March 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Radius/Authentication/Authorization
by Abid Saleem 31 Mar '10

31 Mar '10

Hi Neal and other Contributors to FS, I recieved an answer on the list that mod_rad_auth is ready. I upgraded FS to download and install it by "make current" and it is successfully built and installed. Could you please outline the detailed steps to do all this configuration. If it is still not ready, is there any other method that somebody has already implemented like perl scripts etc. Any help in this regard is badly required. THanks for your cooperation. Regards----------Abid SaleemSr. Product Manager _________________________________________________________________ Hotmail: Trusted email with powerful SPAM protection. https://signup.live.com/signup.aspx?id=60969

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Newbie MWI question
by lloyd thomas 31 Mar '10

31 Mar '10

I am more or less using the default config files with FS. I have a Mitel 5212 phone, which I leave a message for but the MWI light never illuminates Any Ideas?

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DTMF
by Sean Holt 31 Mar '10

31 Mar '10

Hello, I¹ve posted on the question on this topic before but I¹ll try again with more detail. I have this svn version FreeSWITCH Version 1.0.trunk (16952) SIP provider: Bandwidth.com I¹m assuming that DTMF rc2833 is the default, but I can¹t get my ivr to recognize the incoming call tones. Bandwidth can set up either inband or RC2833. They require these codecs, G.711Ulaw and G.729A. The tech at Bandwidth wanted me to setup inband to test if that works better, but wasn¹t sure where to set it up. I added this to my dialplan <extension name="main_number_23" continue="true"> <condition field="destination_number" expression="^3000$"> <action application="answer"/> <action application="start_dtmf"/> <action application="sleep" data="1000"/> <action application="conference" data="conference_1@conference_1"/> </condition> </extension> Should I add it to my sip trunk? Not sure how to troubleshoot this issue Thanks Sean

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Favorite SIP Providers? Australian providers?
by Fraser Redmond 30 Mar '10

30 Mar '10

I was looking through the list of Providers with example configs in the wiki: http://wiki.freeswitch.org/wiki/SIP_Provider_Examples I was wondering if anyone had a provider that they really loved? And why? I've got inbound 1800/0800 in UK & USA already, but might change. Also interested in outbound. Also, I want to get a 1800 number in Australia - any suggestions? Sorry in advance if this is one of those questions that does the rounds a couple of times a year (I did have a look through the archives via Google and didn't see anything), or if I'm treading on dangerous ground asking for recommendations of commercial services - so I guess only recommend services you use, not services you provide ;-) Cheers, Fraser

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Bridge two calls
by Ken Fulmer 30 Mar '10

30 Mar '10

I need to bridge two calls together: one from an ITSP and one from an internal PBX. I've tried the following syntax but can't get it to work. Can someone clarify what I'm doing wrong? Here are the two gateway configs located in the Sip Profiles / External directory: <include> <gateway name="IPCM-VLAB-1"> <param name="proxy" value="10.10.3.10"/> <param name="register" value="false"/> <param name="context" value="public"/> </gateway> </include> <include> <gateway name="SDC1-VGW-1"> <param name="proxy" value="172.16.15.11"/> <param name="register" value="false"/> <param name="context" value="public"/> </gateway> </include> Here is the inbound route in the Dial Plan / Public directory (I'm just trying to get one number working at the moment): <extension name="From SDC1-VGW-1"> <condition field="destination_number" expression="^(2053148491)$"> <action application="bridge" data="sofia/external/IPCM-VLAB-1"/> </condition> </extension> I've also tried the gateway statement with no luck either: <action application="bridge" data="sofia/gateway/mygatewayname/555123456" /> Any ideas? Thanks, Ken Fulmer

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Cisco 7940 To Private
by Troy Anderson 30 Mar '10

30 Mar '10

I have some Cisco 7940's registered and working well with FS, but whenever I dial out from the Cisco phone, it indicates the call is "To Private" rather than showing the number I'm calling or something. I've put the SIP trace in pastebin http://pastebin.freeswitch.org/12589 - there is a Remote-Part-ID field that seems OK. Does anyone know what Cisco is looking for here? Thanks, Troy

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domain name in shared DB
by Madovsky 30 Mar '10

30 Mar '10

2 nodes in shared DB, node1 and node2 respectively with hostname node1.mydomain.org node2.mydomain.org the profile name is called mydomain.org which is shared by the 2 nodes. a user registers on node1 and wait calls. someone from external gateway call him, so user(a)node1.mydomain.org or user(a)node2.mydomain.org both it works well. but from user(a)mydomain.org it reaches the callee one time on three. my question is: can I avoid NAPTR SRV rules and replace it by a TCP load balancing server ? (if I use full TCP for SIP messages and UDP as normal after media negotiation) because it seems that NAPTR SRV work only if the caller proxy understands it. with a TCP load balancer I can avoid this, but not sure if it won't break the SIP protocol. Thanks Franck

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Snom wiki
by Klaus Hochlehnert 30 Mar '10

30 Mar '10

Just saw that the Snom interoperability page for FS was created/updated: http://wiki.snom.com/Interoperability/PBX/FreeSwitch Cheers, Klaus

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ringback with unsupported codec
by Fabio Pietrosanti (naif) 30 Mar '10

30 Mar '10

Hi all, i would like to provide ringback in early media using AMR that's supported only in passtrough. Would it possible, providing static AMR sample to be played back, to play it? Fabio Pietrosanti

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Bandwidth.com
by Sean Holt 30 Mar '10

30 Mar '10

Hello, Just received my bandwidth.com provisioning email with my sip_trunks and DID numbers. However, part of the instructions say I can¹t use a username and password when connecting to the gateway. I tried leaving the username and password blank in the gateway sip profile but it keeps giving me an error message, Error: username param is REQUIRED. Anyone ever setup a gateway with Bandwidth and how does one get around the username param requirement. Thanks Sean

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