.0.0.1:8084 async full)" in >>> order to handle this connection just like inbound call. So far so good. >>> Here come the problems: >>> >>> 1. esl creates new thread for my call - thats ok, it was expected - but >>> I do not receive any events on this new handle... >>> >>> 2. esl_events fails - here is some strace output: >>> >>> [pid 31588] send(5, "filter unique-id >>> a88f1268-2c28-11e3-befa-311d1641c437\n\n", 55, 0) = 55 >>> [pid 31588] recv(5, "Content-Type: command/reply\nReply-Text: +OK filter >>> added. [unique-id]=[a88f1268-2c28-11e3-befa-311d1641c437]\n\n", 65535, 0) = >>> 110 >>> [pid 31588] send(5, "event plain ALL\n\n", 17, 0) = 17 >>> [pid 31588] recv(5, "Content-Type: command/reply\nReply-Text: -ERR >>> command not found\n\n", 65535, 0) = 64 >>> [pid 31588] send(5, "linger", 6, 0) = 6 >>> [pid 31588] send(5, "\n\n", 2, 0) = 2 >>> [pid 31588] recv(5, "Content-Type: command/reply\nReply-Text: -ERR >>> command not found\n\n", 65535, 0) = 64 >>> >>> 3. how can i distinguish, whether this call is successfull or not? >>> >>> What am I doing wrong? >>> >>> _________________________________________________________________________ >>> 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 >>> >>> >> >> >> -- >> Guillermo Ruiz Camauer >> >> _________________________________________________________________________ >> 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 >> >> > > _________________________________________________________________________ > 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 > > -- Guillermo Ruiz Camauer _________________________________________________________________________ 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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Re: [Freeswitch-users] esl dialer
by Jeff Leung 09 Oct '13

09 Oct '13

Guillermo Ruiz Camauer <grcamauer(a)gmail.com> wrote: I handle all ESL communications from just one thread which receives and processes or generates all events and commands. I do have a timer for each call which I use for certain timeouts (user inputs and ring time). Guillermo On Tue, Oct 8, 2013 at 4:20 AM, Marcin S <ms4esl(a)gmail.com> wrote: > Ok, I will try this approach. Am I right, that in order to handle oubound > calls similiar to inbound, I will have to spawn new thread for each call? > > > 2013/10/5 Guillermo Ruiz Camauer <grcamauer(a)gmail.com> > >> I have a Dialer that uses ESL from C, but I make an inbound connection to >> Freeswitch. It is a socket which I keep open permanently. >> I recommend reading chapter 4 of the FreeSwitch Cookbook. >> >> Guillermo >> >> >> On Fri, Oct 4, 2013 at 5:09 AM, Marcin S <ms4esl(a)gmail.com> wrote: >> >>> Hello, >>> >>> I'm trying to switch from dialogic api to freeswitch/esl. Dealing with >>> inbound calls is easy (using esl outbound socket), however i have some >>> problems with outbound calls. >>> My application is written in C. I create a handle to esl (esl_connect), >>> then issue "bgapi originate ... &socket(127.0.0.1:8084 async full)" in >>> order to handle this connection just like inbound call. So far so good. >>> Here come the problems: >>> >>> 1. esl creates new thread for my call - thats ok, it was expected - but >>> I do not receive any events on this new handle... >>> >>> 2. esl_events fails - here is some strace output: >>> >>> [pid 31588] send(5, "filter unique-id >>> a88f1268-2c28-11e3-befa-311d1641c437\n\n", 55, 0) = 55 >>> [pid 31588] recv(5, "Content-Type: command/reply\nReply-Text: +OK filter >>> added. [unique-id]=[a88f1268-2c28-11e3-befa-311d1641c437]\n\n", 65535, 0) = >>> 110 >>> [pid 31588] send(5, "event plain ALL\n\n", 17, 0) = 17 >>> [pid 31588] recv(5, "Content-Type: command/reply\nReply-Text: -ERR >>> command not found\n\n", 65535, 0) = 64 >>> [pid 31588] send(5, "linger", 6, 0) = 6 >>> [pid 31588] send(5, "\n\n", 2, 0) = 2 >>> [pid 31588] recv(5, "Content-Type: command/reply\nReply-Text: -ERR >>> command not found\n\n", 65535, 0) = 64 >>> >>> 3. how can i distinguish, whether this call is successfull or not? >>> >>> What am I doing wrong? >>> >>> _________________________________________________________________________ >>> 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 >>> >>> >> >> >> -- >> Guillermo Ruiz Camauer >> >> _________________________________________________________________________ >> 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 >> >> > > _________________________________________________________________________ > 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 > > -- Guillermo Ruiz Camauer

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Checking call is connected while bridged in lua
by David Villasmil 09 Oct '13

09 Oct '13

Hello Guys, I've had this "problem" since forever :) (I've been a freeswitch user for a long time) I'm using lua and I want to bridge a call and to check immediately whether the call is connected or not WHILE connected. lua execution continues only after "bridge" has returned something.. connected, busy, whatever... But i would like check the status of the call immediately after executing the bridge... Is this possible WITHOUT esl? Thanks! David

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esl socket inbound + playback
by Marcin S 08 Oct '13

08 Oct '13

Hello, I wrote simple C application, wich opens connection to esl - freeswitch and makes call (originate ... &park). So far so good. I get ESL_EVENT_CHANNEL_ORIGINATE, ESL_EVENT_CHANNEL_ANSWER and ESL_EVENT_CHANNEL_PARK. Then I wan't to play wav file - but playback command returns "-ERR invalid session id []". What is wrong? Specifying "originate ... &playback" is out of question.

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FreeSwitch + WebRTC + JsSIP + Chrome no audio
by Rafael Santana 08 Oct '13

08 Oct '13

Hi, I'm new to telephony and FreeSwitch's world, so I apologize in advance for any nonsense I speak here. I've been trying to setup an environment where It can be possible to make a call through Google Chrome Browser using JsSIP to a standard phone device on PSTN. In my network my "PSTN gateway" is an Asterisk 1.4 instance (No, I can't chance it today). To communicate with Chrome I have a FreeSwitch 1.5.5 instance and to get access to PSTN via this instance I had to register my Asterisk instance as a gateway on my Sofia's external profile. This part of my scenario works fine. I'm able to make calls using a softphone registered on FreeSwitch to standard phones on PSTN with no problems. What I wasn't able to do until now was the JsSIP + FreeSwitch integration. To setup FreeSwitch to comunicate with JsSIP, the only thing I did was uncomment the line below on sip_profiles/internal.xml. <param name="ws-binding" value=":5066"/> I really don't know if just this is sufficient. Am I missing something important? To connect on my FreeSwitch instance from Chrome, I'm using the Tryit JsSIP demo. Today, I'm able to register on FS from Tryit demo and perform a call to a PSTN phone. The connection is established but I don't get any audio in both endpoints. The same happens when I try to call the 5000 ivr extension or an user on a softphone at the same network from my Chrome browser. Assuming that all the services I've mentioned here are running on the same network, do you have any idea why I can't get audio in both endpoints of my experiment? Additional information: Ubuntu 12.04 64 bits FreeSwitch version 1.5.5 default install configuration Tryit JsSIP Demo with jssip-0.3.0.js Thanks in advance, Rafael.

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