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Pizza Demo Grammar
by Bryan Smart 22 Jul '11

22 Jul '11

Hi. Trying to install the Pizza demo. The Wiki says to download the grammar from www.bkw.org, but that entire site seems to be down. Are these files available somewhere else? I tried to find them with Google, but only found references to the Wiki page and references to it in FS updates. Thanks Bryan

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DTMF conversion / regeneration with bind_digit_action ?
by Dmitry Sytchev 22 Jul '11

22 Jul '11

Hi All DTMF troubles again :( We are trying to use Linksys PAP2T / SPA8000 with FS on internal profile and Audiocodes Mediant 2000 as gateway on external. When ATAs set to RFC2833 and proxy media is set on both profiles, everything works fine. We get RFC2833 on both sides. If we turn proxy media off to use freeswitch DTMF features like attended transfer by *7 or another in-call DTMF features, things go worse. PAP2T/SPA8000 convert inband DTMF from phone to RFC2833, but there are some inband clips still coming into audio channel. Freeswitch processes RFC2833 as dtmf-type set to 2833 in internal profile, but since we use bind_digit_action, little inband clip of DTMF gets to other side earlier than regenerated 2833, which is delayed by FS in bind_digit_action / bind_meta_app. BTW, bind_meta_app introduces less delay than bind_digit_action. As the result, after media goes through mediant2000, on PSTN side we have double DTMF - little inband clip and generated RFC2833 with really big gap between them. If some local transfers has been done via internal sip profile, more channels are linked in chain, so delay gets even bigger. When we set DTMF to inband on ATAs and internal profile dtmf-type is set to inband, freeswitch detects inband and in-call features work, but after passing inband DTMF through, FS for some reason also generates RFC2833 on external profile, and then Mediant passes inband part and re-generates rfc2833 DTMF part to PSTN. Again, double DTMF on PSTN side :( Can inband <-> rfc2833 conversion be done on FS side with inband DTMF suppression? And what can be done with in-call DTMF detection functions to make FS pass rfc2833 through as fast as it receives it without delay to make no gap between inband clips and regenerated DTMF? I really appreciate any help. I'm trying to handle this in different ways for two month with no result :( -- Best regards, Dmitry Sytchev, IT Engineer

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bridge_answer_timeout problem
by VoIPHwy 22 Jul '11

22 Jul '11

Hello, I am having a problem with bridge_answer_timeout I included the dial plan below. This does timeout and it does transfer but there is no audio when it goes to either voice mail or an ivr. If it goes to another extension or pstn number, there is audio. The trace shows messages being played. I did a wireshark trace and it did not show any rtp going out after the bridge_answer_timeout. In the dial plan, 8205 goes to a dial plan that is the voice mail for extension 205 and if I dial 8205 direct it works. I have installed the latest git. <include> <extension name="205"> <condition field="destination_number" expression="^(205)$"> <action application="set" data="proxy_media=true"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="hangup_after_bridge=false"/> <action application="bridge" data="{bridge_answer_timeout=5}sofia/${use_profile}/16133429652(a)toronto2.voip.ms"/> <action application="transfer" data="8205 XML office.voiphwy.com"/> </condition> </extension> </include> Any help would be appreciated. Darcy Primrose The VoIP Highway

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Complete wholesale app in freeswitch
by David Villasmil 21 Jul '11

21 Jul '11

Hello Guys, I'm finishing a "complete" wholesale application created on freeswitch and I was wondering whether it would be a good idea to put it up on the wiki. I just don't know how. Features include all the following parameters configurable via web interface: - Profile creation based on server IP where traffic is received. You can have multiple IPs, system will create multiple profiles/diaplans so it can differentiate. - i.e. offer to the same customer a "gold" routing on IP1 and cheap routing on IP2 - Customer add/modify/delete - IP source - Rates for client routes based on areacode - Prepaid or postpaid. - When cutomer balance is 0, no more calls are allowed. - limit max channels - Media by-pass - When by-passed, customer and provider will exchange RTPs directly. Else, server will be in the middle. - Provider add/modify/delete - costs for provider routes based on areacode - limit max channels - Routing based on areacode, gives great granularity. - Routes can be assigned multiple gateways/providers which can in turn be distributed based on weigth. Includes overflow to next configured GW. - Basic financial report generation (totals) by customer/provider - Basic traffic ASR/ACD report (totals) by cutomer/provider - Basic user administration. (No access level, only total access) - CDR export to csv file. I also have a prepaid card app... no web interface on that one though... Thanks all David

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Re: [Freeswitch-users] Different REGISTER and INVITE From:
by dave9876＠gmx.com 21 Jul '11

21 Jul '11

Hi, Thanks! I thought I had tried that, but apparently not properly. The next problem was that I needed P-Asserted-Identity set to "12345_01". I added "{sip_h_P-Asserted-Identity='sip:12345_01@gw.example.com',..." and it works fine. Thanks a lot! Dave ----- Original Message ----- From: Steven Ayre Sent: 07/20/11 05:55 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Different REGISTER and INVITE From: Try setting this in the gateway config:    <param name="caller-id-in-from" value="true"/> You'll need to do 'sofia profile NAME killgw NAME', 'sofia profile NAME rescan'. It'll then use the caller id in the From for the INVITE, so this should then work: originate {origination_caller_id_number='+12345'}sofia/gateway/gw.example.com/+10000 5000 -Steve On 20 July 2011 08:34, <dave9876(a)gmx.com> wrote: > Hi, > I'm trying to test a gateway with FS. > > The GW requires me to register with "From: 12345_01(a)gw.example.com", but when calling I need to have "From: +12345(a)gw.example.com" (with "+" and without "_01") > > If I try from the console: > originate {origination_caller_id_number='+12345'}sofia/gateway/gw.example.com/+10000 5000 >ld > I still get "From: 12345_01(a)gw.example.com", since I have to have "param name="from-user" value="12345_01" set in my GW profile. > > Any ideas on how to force a different "From" on the INVITE? > > I tried calling from a softphone registered with FS, but then I had trouble with the from-domain instead. > > Any help is much appreciated. > > > Dave > _______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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Google voice terminating outbound calls
by Rob Hill 20 Jul '11

20 Jul '11

I'm using mod dingaling to connect via Google Voice. Incoming and outgoing calls were working fine until just a few days ago, now outgoing calls are terminated after ~15 seconds. Incoming calls are still fine. Here is the disconnect info : 2011-07-20 17:36:20.485777 [INFO] libdingaling.c:1371 SecRECV: ------------------------------------------------------------------------------- <iq from="+1312xxxxxxx(a)voice.google.com/srvres-MTAuMTc3LjEyMi4xNDo5ODgz" to="r(a)gmail.com/talkCC63ED4F" id="jingle:10.177.122.14-2416924:1:3F7BB3A2" type="set"> <ses:session type="terminate" id="1514126677" initiator="r(a)gmail.com/talkCC63ED4F" xmlns:ses="http://www.google.com/session"> <ses:reason> <ses:timeout></ses:timeout> <ses:text>Session timed out</ses:text> </ses:reason> <pho:recipient-unavailable xmlns:pho="http://www.google.com/session/phone"></pho:recipient-unavailable> </ses:session> </iq> To make matters more confusing I've tested my configuration on two different hosts and outgoing calls work fine on the second host (using the exact same configuration files). Any ideas where this might be going wrong? a O

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ClueCon 2011 - Sharing rooms
by Michael Collins 20 Jul '11

20 Jul '11

Hello all! It seems that the Sofitel is getting booked up rapidly and it's becoming difficult to find rooms. If you are looking for someone with whom you may share a room, or you have a room that you would like to share, please email me off list. Thanks, Michael

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FreeTDM viewer
by Valery Kalinin 20 Jul '11

20 Jul '11

I'm write small web utility for FreeTDM channel statuses view: FreeTDM viewer Web-based online view FreeTDM spans & channels statuses. Download from: https://sites.google.com/site/freeswitched/home/downloads Installation guide: Upload files (ftdmv.php & jquery-1.6.1.min.js) to your web-server with PHP support. Check modules.conf and event_socket.conf for enable module mod_event_socket Change IP address, port, password for FreeSWITCH server and FreeTDM spans in ftdmv.php file Enter in browser http://you-server-name/ftdmv.php

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SIP Presence Subscription Management
by Dunkel, Christopher 20 Jul '11

20 Jul '11

I'm having an issue with SIP presence through Freeswitch and I'm hoping that someone might be able to help me determine if my problem is with Freeswitch or with the client(s) I'm using. The system I'm using includes Freeswitch (version 1.0.6) running on a Linux box and two soft phone clients running on iPhone and Android (PJSip on iPhone and Linphone on Android). The problem that I'm having appears the same on both clients, which leads me to believe that this may be a Freeswitch config problem. The clients are able to send a SUBSCRIBE message and receive a NOTIFY immediately when one of the subscribed users registers with Freeswitch. They also receive the offline NOTIFY when the user unregisters from Freeswitch. However, an online NOTIFY message is not sent when the user comes back online. It seems as if Freeswitch is clearing all presence subscriptions for a particular user when they unregister from the server. Is there any way to prevent Freeswitch from clearing these subscribes unless specifically directed to do so? Also, on a possibly related subject, I am only able to receive "online" and "offline" presence notifications. I only receive "offline" presence notifications when a user unregisters from Freeswitch. Attempting to send any other presence value from one of the clients (i.e. "busy", "be right back", "offline", etc...) only results in an "online" notification being sent. Has anyone else seen similar behavior? Thanks, Chris

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Different REGISTER and INVITE From:
by dave9876＠gmx.com 20 Jul '11

20 Jul '11

Hi, I'm trying to test a gateway with FS. The GW requires me to register with "From: 12345_01(a)gw.example.com", but when calling I need to have "From: +12345(a)gw.example.com" (with "+" and without "_01") If I try from the console: originate {origination_caller_id_number='+12345'}sofia/gateway/gw.example.com/+10000 5000 I still get "From: 12345_01(a)gw.example.com", since I have to have "param name="from-user" value="12345_01" set in my GW profile. Any ideas on how to force a different "From" on the INVITE? I tried calling from a softphone registered with FS, but then I had trouble with the from-domain instead. Any help is much appreciated. Dave

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