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RTCP
by Alex Lake 26 Apr '13

26 Apr '13

Interesting to see a reference to RTCP in a recent thread. I looked it up on the wiki and there's not much there about how to use it for giving diagnostic output on actual calls. I would be interested in investigating (and would be happy to update the Wiki) but where should I start?!

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How to set Callee ID Name globally or on gateway level
by Deon Vermeulen 26 Apr '13

26 Apr '13

Hi I've stumbled upon the following links: http://freeswitch-users.2379917.n2.nabble.com/effective-callee-id-name-beha… http://lists.freeswitch.org/pipermail/freeswitch-dev/2011-November/005355.h… Am I correct to say that these set the callee_id_name to be passed in the SIP INVITE when configured on the extension? We require the following INVITE format to be sent over a specific gateway: INVITE sip:1234567890@192.168.200.153;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.200.130:5070;branch=z9hG4bK75b28f842f76d9c9e299a80c0;rport;X-DispCookie=1013;X-HwTRC=259446;X-DptMsg=1405 Route: <sip:192.168.200.153:5060;transport=udp;lr> Call-ID: bb4bcb34ec7041237fbfc1e63f687744(a)10.18.5.64 From: "0987654321"<sip:0987654321@192.168.200.130;transport=udp;user=phone>;tag=fd64e901-CC-1013-TRC-259446 To:"1234567890"<sip:1234567890@192.168.200.153;transport=udp;user=phone> How could one go about to achieve the above? Thanks for any help. Kind Regards Deon Vermeulen

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Friday Free For All Activate!
by Ken Rice 26 Apr '13

26 Apr '13

Hey Guys, Its that time of the week again... Lets get the Productivity Killer started... sip:888@conference.freeswitch.org Or if you need alternative access http://wiki.freeswitch.org/wiki/Weekly_Conference_Call_Calling_Instructions Join us... Todays topic is... Well its a free for all -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

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Cisco CUBE Gateway Configuration
by Alex Ynema 26 Apr '13

26 Apr '13

I'm trying to setup a Freeswitch system to talk to a Cisco CUBE [Cisco Unified Border Element] which only has an IP Address but I can't seem to find any information on how to get it working. Anyone got any details on configuring for one or working configs? Cheers Alex Ynema | IT Consultant alex(a)opensystems.net.au IT Consultant for Open Systems Support www.opensystems.net.au

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FSW not matching ACK transaction
by David Escartin 26 Apr '13

26 Apr '13

Hello all we are using freeswitch with a sip-proxy setup and we are trying to make match in a dialog with the ACK transaction, by receiving direclty the ACK from the UAC not passing through the proxy (as the initial INVITE does) this is the scenario for the INVITE UAC ----> PROXY ----> FSW in the proxy we are not adding record-route header, so when the UAC generates the ACK, it goes like this UAC ---------------------> FSW however, seems the FSW is not recognizing the ACK and keeps sending several 200OK back to the PROXY (because we did include Via: in the INVITE) for the INVITE here we have the parameters needed to match in the ACK construction Request-URI: sip:34666988297@79.170.68.132:5060 Call-ID: 61E072E1-A80F11E2-9934BA77-E98AD4CD(a)79.170.64.56 From: "EXTERNAL-CALL" <sip:34917019888@79.170.64.56>;tag=4B21F0A0-CFE for the ACK here are the parameters Request-URI: sip:34666988297@79.170.68.132:5060;transport=udp Call-ID: 61E072E1-A80F11E2-9934BA77-E98AD4CD(a)79.170.64.56 From: "EXTERNAL-CALL" <sip:7299@79.170.64.56>;tag=4B21F0A0-CFE maybe the difference in the user of the From and the "transport=udp" of the R-URI could explain the issue or do you know if it could be needed to add any configuration in the FSW to make the transaction match? thanks a lot and regards david escartin

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Skinny gateway
by Lesley Pervis 25 Apr '13

25 Apr '13

Hi all, I've been looking around, and I'm pretty sure it's not possible, but I thought I'd check here. mod_skinny seems to be written to allow SCCP devices to register with FS, and FS acts as a CallManager replacement However, I'm wondering if it's possible to configure FS to act as an SCCP client. I want to set up something similar to a Sofia profile gateway that, instead of registering with a SIP server, registers with CallManager so that FS can bridge calls between SIP devices and the SCCP-only CallManager. Thanks for any info, Les

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ignore polycom callforward when dialing a callgroup
by Jason Holden 25 Apr '13

25 Apr '13

Hi Currently I have users who wish to enable CF on their Polycom phone but once this is done the refer takes the call and it no longer rings the other phones in the callgroup. Does anyone have any suggestions?

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page application
by Abaci 25 Apr '13

25 Apr '13

There is an application "page" (in mod_dptools), I didn't find any documentation for it on the wiki so I'm sending here if someone can provide details how to use it. see http://fisheye.freeswitch.org/browse/freeswitch.git/src/mod/applications/mo… Thanks

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Freeswitch silent crash
by Antoine Reversat 25 Apr '13

25 Apr '13

Hi, I have a weird issue where once in a while (apparently randomly) freeswitch crashes silently (that is I don't see any mention of freeswitch having sgfaulted in the output of dmesg and the logs aren't helpgul either). The version of freeswitch is the latest 1.2 stable branch from git packaged by myself for ubuntu. The OS is Ubuntu 12.04 with the latest 3.2.0-39-generic #62-Ubuntu kernel. I have 2 servers and both randomly crash in the same way so I ruled out a hardware issue. Our whole call processing is done in LUA but the crash occurs with no call in progress. I know it's not a lot to work with but if you have pointers on how to get more information about what happens I'd be happy to try it out. -- Antoine Reversat Fibrenoire - www.fibrenoire.ca A: 550 ave Beaumont, suite 320, Montreal, Qc, H3N 1V1 T: 514 907-3002 x132 antoine(a)fibrenoire.ca <jf(a)fibrenoire.ca>

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Fwd: How to change Music-on-hold while digit_bind_action is used ? (Michael Collins)
by julien terrasson 25 Apr '13

25 Apr '13

Subject: Re: How to change Music-on-hold while digit_bind_action is used ? (Michael Collins) Hi Michael, Please find the debug information in the pastebin : http://pastebin.freeswitch.org/20831 Not sure this will be enought to understand the root cause. Don't hesitate to ask for more.. Regards, Julien On Wed, Apr 24, 2013 at 2:44 PM, julien terrasson < julien.terrasson(a)gmail.com> wrote: > Hi Michael, > > Thanks for your answer (quoted hereafter).. > > "To answer your initial question: yes, what you are trying to accomplish > is definitely possible. However if you want the caller ("A" in your > scenario) to press only a single key then you can't use bind_meta_app > (BMA). BMA requires two key presses: * and one other DTMF. That's why your > scenario requires ** to trigger." > > ... but i'm already using bind_digit_action :-) > > I probably gave too much information and made the problem description > unclear. > Sorry about that & let me try to explain again : > > Every time the star key is pressed, it will correctly execute the lua > script, no problem here.. But > > When executing that lua script : > The first time, the standard freeswitch moh is heard (on the b_leg) when > playAndGetDigits is called (on the a_leg). > The 2nd time, my custom freeswitch moh is heard (on the b_leg) when > playAndGetDigits is called (on the a_leg). > > Here are the instruction which are causing me headaches : > > */* Change the moh on the A leg */ > *api:executeString("uuid_setvar ".. b_leg_uuid .. " hold_music > /usr/local/freeswitch/sounds/fr/ca/june/voxService/*fr_wait_**email.wav* > ");* > /* Play and request digit (B leg) */* > digit = session:playAndGetDigits(1, 1, 1, 5000, '#', > '/usr/local/freeswitch/sounds/ > fr/ca/june/voxService/fr_ask_email.wav', '', '\\d+'); > > I was expecting fr_wait_email.wav to be played every time, but it look > like "uuid_setvar <b_leg_uuid> hold_music <custom_wav>" is not taken into > account on-the-fly, but only after bind_digit_action is finished. > Is that a normal behaviour ? If yes, what would be the correct way of > changing the moh when bind_digit_action is executed ? > > Regards, > > Julien > > > > > > >

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