July 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Virtual DB FreeSWITCH
by Usama Zaidi 30 Jul '15

30 Jul '15

Hi all, I wanted to know if there's something similar to http://www.opensips.org/About/News0035 available in freeSWITCH, I want to write CDRs etc to DB but my environment doesn't support multicast/floating IPs. Regards. -- I'd love to change the world, but they wont gimme the source code to it

1 0

ignore_early_media=true behaviour
by Vishal Sharma 30 Jul '15

30 Jul '15

Hi All, I see a strange behaviour for ignore_early_media variable. I need to play a hold music file on LegA while Freeswitch tries to bridge call to one of the listed numbers. Here in dial plan I have used only one number. If I use following dialplan without ignore_early_media=true in bridge, Leg A is able to disconnect the call when Freeswitch is trying to bridge the call to one of the number in legB. <extension name="SIP-TO-SIP"> <condition field="destination_number" expression="^91(\d+)$"> <action application="answer"/> <action application="set" data="hold_music=/srv/sounds/123.mp3"/> <action application="bridge" data="{caller_id=+2000,ignore_early_media=true,RECORD_STEREO=true,originate_timeout=150,hangup_after_bridge=true,legA_UUID=${uuid}}sofia/gateway/anveo/${destination_number}"/> <action application="hangup"/> </condition> </extension> caller ------------------FS-------------------Callee INVITE-------------->| <-------------------100 <-------------------200 ACK-------------------> ###############| INVITE------------------> ###############| <------------------------100 BYE------------------>| <-------------------200| ###############| CANCEL------------------> ###############| <------------------------200 ###############| <---------------------------487 ###############| ACK-----------------------> But when I include ignore_early_media=true in bridge command, Freeswitch ignores BYE sent by legA ,when it is trying to bridge the call with leg B. caller ------------------FS-------------------Callee INVITE-------------->| <-------------------100 <-------------------200 ACK-------------------> ###############| INVITE------------------> ###############| <------------------------100 BYE------------------>| BYE------------------>| BYE------------------>| BYE------------------>| BYE------------------>| BYE------------------>| ###############| <---------------------------486 ###############| ACK-----------------------> <------200 ok BYE I would like to end the call if Leg A disconnects the call while I want to keep ignore_early_media = True so that I can play a hold music file instead of letting legA user listen to call failure message when FS tries multiple numbers Any suggestions .... Thanks a lot Vishal Sharma -- SuperReceptionist is now available on Android mobiles. Track your business on the go with call analytics, recordings, insights and more: Download the app here <https://play.google.com/store/apps/details?id=com.superreceptionist>

2 1

recording ring tong
by Pete Kay 30 Jul '15

30 Jul '15

Hi I am trying to record the incoming ring tone. Does anyone know how to do that? The record app does not seem to be able to record ring tone. Thanks, Pete

3 3

FS and T.38 fax
by Mimiko 30 Jul '15

30 Jul '15

Hello. I'm on FS 1.3.13b_cad607d72e. I have two D-Link VoIP gateways DVG-5008S and DVG-2024. Also there are two Canon MFP with faxes. Each DVG have on one port connected a Canon MFP. The FXS ports are configured to use T.38 for fax. Sending fax from one MFP to another starts, but ends with error. Switching ports to use T.30 allows to send and receive fax between MFPs. Doesn't this version of FS support T.38? Does latest version support? Or I miss something in configuration? -- Mimiko desu.

3 8

Bridge incoming call to internal extension and to external gateway
by Vivat 29 Jul '15

29 Jul '15

Hi, I try to ring an incoming call simultanously at my office phone (Snom IP phone) and at my mobile (via a second SIP gateway). Actually it should be simple: bridge with two targets. So I set up in the dialplan something like <action application="bridge" data="user/1000,sofia/gateway/MyTelco/[my mobile no]"/> But the office phone rings only for less then a second and ends then with "Call completed elsewhere". That means I can not answer the call at the office phone. The forwarding to the mobile works perfect. Each single target works (i.e. if I remove the mobile gateway from "bridge" then my office phone rings for longer and I can answer the call there). I read the wiki for the FollowMe dialplan but that is actually not what I want. There should be no first phone with a timeout and then ringing the second phone but both phones should ring in parallel and the phone with the first answer wins. Any hints how to solve this? Thanks in advance.

2 1

It's not too late to register for the OpenSIPS Workshop @Cluecon Monday August 3, 2015
by Alex Goulis 29 Jul '15

29 Jul '15

There's still room for the OpenSIPS Workshop @ClueCon. The OpenSIPS Workshop is a new format that will focus more on providing attendees with more of a technical approach to integrating OpenSIPS rather than case studies - the sessions will provide working setups (with explanations) for OpenSIPS in different scenarios . The Workshops will cover new OpenSIPS release and features, FreeSwitch integration , Edge Proxy setup, Asynchronous I/O support and Fraud prevention. The Official schedule can be found at: http://www.opensips.org/Community/Workshop-2015Chicago-Schedule But we're happy to announce the following workshop leaders: Razvan Crainea - OpenSIPS project Dan Christian Bogos - ITsysCOM Vlad Paiu - OpenSIPS Project Eric Tamme - OnSIP Alex Goulis - Ratetel Pete Kelly - SourceVox Whether you are a beginner or a seasoned professional, the OpenSIPS Workshop @ClueCon has something for you. For beginners, get answers to questions like: * how to extend, expand and enhance FreeSwitch with OpenSIPS * intelligent load balancing of freeswitch servers * why OpenSIPS is the right choice for your network * Building a calling card solution using OpenSIPS and Freeswitch Seasoned professionals can: * Learn about new modules and features for the 2.x release * Discuss advanced topics * Discuss the development roadmap for OpenSIPS Once we've wrapped up the workshop, head over to the Hackathon with the OpenSIPS Team. The registration fees (per person) are: $199 for those attending only the OpenSIPS Workshop $149 for those attending ClueCon as well or having a discount code (*CLUECON2015*) or are registering 3 or more participants Space is limited, so don't wait too long to register. More information can be found at: http://www.opensips.org/Community/Workshop-2015Chicago We're looking forward to seeing you in Chicago! - The OpenSIPS Team

1 0

G729 both in hardware and bypass
by Guillermo Ruiz Camauer 29 Jul '15

29 Jul '15

OK, this is probably not a usual setup, but I would like to know if it is possible: I have a FS server with 2 IP trunks configured. It also has a Digium hardware transcoding card which I use for 120 sessions of G729. I would like to set FS up in such a way that one of the IP trunks uses G729 in bypass mode. It is an outbound IVR and all voice prompts are pre-recorded in G729. I would like that the other trunk use the Digium transcoding hardware. Is there any way to configure this? Thanks in advance, -- Guillermo Ruiz Camauer

1 0

freeswitch never seems to call sendmail when trying to email voicemails
by Jason Lewis 29 Jul '15

29 Jul '15

Hi, I've been trying to make freeswitch email voicemails but as far as I can tell, it never even calls sendmail. I have setting mailer-app to "sendmail" and "/usr/sbin/sendmail" to no avail. I can successfully send an email from the commandline using sendmail. (sendmail in this case is provided by postfix) I see no emails in the postfix mail logs when I leave a voicemail message. I also tried creating a shell just to see if it even gets called from fs, but it does not get called when a voicemail is deposited: #!/bin/bash echo $(date --rfc-3339=ns): $* >> /tmp/freeswitchsendmail.log After every change, I have run reloadxml and reload mod_voicemail. I have also tried restarting freeswitch. I am running the debian packages of FreeSWITCH Version 1.4.15-1~64bit (-1 64bit) my configuration is based on the vanilla configuration with only very minor changes. I'm at a loss as to how to debug further, but I'm pretty sure the mailer-app is never called. Is there some setting I'm missing or something obvious I'm not doing? My config: 1001.xml: <include> <user id="1001"> <params> <param name="password" value="$${default_password}"/> <param name="vm-password" value="1001"/> <param name="vm-mailto" value="jason(a)xyz.com.au"/> <param name="vm-mailfrom" value="vm(a)freeswitch.xyz.com.au"/>  </params> <variables> <variable name="toll_allow" value="domestic,international,local"/> <variable name="accountcode" value="1001"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Extension 1001"/> <variable name="effective_caller_id_number" value="1001"/> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/> <variable name="callgroup" value="techsupport"/> </variables> </user> </include> and in switch.conf.xml I have the following set: <param name="mailer-app" value="/usr/sbin/sendmail"/> <param name="mailer-app-args" value="-t"/> I made a log at level 7 and put it on the pastebin: https://pastebin.freeswitch.org/23921 Jason Lewis http://emacstragic.net

7 16

[ERR] sofia_glue.c:313 Invalid tls-verify-policy value: none
by Kamil Nigmatullin 29 Jul '15

29 Jul '15

After fresh installation of 4.20 I keep getting this erroe sometimes 2015-07-29 08:54:37.376432 [ERR] sofia_glue.c:313 Invalid tls-verify-policy value: none Can I fix it somehow? Thank you -- Kamil Nigmatullin Tel: 77272323748 mob: 7 (707) 2517003 Skype: kamil.nigmatullin

1 0

Ad hoc conference
by Yehavi Bourvine 29 Jul '15

29 Jul '15

Hello, It has been some time since the last time I wrote a dialplan for Freeswitch, so my mind is slightly rusted :-) I've been asked to do ad-hoc conferencing support for multi users conference: A calls B, puts him on hold, calls C, connects all together, put them on hold, call D and connects all together, and so on. This is the way they want it, like the old PBX we have. Is there some example of how to do such a thing? if not, I thought of the following: - When A presses *1 the first time, a conference room is created and the other party is transfered to it. - For the second and others: A initiates a new call, and when connected press *2 to connect the new party as well as A into the conference room, What I don;t know how to do is: How to take a bridged call, split it into two legs and move one or both into the conference room. Any idea will help. Thanks, __Yehavi:

3 3