- FreeSWITCH-users - FreeSWITCH Mailing Lists

Is there anyone who is connected to PCCW?
by Edmar Cruz 16 Jun '09

16 Jun '09

PCCW is use for making calls through IP connected through cellphone just enter the areacode for example 900639274522123 900-prefix 63-areacode 9274522123 - number? Has anyone has tried it? Please help me how to connect to it -- View this message in context: http://www.nabble.com/Is-there-anyone-who-is-connected-to-PCCW--tp24049302p… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Re: [Freeswitch-users] funny effect after minimizing xml files
by Durk de Beer 16 Jun '09

16 Jun '09

>> >> Hello I've minimized de xml files where possible to make a dialplan that is >> as short as possible. Now do I've this funny effect to dial my extensions >> who are running from 200 to 207. It seams that I'm able to dial an >> extension in closed in a number. So for instants if I dial 120275 extension >> 202 will ring even tried it whit two extensions in a number like 202205 . >> This results in the first extension ringing so 202205, 202 will ring >> 205202, 205 will ring. At this time I'm unable to pinpoint the cause of >> this behaviour. Could someone point me to the cause of this effect > I don't understand the problem, but my general advice is this: learn to read > the FreeSWITCH logs carefully. Make sure that the log level is set to "debug", > as it is in the default configuration, then carefully check the log files to > see which dialplan extension matched and how the call was processed. After reading this, a colleague of mine had a look at the logs and found out that we had goofed up the regular expressions in the dialplan. This made Freeswitch dial the number of an extension if its sequence was found in the dialed number so lets say the extension has number 202 and the number dialed was 15320264 it would find the 202 sequence in the dialed number and then it would dial the 202 extension. So it seems this one is a stupid mistake of us. Thanks to all that responded \d

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created external5090 on profile not working?
by Edmar Cruz 16 Jun '09

16 Jun '09

I created a profile name external5090 on /usr/local/freeswitch/conf/sip_profiles/external5090.xml... Change ext-sip-ip and ext-rtp-ip for a server 192.168.0.104 with sip-port: 5090... My local Ip is 192.168.0.105... I see it with I type it on the API freeswitch and type sofia status is there... How can I know that it is working? can u send me a API freeswitch for it? may code is originate sofia/external5090/1002@192.168.0.104:5090 5090 is this correct? -- View this message in context: http://www.nabble.com/created-external5090-on-profile-not-working--tp240482… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Access MySQL directly via Javascript using SSJS Engines .... "really!"
by EdPimentl 16 Jun '09

16 Jun '09

FYI Now you can directly connect to MySql via javascript http://jsext.webloji.net/ http://jsext.sourceforge.net/JSEXT1.Mysql.html http://www.brainonfire.net/blog/ssjs-on-ubuntu/ And if you wish you had a V8, check out Google SSJS Engine BTW: JS is now 7x faster than it last year. -E http://Gpro.ws http://twitter.com/edpimentl http://facebook.com/facevalu

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Is Freeswitch ready for prime time?
by Paul Mahler 16 Jun '09

16 Jun '09

Hello Everybody, I have a large project coming up. I'm interested in using Freeswitch instead of SER and Asterisk. What is the current status of Freeswitch? Can I safely use it in a large scale commercial environment? How active is the Freeswitch developer community? I am concerned that Freeswitch doesn't seem to have gained much traction compared to Asterisk. How viable is the Freeswitch project? Please help me understand why I can safely shit can Asterisk and move to Freeswitch. Thank You, Paul _________________ Paul Mahler paul(a)ringcarrier.com

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Gateway conference handshake trouble
by Lon Baker 15 Jun '09

15 Jun '09

Hi, We are trying to work freeswitch into an older system as a conference bridge. Our existing gateways can hand off the call and they pass some DTMF signals to route everything. Currently, the gateway sends * and then freeswitch returns a # to accept the call for the conference. When that transaction is done the gateway waits 2 seconds to send either a 0 or 1 and indicate if the caller is a moderator/admin of the conference. Everything works until we get to the moderator flag. That part never appears in the logs or debug info in the console. But the person on the call, can hit 0 or 1 and then enter the conference call correctly. If they don't it will timeout and drop them. >From what we know the gateway is sending the moderator DTMF flag, but freeswitch is only listening on the call for it. Any idea would appreciated. Lon

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Live Upgrade Techniques
by John Dalgliesh 15 Jun '09

15 Jun '09

Hi, I am slowly gaining confidence using FreeSWITCH in production, but there is one issue that I'm still wondering about: how are people upgrading their FreeSWITCH installation binaries without dropping all current calls? So far I have been upgrading in the dead of night, after pausing for 5 minutes then dropping the stragglers, but this is hardly ideal. What I would like to do is to run an upgraded instance of FreeSWITCH on the same machine, and have it handle all new call packets, whereas the old instance continues to handle the existing call packets, until there are no more old calls left. I can think of about seven ways to accomplish this, but before I dive into the code I thought I'd better ask what everyone else has been doing :) (The only standard way I can think of doing this is to have a SIP proxy sitting in front of FS the whole time, just to handle these upgrade windows. It seems like a bit of a waste.) So how are you handling your FS software upgrades? {P^/ John

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Asterisks to Freeswitch CALL REJECTED
by Edmar Cruz 15 Jun '09

15 Jun '09

I am trying to call Freeswitch using Asterisks and using a softphone X-Lite but the issue is call rejected by freeswitch? Is their any configuration files to allow asterisks to call to freeswitch? -- View this message in context: http://www.nabble.com/Asterisks-to-Freeswitch-CALL-REJECTED-tp24031735p2403… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Re: [Freeswitch-users] funny effect after minimizing xml files
by David Knell 15 Jun '09

15 Jun '09

You've probably deleted the start/end markers from your dialplan matches..? It might be easier to help if you posted or pastebinned your dialplan. --Dave > Hello I've minimized de xml files where possible to make a dialplan > that is as short as possible. Now do I've this funny effect to dial my > extensions who are running from 200 to 207. It seams that I'm able to > dial an extension in closed in a number. So for instants if I dial > 120275 extension 202 will ring even tried it whit two extensions in a > number like 202205 . This results in the first extension ringing so > 202205, 202 will ring 205202, 205 will ring. At this time I'm unable > to pinpoint the cause of this behaviour. Could someone point me to the > cause of this effect > > > /d > > > _______________________________________________ > 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 -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: dave(a)3c.co.uk W: http://www.3c.co.uk

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Re: [Freeswitch-users] funny effect after minimizing xml files
by dujinfang 15 Jun '09

15 Jun '09

how to help without seeing your dialplan? On Jun 15, 2009, at 6:26 PM, Durk de Beer wrote: > Hello I've minimized de xml files where possible to make a dialplan > that is as short as possible. Now do I've this funny effect to dial > my extensions who are running from 200 to 207. It seams that I'm > able to dial an extension in closed in a number. So for instants if > I dial 120275 extension 202 will ring even tried it whit two > extensions in a number like 202205 . This results in the first > extension ringing so 202205, 202 will ring 205202, 205 will ring. At > this time I'm unable to pinpoint the cause of this behaviour. Could > someone point me to the cause of this effect > > /d > > _______________________________________________ > 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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