October 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

anyone can help me to configure TLS + ZRTP
by dwi yulianto 28 Oct '13

28 Oct '13

i've try configure TLS and zrtp in freeswitch, but i've problem with TLS, i've using tutorial from this website wiki.freeswitch about SIP_TLS. when i activated in my softphone (i'm use phonerlite) it say TLS registered, but there is no sound feedback. i try using echo test by calling 9196 but there is no sound feedback. but, if i'm use TCP or UDP , there is no problem with that, i can hear voice feedback when try call 9196. someone told me to use ssldump about the result is in attahcment in this email is there any configuration when wanna try using TLS? thanks.

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Delete unused voice mail files
by Tim St. Pierre 28 Oct '13

28 Oct '13

Hello, I'm trying to find a good solution for distributed voice mail storage. Multi-master database seems to work well for the metadata, but I have yet to find an NFS based storage system that can do replication between sites, with both systems in use. I'm thinking of having a script fire after a message is left that will replicated new messages to all other systems. Since the filenames are unique, there is no danger in replacing other files, but messages would never be deleted. Is there a way to have FreeSWITCH compare the list of messages in the database with the voice mail directory and delete any files that are not referenced, or is that better done externally by looking through the database? Likewise, if someone knows of some kind of NFS server that will automatically copy or delete files to a mirrored system, that would be a better solution. The catch seems to be that I want to be able to read or write to EITHER system in the pair. Thanks! -Tim

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Adding headers to provisional responses
by Oleg Stolyar 28 Oct '13

28 Oct '13

Hi, I am trying to add a header to my responses according to this: http://wiki.freeswitch.org/wiki/Sofia-SIP#Adding_Response_Headers When I add a sip_rh_ header, it appears fine in my 200 OK response. However, when I try to add a sip_ph_ header, it does not appear in the 100 Trying response. Am I doing something wrong or is there a bug?

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Re: [Freeswitch-users] raspberry pi B FS latest not start
by Sergey Zhuravlov 28 Oct '13

28 Oct '13

Hi Twice tried to run the latest - both without success Compiles fine Before and after compiling and successfully launched a stable version -- WBR, Sergey GTALK/JABBER:4orbit@gmail.com

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connect two freeswitch boxes
by satya 28 Oct '13

28 Oct '13

Am trying to connect 2 freeswitch boxes following the procedure given in this link,http://wiki.freeswitch.org/wiki/Connect_Two_FreeSWITCH_Boxes . I am getting the following console log when I try to make a call.http://pastebin.freeswitch.org/21577 . I am a newbie, please help. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/connect-two-freeswitch-boxes-… Sent from the freeswitch-users mailing list archive at Nabble.com.

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FS + ZRTP + SAS
by Yuriy Nasida 28 Oct '13

28 Oct '13

Hi guys! I have problem with transmission of SAS (Short Authentication String).FS version 1.2.10 I enabled<param name="inbound-proxy-media" value="true"/>but SAS strings is still different. I also tried <param name="inbound-zrtp-passthru" value="true"/> but this didn't help. I heard that it is fixed with 1.3.X branch. Can somebody please confirm or deny this? Please advice.

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Sound issue on bridging one inbound gateway call to another gateway
by obbyone 28 Oct '13

28 Oct '13

Hello everybody, I need to bridge an inbound call from a gateway number to another gateway number (same provider). Call is made, but there is no sound exchange, nor any ringback while calling "GATEWAY_1_NUMBER". Here is the public dialplan (in /usr/local/freeswitch/conf/dialplan/public/.): <include> <extension name="EXT_NAME"> <condition field="destination_number" expression="^(GATEWAY_1_NUMBER)$"> <action application="bridge" data="sofia/gateway/GATEWAY_2_NAME/FIXED_NUMBER"/> </condition> </extension> </include> I have searched in this mailing list's archives but have not found any answer. Is there anyone who knows what appends and what can I do to make it work. Thanks in advance, TB -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Sound-issue-on-bridging-one-i… Sent from the freeswitch-users mailing list archive at Nabble.com.

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centos/rhel rpms not signed for 1.2.14 ?
by Ivan 28 Oct '13

28 Oct '13

Hi there, Did something change with how rpm packages are signed ? Yum complains that the 1.2.14 packages are not signed, which is indeed the case. 1.2.12 was OK: $rpm -K freeswitch-1.2.12-1.el6.x86_64.rpm freeswitch-1.2.12-1.el6.x86_64.rpm: (sha1) dsa sha1 md5 gpg OK 1.2.14 is not (missing gpg/dsa flags): $rpm -K freeswitch-1.2.14-1.el6.x86_64.rpm freeswitch-1.2.14-1.el6.x86_64.rpm: sha1 md5 OK I was waiting to see if other people had this problem - it's already 10 days since the rpms are available, so either I'm doing something wrong, or people don't care about package signing (bad!). The yum repo release rpm is set up with gpgcheck=1 but the wiki mentions using "--nogpgcheck" [1]. What's the "official" policy ? cheers ivan [1] http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide

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How to failover with fail_on_single_reject construct
by Johny Kadarisman Kwan 28 Oct '13

28 Oct '13

I’m attempting to setup FS to failover from gw1, gw2, and gw3 subsequently. AND if any gateway response with NO_ANSWER or ALLOTED_TIMEOUT, then the next attempt should be stop. Using “fail_on_single_reject” var, I tried following script : <extension name="public_extensions"> <condition field="destination_number" expression="^1234$"> <action application="sleep" data="2000"/> <action application="answer"/> <action application="bridge" data="{ignore_early_media=true,fail_on_single_reject=NO_ANSWER, ALLOTTED_TIMEOUT}[leg_timeout=20}]sofia/gateway/gw1/7321231234|[leg_timeout=20]sofia/gateway/gw2/7321231234|[leg_timeout=20]sofia/gateway/gw3/7321231234}"/> <action application="hangup"/> </condition> </extension> Bridge app still continue attempting call to each gw1, gw2, and gw3 when ALLOTED_TIMEOUT response. From searching lists, above dialplan should work out fine. Did I misses any steps? Btw, tried above on FreeSWITCH Version 1.5.6b+git~20131010T172322Z~1cd6d44b06 (git 1cd6d44 2013-10-10 17:23:22Z) windows built. Thanks, Jkwan

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ivr application
by Roman Gelfand 28 Oct '13

28 Oct '13

I am new to freeswitch and would like to develop an ivr application. I understand there are many ivr freeswitch-based platforms. Could you point me to the most widely used? Coding examples,etc...? Thanks in advance

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