August 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Recording Volume
by Balamurugan Mahendran 26 Aug '12

26 Aug '12

Guys, Please let me know how to tune call recording volume, Its very poor to listen my calls on any Iphone/ipad & android devices. Thanks, Bala

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Problem with linking freeswitch with GUI
by Ikenna Onyeagba 26 Aug '12

26 Aug '12

Hello, am having problems with downloading freeswitch GUI i have already installed freeswitch in fedora linux but am stuck from there am using freeswitch in my final project am implementing voip on it. I want to link freeswitch with GUI so I can tweak it. Regards Ikenna

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Mod Erlang Event in Freeswitch
by Eranga Udesh 25 Aug '12

25 Aug '12

Hi, I'm trying mod_erlang_event module in Freeswitch. However when a call comes to extension 111, the Erlang node receives RPC call correctly (my config as below) but it seems Freeswitch module does not receive the returned {Ref, NewPid} tuple of the Erlang function. I use Erlang R15B01 version. I also tried with Registered process configuration as in the Mod Erlang Event wiki, but there also it timeout. I wonder if it is an incompatibility with Erlang 15B? Any ideas what could be wrong? Config: <extension name="to_erlang" continue="true"> <condition field="destination_number" expression="^111$" > <action application="erlang" data="fsivr:new_call ivr@linux" /> </condition> </extension> Output in freeswitch debug: 2012-08-23 17:06:34.057997 [DEBUG] mod_erlang_event.c:1519 enter erlang_outbound_function fsivr:new_call ivr@linux 2012-08-23 17:06:34.057997 [DEBUG] mod_erlang_event.c:1525 Creating new listener for session 2012-08-23 17:06:34.098399 [DEBUG] mod_erlang_event.c:1527 Launching new listener 2012-08-23 17:06:34.098399 [DEBUG] mod_erlang_event.c:1537 Creating new spawned session for listener 2012-08-23 17:06:34.098399 [DEBUG] mod_erlang_event.c:1423 rpc call: fsivr:new_call(Ref) 2012-08-23 17:06:34.098399 [DEBUG] mod_erlang_event.c:1008 Connection Open 2012-08-23 17:06:39.177997 [WARNING] mod_erlang_event.c:1436 Timed out when waiting for outbound pid 12.0.0@freeswitch@linux 404fc3c7-45d1-4194-ad37-1e118fdee8d0 2012-08-23 17:06:39.177997 [DEBUG] switch_channel.c:2919 (sofia/internal/5000@linux4) Callstate Change RINGING -> HANGUP Thanks in advance. - Eranga

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Shared Call Appearance and Transferring Calls on Cisco SPA504G
by Colin Mason 25 Aug '12

25 Aug '12

So I have been experimenting with a key system for an office with 3 Cisco SPA504G phones and FreeSWITCH 1.2.1. They require 3 shared lines. I have inbound and outbound calls working properly with the shared lines and the phones are configured properly with shared lines, Broadcom etc. Visually everything seems to work with the line notifications on the 3 phones. My internal profile has: <param name="manage-shared-appearance" value="true"/> <param name="manage-presence" value="true"/> <param name="dbname" value="share_presence"/> <param name="presence-hosts" value="$${domain}"/> <param name="multiple-registrations" value="true"/> <param name="force-register-domain" value="$${domain}"/> <param name="force-subscription-domain" value="$${domain}"/> <param name="force-register-db-domain" value="$${domain}"/> <param name="presence-probe-on-register" value="true"/> <param name="presence-privacy" value="$${presence_privacy}"/> <param name="send-presence-on-register" value="true"/> I am having problems with transferring calls. If I put a call on hold on phone 1 and press the line 1 button on phone 1, the call resumes just fine. But if I put a call on hold on phone 1 and try to resume the call on phone 2 by pressing the blinking red line 1 button, the phone tries to establish a new call to the user associated with line 1 instead of taking (transferring) the call from phone 1 to phone 2. I was wondering if anybody could help? Colin Mason

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Voice broadcasting
by BookBag 25 Aug '12

25 Aug '12

Hello all, can anyone direct to some information on how I can setup a voice broadcast on FS. I tried looking in the wiki but it justs redirects you to heading "Robocalls", which in turn redirect you "voice broadcast" but there is no information under it. Basically I just want to call about 100 people or so and play a recording and then hangup. If anyone could help me , i'd really appreciate it.

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Wideband Skype to SIP?
by Emrah 25 Aug '12

25 Aug '12

Hi guys, I am looking for a SIP based solution to interconnect with Skype, with wideband codec support if possible. Does that exist? Otherwise, does Skypopen run on Debian Squeeze? Best, Emrah

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Re: [Freeswitch-users] [Freeswitch-dev] Scaled or distrbuted mod_conference?
by Muhammad Shahzad 24 Aug '12

24 Aug '12

hummm, interesting, i was just wondering if it is possible to make a tree shape conferencing bridge which would work something like this. 1. We will have many freeswitch boxes, each will running a single conference of its own up to say 50 users. These freeswitch box may be geographically distributed as needed, (though it may cause significant latency). 2. Then there is a freeswitch box running on top, all freeswitch boxes dial this top level freeswitch box as normal call. On top level freeswitch these calls will be answered and joined in a single so called top level conference. The second leg of this call will be joined to conference already running in lower level freeswitch boxes. 3. The main speaker / teacher of this virtual class room will just dial to top level freeswitch and join in the top level conference.. 4. For listeners / students of this virtual class to dial in and join, we can setup an OpenSIPs / Kamailio boxes which authenticates users first then forwards call to lower level freeswitch boxes. 5. All student calls will be mute when they join the conference. The teacher / speaker will have a web management interface which will query each lower freeswitch box and generate a participants list. Teacher will be able to unmute any student to listen to their questions etc. and then mute them again to answer the question etc. What do you guys think? Please suggest. Thank you. On Fri, Aug 24, 2012 at 1:15 PM, Stephen Dame <sdame(a)207me.com> wrote: > ** ** > > *I know a few commercial companies have scaled freeswitch to handle > 250-500 callers in a single conference across boxes with direct audio cable > connections in data center.* > > * * > > * I have an existing application that uses speex16(flash voip) and I can > get 50 callers in a single conference before cpu gets to 70% on a c1.medium > instance. I know I can throw more hardware at the problem, but > interested in bridging the same conference number between multiple > freeswitch instances and presenting single ESL notifications back to the > existing application so all users events are seen.* > > * * > > *This is a distance learning app, and not business audio conference, so > some latency is tolerable. I would locate the multiple freeswitch servers > in same zone and possibly use the new high I/O EBS instances running in SSD. > * > > * * > > *Any thoughts on this?* > > * * > > *Regards,* > > *Stephen* > > * * > > * * > > _________________________________________________________________________ > 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-dev mailing list > FreeSWITCH-dev(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk(a)hotmail.com Email: shaheryarkh(a)googlemail.com

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Compilation error on debian
by Saugort Dario Garcia Tovar 24 Aug '12

24 Aug '12

Hi guys, I have got a error trying to compile FS in debian. I have configured a small virtual machine, installed debian 6.2. I pull FS from git. I did bootstrap and compile. When I invoke make I got this: make "OUR_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /usr/local/src/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-all ; done )"; echo $mods )" "OUR_CLEAN_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /usr/local/src/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-clean ; done )"; echo $mods )" "OUR_INSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /usr/local/src/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-install ; done)"; echo $mods )" "OUR_UNINSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /usr/local/src/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" "OUR_DISABLED_MODULES=$(tmp_mods="$(grep "#" /usr/local/src/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-all ; done )"; echo $mods )" "OUR_DISABLED_CLEAN_MODULES=$(tmp_mods="$(grep "#" /usr/local/src/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-clean ; done )"; echo $mods )" "OUR_DISABLED_INSTALL_MODULES=$(tmp_mods="$(grep "#" /usr/local/src/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-install ; done)"; echo $mods )" "OUR_DISABLED_UNINSTALL_MODULES=$(tmp_mods="$(grep "#" /usr/local/src/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" `test -n "" || echo -s` all-recursive quiet_libtool: compile: gcc -DHAVE_CONFIG_H -I. -I./src/include -I./libs/xmlrpc-c -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -pthread -DLINUX=2 -D_REENTRANT -D_GNU_SOURCE -D_LARGEFILE64_SOURCE -I/usr/local/src/freeswitch/libs/apr/include -I/usr/local/src/freeswitch/libs/apr-util/include -I/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib -I/usr/local/src/freeswitch/libs/stfu -I/usr/local/src/freeswitch/libs/sqlite -I/usr/local/src/freeswitch/libs/pcre -I/usr/local/src/freeswitch/libs/speex/include -Ilibs/speex/include -I/usr/local/src/freeswitch/libs/srtp/include -I/usr/local/src/freeswitch/libs/srtp/crypto/include -Ilibs/srtp/crypto/include -I/usr/local/src/freeswitch/libs/spandsp/src -I/usr/local/src/freeswitch/libs/tiff-3.8.2/libtiff -DENABLE_SRTP -DSWITCH_HAVE_ODBC -I/usr/include -I/usr/local/src/freeswitch/libs/libedit/src -DSWITCH_HAVE_LIBEDIT -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement -g -O2 -MT libfreeswitch_la-switch_core_session.lo -MD -MP -MF .deps/libfreeswitch_la-switch_core_session.Tpo -c src/switch_core_session.c -fPIC -DPIC -o .libs/libfreeswitch_la-switch_core_session.o *cc1: warnings being treated as errors src/switch_core_session.c: In function 'switch_core_session_thread_pool_worker': src/switch_core_session.c:1478: error: format '%ld' expects type 'long int', but argument 9 has type 'switch_size_t' src/switch_core_session.c:1483: error: format '%ld' expects type 'long int', but argument 9 has type 'switch_size_t' make[1]: *** [libfreeswitch_la-switch_core_session.lo] Error 1 make: *** [all] Error 2 * What is wrong? Thanks in advance

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FreeSWITCH Auto Dialer - Open source
by Obie Brown 24 Aug '12

24 Aug '12

Hi, I really want to work with an Open Source dialer and FreeSWITCH but the only thing I can find for this is Newfies Dialer which does not have features of a Auto Dialer as I need. Also its use for Voice Broadcasting is limited. My question is. What other Auto Dialers that run on FreeSWITCH are there. If any. Thanks, Obie Brown

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Dial out within a conference with early media
by Emrah 24 Aug '12

24 Aug '12

Hi guys, Is there an option to dial with early media support from within a conference? This is what happens now: 1. I have an active conf. 2. I decide to pull in an additional user by calling her mobile. 3. From fs_cli, I execute conference <conf-id> dial Sofia/gateway/xyz/1234567890. Then 4. Either she picks up, we her the enter sound and we know she's joined in. 5. Or her mobile is switched off, with no voicemail and just an INFO message... And we will never know what happened. Is there a way to accomplish the following? Have the conference broadcast the early media portion of the call, and still play the enter sound when the called party picks up? If this is not possible, I think it would be a great addition to the conferencing module of FS to have a dedicated dial command with support for Early Media. Best and thanks, Emrah -- Emrah “In theory, theory and practice are the same. In practice, they are not.” Albert Einstein

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