January 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Kudos to the FreSWITCH team!
by David Villasmil 03 Jan '13

03 Jan '13

Hello FS team, I'm just writing thie short email to congratulate you guys for the amazing work you're doing with FS since the very beginning. As some of you may know, some time ago I developed the FreeSWITC-Billing platform on top of FreeSWITCH (although it's been some time since I updated it). Some of my customers have been using it for 2 years now and have good traffic running on it. Long story short, I was called by 2 of my customers who were in turn called by a couple of their providers asking to please, please, pretty please slow doen the rate with which my customers were sending calls to them or to limit the channels, because my customers were bringing their soft-switches DOWN... I don't know the other switch's brand , but I know one is a Quintum. So, like I said: KUDOS! Cheers to all and Happy New Year!! David Villasmil https://github.com/davidcsi/FreeSWITCH-Billing

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Problem with Polycom
by Abaci 03 Jan '13

03 Jan '13

We have OpenSIPS in front of FreeSWITCH, phones register to OpenSIPS, OpenSIPS sends all calls to FreeSWITCH. the way we have it set up is that both OpenSIPS and FreeSWITCH auth all the invites (we have it like this for a reason). this setup works fine (with Yealink at least). Problem we have is with Polycom phones, the phone registers just fine but can't make calls. When OpenSIPS gets the INVITE it responds with a 407, to which the Polycom responds, OpenSIPS is then sending the call to FreeSWITCH which sends a 407 again, OpenSIPS proxies it back to the polycom, however the Polycom does not respond to the challenge. I don't think this is a FreeSWITCH problem, I would still like to hear if anyone has an idea how I can get this working (config option?) Thanks

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implementing missed call without a race
by Neil Patel 03 Jan '13

03 Jan '13

Hi All, I recently implemented "missed call" functionality into my lua app: an inbound call comes in, I check the DB if it should be a "missed call". If so, the call is terminated and the caller is called back through an outbound call. This is useful in places like India where people are charged for outbound calls only. Thanks to help from folks on FS IRC, I implemented like so: 1. script1.lua takes the inbound call 2. If missed call is to be made, invoke script2.lua (passing script1's uuid) through a bgapi call, and let script1.lua terminate 3. in script2.lua, uuid_kill the session from script1.lua, and make an outbound call to the original caller Two questions: 1. Are there any problems with this implementation from correctness and performance standpoints? 2. I am using this app over a TDM (PRI) connection. During very heavy load, there is a race condition: after killing the original session but before initiating the new outbound call in script2, the available channel could be grabbed. Is there any way to fix or mitigate this race? Thanks, Neil

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Need to recompile fs after ODBC version update?
by Dennis 03 Jan '13

03 Jan '13

hi, i just like to ask, if someone could tell me, if i have to recompile fs after updating the odbc-version on the server? thanks & kind regrads dennis

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FAX in Freeswitch
by Archana Venugopan 03 Jan '13

03 Jan '13

Hi, Is it possible to send/receive fax using Freeswitch? If so can you please let me know what changes need to be made? Regards, Archana

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Sofia fails to detect offline gateway
by Steven Ayre 03 Jan '13

03 Jan '13

Hi Guys, A couple of our gateways have been shut down for hardware upgrades. However, the FreeSWITCH servers didn't spot the gateways go offline so they're still trying to send calls to them. I see this in the logs: 2011-01-11 09:35:05.396610 [WARNING] sofia.c:3981 Ping failed gw4 with code 503 - count 1/0/10, state UP 2011-01-11 09:35:05.396610 [WARNING] sofia.c:3981 Ping failed gw3 with code 503 - count 1/0/10, state UP 2011-01-11 09:35:13.408674 [WARNING] sofia.c:3981 Ping failed gw4 with code 503 - count 1/0/10, state UP 2011-01-11 09:35:13.408674 [WARNING] sofia.c:3981 Ping failed gw3 with code 503 - count 1/0/10, state UP 2011-01-11 09:35:21.420884 [WARNING] sofia.c:3981 Ping failed gw4 with code 503 - count 1/0/10, state UP 2011-01-11 09:35:21.420884 [WARNING] sofia.c:3981 Ping failed gw3 with code 503 - count 1/0/10, state UP 2011-01-11 09:35:29.458114 [WARNING] sofia.c:3981 Ping failed gw4 with code 503 - count 1/0/10, state UP 2011-01-11 09:35:29.458114 [WARNING] sofia.c:3981 Ping failed gw3 with code 503 - count 1/0/10, state UP 2011-01-11 09:35:37.472424 [WARNING] sofia.c:3981 Ping failed gw4 with code 503 - count 1/0/10, state UP 2011-01-11 09:35:37.472424 [WARNING] sofia.c:3981 Ping failed gw3 with code 503 - count 1/0/10, state UP 2011-01-11 09:35:46.475934 [WARNING] sofia.c:3981 Ping failed gw4 with code 503 - count 1/0/10, state UP 2011-01-11 09:35:46.475934 [WARNING] sofia.c:3981 Ping failed gw3 with code 503 - count 1/0/10, state UP 2011-01-11 09:35:55.489967 [WARNING] sofia.c:3981 Ping failed gw4 with code 503 - count 1/0/10, state UP 2011-01-11 09:35:55.489967 [WARNING] sofia.c:3981 Ping failed gw3 with code 503 - count 1/0/10, state UP ... you get the idea. I've done some testing and it happens both when FS is running and the gateway goes offline (code 408 then) and when FS starts up with the gateway already offline (code 503 as above). Both IP addresses are entirely offline. I had expected these would go to state DOWN but they haven't, I just get the above repeating. The gateway definitions look like this: <gateway name="gw3"> <param name="realm" value="a.b.c.d"/> <param name="username" value="nouser"/> <param name="password" value="nopass"/> <param name="register" value="false"/> <param name="caller-id-in-from" value="true"/> <param name="ping" value="5"/> <param name="ping-max" value="10"/> <param name="ping-min" value="1"/> <param name="retry-seconds" value="5"/> <param name="expire-seconds" value="60"/> </gateway> Is there a mistake in my configuration? -Steve

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Naming convention - sip_profile_name / sofia_profile_name / sip_gateway_name
by Cal Leeming [Simplicity Media Ltd] 03 Jan '13

03 Jan '13

Hello, The following ticket has been raised in regards to the somewhat confusion naming convention of sip_profile_name / sofia_profile_name / sip_gateway_name. http://jira.freeswitch.org/browse/FS-4980 To avoid confusion, I have also updated the documentation to reflect the current behavior; http://wiki.freeswitch.org/wiki/Variable_sofia_profile_name http://wiki.freeswitch.org/wiki/Variable_sip_profile_name http://wiki.freeswitch.org/wiki/Variable_sip_gateway_name If anyone spots any mistakes in the above, please let me know. Thanks Cal

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mod_mp4 working and usage
by Santosh Kadam 02 Jan '13

02 Jan '13

Hello there, Can someone explain if the mod_mp4 is working ? If yes, can someone forward the procedure to make the same work ? Currently - I am using FreeSwitch with Xlite 5.0 as my SIP client. Does this client support playback of MP4 video? If not, please suggest an appropriate SIP client as well. thanks in advance for the response. Warm Regards santosh

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regarding attendent transfer
by veerabhadrarao` 02 Jan '13

02 Jan '13

hai Can anyone plaese tell me how to make attendent transfer. I am able to working on Blind transfer using default.xml and features.xml but not attendenet transfer successfully. I am using Freeswitch 1.2.4 version. thank you

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5 for 1 offer - senior/core devs
by Cal Leeming [Simplicity Media Ltd] 02 Jan '13

02 Jan '13

Hello, Sometimes it's difficult to get the attention of the core devs, they have to prioritize like everyone else. One of the approaches used on the Django mailing list was a 5 for 1 offer.. this is where you help out on 5 tickets (usually low priority ones), and a senior/core dev will look at one of your tickets in return (no guarantees of a fix, just purely to get their attention/comments) Would this approach be welcomed by the FS core team? Could be a great way of not only getting more tickets looked at, whilst providing a reward for doing so (as an alternative to bountys) Cal

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