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Best practices for parallel fork with redirect on one branch
by DanB 29 Oct '13

29 Oct '13

Hey Guys, Have met a situation where I need to do parallel call forking with bridge application and on one of the legs will receive a 302 which could mean another parallel fork there. Have tested the following scenarios: 1. Sip profile with manual-redirect=true. Here I fork the call using bridge with "," as separator and parallel fork works. The issue is that as soon as one of my forked legs receives 302, the other will be cancelled also and 302 will be handled further in it's context. Is this the expected behaviour from bridge? Question here: should it drop one branch of the parallel fork if other is receiving redirect? 2. Sip profile with manual-redirect=false, leaving mod_sofia doing the handling of the redirect. After I fork the call here with bridge, only the leg which receives 302 will be redirected which is the expected behaviour by me. The issue I have here is that I do not know how to influence mod_sofia to handle parallel forking out of 302 and not very obvious out of the documentation I have digged so far (I receive 2 contacts in redirect btw which I would like to fork also here). Question here: may I specify the type of fork mod_sofia should do in some way? Thanks in advance for any tip! DanB PS: Logs available at request if needed.

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Conference session doesn't get end in case of incoming conference.
by Harsimran singh 29 Oct '13

29 Oct '13

Hi, I am creating conference with dtmf facility . When session is created with outgoing call through freeswitch then conference session gets end when no member remains in the session or when no one picks the call , but the same scenario is not happening with the incoming conference session . I am using the same code for both the cases . Can anyone help me out in this ? With Regards Harsimran Singh +91-9711271158

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SRTP protection failed with code 9
by Vincent Xia 29 Oct '13

29 Oct '13

im using TLS + SRTP, when making calls, im getting "SRTP protection failed with code 9" error at the FS console all the time these days, the call is normal, but no SRTP, why is this happening? the log is pastebined at http://pastebin.freeswitch.org/21578 any response is appreciated.

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dev environment
by Valter Nogueira 29 Oct '13

29 Oct '13

I want to setup emacs and related tools in order to edit and mainly understand source code. Is there any tutorial about this? thanks Valter * *

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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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