January 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Eavesdrop in LUA
by rod 16 Jan '10

16 Jan '10

Hi all, I'm trying to do this in LUA: A call B and I'd like to setup a new call to C with eavesdrop of A conversation with B. I have no idea how to do this if someone can help. I switched to LUA cause I see no way to achieve this with dialplan (snippets are welcome). regards, rod

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FIFO Originate caller ID
by Steve Steffler 16 Jan '10

16 Jan '10

Hello, I found an archived conversation on this list regarding FIFO origination caller ID, and how to modify it. http://old.nabble.com/FIFO-Orgination_caller_id-td26487628.html There seems to be no easy way to customize the caller ID on originated calls from the FIFO to on-hook agents who were registered dynamically. Anthony states a method to do it using static entries in the fifo conf file, and the general rationale is that SCREEN POPS be the preferred method, with the added nudge that good SIP phones can change the caller ID when the bridge is complete as well, which is all well and good. The problem is for my application, all on-hook agents are using cellular phones, and they register dynamically. Also, none of my agents are in front of a computer, so a SIP display update on the phone or screen pop on the computer in front of them is not really an option, and the only way they can identify calls from my FIFO right now is because they are the ones with NO CALLER ID (in other words, their mobile phones do not display the name, and the number is not recognized because it is set by FreeSWITCH to be "fifo+fifoname" instead of being numeric. This is far from ideal. I am wondering if there is anyone on the list who knows how to configure the origination_caller_id_number/name variables for dynamically registered on-hook agents so that the caller ID from the FIFO customer's incoming call is displayed to them instead of the above mangled caller ID? I'm not disagreeing that it is an old-skewl way of thought, but in actuality it is just a way to interface with old-school telephony devices (i.e. non-Smartphone mobile phones) and I am not sure how to accomplish this. Any help/input would be appreciated. Regards, Steve

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Re: [Freeswitch-users] multiple leg and multiple rtp
by Anthony Minessale 16 Jan '10

16 Jan '10

Perhaps best not to help him anymore without an apology for the snap judgement and comparison to asterisk clearly designed to push our buttons. We'll be here when you realize we were trying to help you but I can't promise we will still have paitence...... On Jan 13, 2010 10:54 PM, "Michael Jerris" <mike(a)jerris.com> wrote: You already are running on trunk. Mike On Jan 13, 2010, at 11:26 PM, Sergey Okhapkin wrote: > I run FreeSWITCH Version 1.0.5pre10 (16012M... _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists...

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Domains.
by FERNANDO VILLARROEL 16 Jan '10

16 Jan '10

Dear. I installed FS FreeSWITCH Version 1.0.trunk (16144) I have a problem when i send traffic from a external gateway, the calls are rejected: 2010-01-05 16:52:37.254194 [DEBUG] sofia.c:5359 0 acls to check for proxy 2010-01-05 16:52:37.254194 [DEBUG] sofia.c:5377 network ip is a proxy [0] 2010-01-05 16:52:37.254194 [DEBUG] sofia.c:5405 IP XXX.XXX.XX.125 Rejected by acl "domains". Falling back to Digest auth. Anyone could me explain like i can do. Regards. Fernando

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Issue Compiling ESL (PHP)
by Alfredo Quiroga-Villamil 16 Jan '10

16 Jan '10

All: System: Linux usnc-rtp-01-fs 2.6.23.1-42.fc8 #1 SMP Tue Oct 30 13:55:12 EDT 2007 i686 i686 i386 GNU/Linux I am trying to compile ESL, following the following steps: 1- cd to my libs/esl directory as the wiki page indicates. 2- run make I then get right away the following: cc -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes -c src/esl.c -o src/esl.o cc -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes -c src/esl_event.c -o src/esl_event.o cc -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes -c src/esl_threadmutex.c -o src/esl_threadmutex.o cc -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes -c src/esl_config.c -o src/esl_config.o g++ -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable -c src/esl_oop.cpp -o src/esl_oop.o ar rcs libesl.a src/esl.o src/esl_event.o src/esl_threadmutex.o src/esl_config.o src/esl_oop.o ranlib libesl.a cc -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes fs_cli.c -o fs_cli -L. -L../../libs/libedit/src/.libs -lncurses -lpthread -lesl -ledit cc -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes testclient.c -o testclient -L. -lncurses -lpthread -lesl cc -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes testserver.c -o testserver -L. -lncurses -lpthread -lesl cc -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes ivrd.c -o ivrd -L. -lncurses -lpthread -lesl 3- I try typing then: make phpmod and get the following: make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x" CFLAGS="-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes" CXXFLAGS="-I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable" CXX_CFLAGS="" -C php make[1]: Entering directory `/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/php' g++ -I/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable -I/usr/include/php -I/usr/include/php/main -I/usr/include/php/TSRM -I/usr/include/php/Zend -I/usr/include/php/ext -I/usr/include/php/ext/date/lib -Wno-unused-label -Wno-unused-function -c esl_wrap.cpp -o esl_wrap.o g++ -shared -Xlinker -x esl_wrap.o ../libesl.a -L/usr/kerberos/lib -lcrypt -lcrypt -lncurses -laspell -lpspell -lcurl -lresolv -lm -ldl -lnsl -lm -ldl -lcurl -ldl -lm -lcrypt -lm -lm -lcrypt -lpthread -o ESL.so -L. make[1]: Leaving directory `/usr/src/freeswitch-1.0.5-20100113-0400/libs/esl/php' I've installed I think all the -dev dependencies listed in the wiki. Any ideas? Thanks in advance, Alfredo Q-V

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xset warning message starting FS as daemon
by Robert Hadley 15 Jan '10

15 Jan '10

Hi, Running trunk on CentOS 5.3, I don't get any warning messages starting FS manually with the -nc option. I get this message when I start FS as a daemon. [root@roberth-c53 fstrkbld]# service freeswitch start Starting freeswitch: xset: unable to open display "" xset: unable to open display "" [ OK ] Is this due to something I am doing or is it FS? Thanks, Robert

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OT: Voip presentation
by Joseph L. Casale 15 Jan '10

15 Jan '10

I have to do a voip presentation to a group and need something elaborating on the whole process, where does a call go once it leaves it the building on the wire, how it re-enters the pstn, wtf is a pstn clec etc... Rather than make something from scratch (although prolly not hard as the level of detail needed is low) I thought I would ask if anyone knew of something in the public domain I could snag and start with? Hopefully the next project is dropping 1/2 or a dozen ports into a Nortel Meridian dinosaur with fs behind it... Thanks! jlc

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Conference Questions
by Alfredo Quiroga-Villamil 15 Jan '10

15 Jan '10

Hello: I've been using asterisk for a little bit over three years now. A couple of months ago I found out about freeswitch, took a look at it, thought it was interesting and moved on. A few weeks ago, I started looking at a project I've been wanting to work on for quite a while using conferences and started exploring systems and different approaches. Based on the requirements I have, I decided to use freeswitch. It seemed like it had the best support for conferencing so I went for it. According to some documentation I found it also seems to allow for more concurrent calls than asterisk which is an added bonus. I got a server ready, installed FC8 on it which is what I have in production now, unpacked freeswitch there and so far it's running beautifully. Very painless process really to get it installed, I was happy to see that. Configuration seems a bit different since it's XML; but being a developer myself I can see many advantages to having done that in the future as the system scales and grows in complexity. Sorry for the long introduction, getting to my question now. So ... What I want to be able to do is the following: Create and control conferences via the HTTP API. I've been reading a bit for the past two days the documentation and I am becoming more familiar now with how things are done using ESL, the support for PHP, perl and I believe others. a) It seemed to me like the way to setup the moderator of the conference is by setting a parameter in the DialPlan and specifying based on a condition who the moderator is, say for instance the destination number. That's fine and it makes sense, however, say that I am creating a new conference and I want to have 3 participants where one of them is the moderator. What would I have to do to specify that person A dialing for example number xxx-xxx-xxxx is the moderator (via HTTP)? Would I have to create my own call to the system and add say an entry to DialPlan with the right parameter for the moderator, then create the conference? b) When a conference is created, or when I go to create a new conference via HTTP using the API, does it allow for example for all numbers that will be added to be dialed at once? Or should the process be dial each participant, sending say 3 http requests via the API? The API command "conference dial" seems to only take one argument for destination number; but I am asking just in case I missed something. Thanks in advance for the help and I apologize for the long email. Alfredo

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proxy_media seems to be broken
by Klaus Hochlehnert 15 Jan '10

15 Jan '10

Hi, I just tested with the latest tarball (11. Jan). And now my T.38 Fax config with proxy_media doesn't work anymore. Here's the config: Fax <-> Cisco SPA2102 <-> FS <-> Lancom SIP/ISDN-Gateway <-> ISDN proxy_media and late negotiation is turned on. Here's what happens: - Fax makes call - Lancom routes the call to ISDN - Remote fax takes the call - Lancom sends reinvite with T.38 - The Cisco just keeps on ringing without noticing that the call was answered This config worked in older FS versions without problems. Might this already be fixed in rev 16250 (bypass_media problem)? Or did the config of proxy_media change? Thanks, Klaus

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Changing the username portion of the RURI with registered devices
by Kristian Kielhofner 15 Jan '10

15 Jan '10

Hello everyone, Is there (or can we get) a way to set the username portion of the request URI (along with To:) arbitrarily? Obviously this can be done when specifying a bridge string but I'm wondering if it would be possible with registered contacts. So... Let's say I have another FS/Asterisk/etc system registered with the username "gw". Let's say I want to direct multiple DIDs to that registered endpoint. I'd bridge to: sofia/internal/gw%domain.local (or whatever my domain was) The INVITE will go out with a destination URI of the registered contact. Probably something like: gw(a)192.168.1.10 (assuming that's where I was registered) I'd like to be able to do something like: {user_uri=9415551212}sofia/internal/gw%local.domain Before sending the INVITE to the device, FS/Sofia would replace the username portion in the RURI/To: with 9415551212, something like: 9415551212(a)192.168.1.10 Is this currently possible? Basically I don't want to have to independently register each DID in this scenario. -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com

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