January 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Non-numeric extension - bad idea?
by Steven Schoch 31 Jan '13

31 Jan '13

We have a main number and extensions, but some people have their own DID number that rings on line 2 (of a 2-line Polycom phone), with a separate voice mailbox. This "line 2" is not supposed to be an extension, so in its directory entry, I have the "effective_caller_id_number" set to the DID number. I don't want this line to be a real extension, so I set the user id to be a descriptive short name, not a number. Is this a bad idea? The problem is that it's impossible to check voice mail on that line because the user can't dial the user id. Would it be better to use the 10-digit number as the user id? Or should I use an extension number that starts with a different digit? That would solve the voice mail problem as well, because the dialplan passes the caller-id to the voicemail check application. -- Steve

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Execute app in sync mode when using event sockets
by Hector Geraldino 31 Jan '13

31 Jan '13

Hi, I have a java app that uses mod_event_socket (inbound mode) to control FreeSWITCH. I can send api commands which, by default, are synchronous, so the thread locks until the command is successfully executed. The problem I'm facing is when I have to execute commands that are in other modules (like playback, or speak). As they are not api commands, I must send an execute appName + arguments, which by nature is executed in async mode. What I want to achieve is to execute an speak command and wait until the TTS operation ends (blocking the thread). This sounds like a trivial use case: play some audio (TTS, file) and then execute the rest of the logic. It is easy to accomplish on the dialplan, but I just can't figure out how to do it on an ESL application. Here's what I've tried so far: + send execute speak command + api sleep and wait for the CHANNEL_EXECUTE_COMPLETE (to simulate a lock [sleep] and continue [break after event is received]), but the event arrives after the sleep command completes. + use uuid_broadcast uuid speak::args, doesn't work Any hints, ideas are welcomed. Thank you! Hector

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Recording / concurrent calls and performance
by Malay Thakershi 31 Jan '13

31 Jan '13

Hello, Does recording audio sessions slow down performance of concurrent calls / FreeSwitch? I am doing: application="record_session" in the dial plan before transferring the call to managed DLL. Thanks for help.

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Routing inbound call to multiple extentions
by Todd Bailey 31 Jan '13

31 Jan '13

Hi All, At witt's end on this, how do I get a inbound call to go to multiple extension? I've tried groups but so far no avail I using a spa 3102 as a analog adapter that is configured to rout calls to extension 1000, I want FS to automatically forward this call to extensions 1001-1009. Here is an extract from default.xml (/usr/local/freeswitch/conf/directory) <group name="frontoffice"> <users>  <user id="1001" type="pointer"/> <user id="1002" type="pointer"/> <user id="1003" type="pointer"/> <user id="1004" type="pointer"/> <user id="1005" type="pointer"/> <user id="1006" type="pointer"/> <user id="1007" type="pointer"/> <user id="1008" type="pointer"/> <user id="1009" type="pointer"/> </users> </group>

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ZRTP Cache file location on FreeSWITCH fixed/updated
by Ken Rice 31 Jan '13

31 Jan '13

Hey Guys, If you have noticed theres been an issue with ZRTP enabled FreeSWITCH where ZRTP keeps a cache of hashs identifying previously contacted ZRTP endpoints. (This is part of the mechanism to help detect Man in the Middle attacks). Prior to today that cache file was being improperly named ! And was being dropped in the Directory FreeSWITCH was being ran from. Travis Cross and I got a patch together today and it has been pushed into tree. Starting today the file has been moved over to the proper FreeSWITCH db directory (/usr/local/freeswitch/db on default from source installs) with the name zrtp.dat . If you are currently using this functionality, you can keep your cache around by moving it to the new location. (ex: # cd /usr/local/freeswitch && mv \! db/zrtp.dat ) This patch is currently only in master, but will be rolled into the pending stable updates. Thanks for the help Travis! K -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

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Re: [Freeswitch-users] Postgres in the Core
by Andrew Cassidy 31 Jan '13

31 Jan '13

I guess my main question is do the other modules accept the new postgres:// syntax or do I still need to use odbc for those? > -- > Andrew Cassidy BSc (Hons) MBCS > http://www.jonisdumb.com/ > > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T <info(a)cassidywebservices.co.uk> *03300 100 960 *F<info(a)cassidywebservices.co.uk> *03300 100 961 *E <info(a)cassidywebservices.co.uk> *andrew(a)cassidywebservices.co.uk *W <info(a)cassidywebservices.co.uk> *www.cassidywebservices.co.uk

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IAX
by S. Scott 31 Jan '13

31 Jan '13

I've done the google searches and it's not clear what the current status of IAX facility is available in fs. We have a project needing IAX trunk with registration support. With thanks, ————— iThing: Big thumbs & little keys. Please excuse typo, spelling and grammar errors • Thought of the Day – "With all this manure, there must be a pony in here somewhere.”

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Limit calls per user and transferring issue
by Colin Mason 31 Jan '13

31 Jan '13

Hello, I am running 1.3.13b git 8859eb0 2012-12-30 19:08:55Z I am trying to use limit to keep the maximum calls (in+out) per registration at one. I am using 2 Linksys SPA303 phones both registered as extension 1000. I am using BLF and it works. I can transfer a call to the other phone by pressing hold on phone 1 and pressing the blinking BLF light on phone 2. I am using limit_ignore_transfer=true. If I transfer an inbound call to the other phone, the limit for that user persists and everything works. If I transfer an outbound call over to the other phone, the limit decrements to 0 which is unintended in my setup. I was wondering if there was a way to make transferring outbound calls to another phone have the same limit behavior as transferring inbound calls. I read that this can be done using loopbacks but I am unsure how to approach using a loopback to perform this function. Any help is much appreciated. Colin

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Two Freeswitches - Invalid Gateway
by Pete Ashdown 31 Jan '13

31 Jan '13

Hello, I'm trying to do something that appears outwardly simple, but is causing me grief. I asked the question on the IRC channel, but I'm still waiting for the ticket to be answered. I followed the instructions for connecting two Freeswitch boxes here: http://wiki.freeswitch.org/wiki/Connect_Two_FreeSWITCH_Boxes The result is that I get an "Invalid Gateway" when trying to dial extensions on either box, like it is trying to go through the SIP external profile. Any thoughts as to what I'm doing wrong, or what is missing from the above configuration? Thanks in advance.

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Re: [Freeswitch-users] DID providers - any thoughts? (jan 2013)
by Luke Milbourne 31 Jan '13

31 Jan '13

Hi Cal, Have you investigated AQL? I use them currently for my DID, not on the type of scale you are talking about but I have been impressed with their services so far. Luke On 29 January 2013 20:55, <freeswitch-users-request(a)lists.freeswitch.org>wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Re: SPA-2102 Gives 486 instead of Call Waiting (Gilad Abada) > 2. DID providers - any thoughts? (jan 2013) > (Cal Leeming [Simplicity Media Ltd]) > 3. Rejecting undesired codecs in bypass media mode > (Victor Chukalovskiy) > 4. Re: DID providers - any thoughts? (jan 2013) (Avi Marcus) > > > ---------- Forwarded message ---------- > From: Gilad Abada <ga(a)steadfasttelecom.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Tue, 29 Jan 2013 14:34:47 -0500 > Subject: Re: [Freeswitch-users] SPA-2102 Gives 486 instead of Call Waiting > I think if you turn debug and syslog on level 3 it gives more info > > Sent from my mobile device. > > On Jan 29, 2013, at 2:28 PM, Avi Marcus <avi(a)avimarcus.net> wrote: > > I did, that's how I got the log in my first email... > Was I supposed to find more than that? > -Avi > > > On Tue, Jan 29, 2013 at 9:08 PM, Gilad Abada <ga(a)steadfasttelecom.com>wrote: > >> Avi, >> >> You can try to enable syslog trough the GUI and see what the spa 2102 os >> actually doing. I think it's under the system tab you have syslog and debug >> server. >> >> Sent from my mobile device. >> >> On Jan 29, 2013, at 2:05 PM, Avi Marcus <avi(a)avimarcus.net> wrote: >> >> Any suggestions on how to figure this out? I'd really appreciate it! >> >> -Avi >> >> >> On Mon, Jan 7, 2013 at 3:12 PM, Avi Marcus <avi(a)avimarcus.net> wrote: >> >>> Here's an intersting one that seems to have no results on google or in >>> debug info: >>> >>> I have an SPA-2102 (firmware 5.2.10) that returns 486/Busy Here when a >>> call comes in to the active line. >>> >>> "call waiting" is turned on, so it should be beeping in instead... >>> >>> I got these logs off the SPA but it doesn't seem very helpful: (and >>> there should be more, I think) >>> >>> ==> /var/log/syslog <== >>> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il<http://client_ip.smc.012.net.il/>[0]<<$FreeSWITCH: 5060(1148) >>> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il<http://client_ip.smc.012.net.il/>[0]<<$FreeSWITCH: 5060(1148) >>> Jan 7 14:38:14 INVITE sip: 1632@$CLIENT_IP:5060 SIP/2.0#012 >>> Jan 7 14:38:14 INVITE sip: 1632@$CLIENT_IP:5060 SIP/2.0#012 >>> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il<http://client_ip.smc.012.net.il/>[0]->$FreeSWITCH: 5060(312) >>> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il<http://client_ip.smc.012.net.il/>[0]->$FreeSWITCH: 5060(312) >>> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il<http://client_ip.smc.012.net.il/>SIP/2.0 100 Trying >>> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il<http://client_ip.smc.012.net.il/>#012 >>> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il<http://client_ip.smc.012.net.il/>#012 >>> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il<http://client_ip.smc.012.net.il/>[0]->$FreeSWITCH: 5060(338) >>> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il<http://client_ip.smc.012.net.il/>[0]->$FreeSWITCH: 5060(338) >>> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il<http://client_ip.smc.012.net.il/>SIP/2.0 486 Busy Here >>> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il<http://client_ip.smc.012.net.il/>#012 >>> Jan 7 14:38:14 $CLIENT_IP.smc.012.net.il<http://client_ip.smc.012.net.il/>#012 >>> Jan 7 14:38:15 $CLIENT_IP.smc.012.net.il<http://client_ip.smc.012.net.il/>[0]<<$FreeSWITCH: 5060(334) >>> Jan 7 14:38:15 $CLIENT_IP.smc.012.net.il<http://client_ip.smc.012.net.il/>[0]<<$FreeSWITCH: 5060(334) >>> Jan 7 14:38:15 ACK sip: 1632@$CLIENT_IP:5060 SIP/2.0#012 >>> Jan 7 14:38:15 ACK sip: 1632@$CLIENT_IP:5060 SIP/2.0#012 >>> >>> >>> I saw one suggestion to do a factory reset that fixed the problem, but >>> that's a bit troublesome at the moment since the client can't >>> load provisioning files (dumb network.) >>> >>> -Avi Marcus >>> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > > > > ---------- Forwarded message ---------- > From: "Cal Leeming [Simplicity Media Ltd]" < > cal.leeming(a)simplicitymedialtd.co.uk> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Tue, 29 Jan 2013 20:13:54 +0000 > Subject: [Freeswitch-users] DID providers - any thoughts? (jan 2013) > Hello everyone, > > I've been recently reviewing several DID providers, and looking for some > feedback. > > I appreciate this isn't totally on topic for FreeSWITCH, but over the last > few months I've found the community feedback to be invaluable - so I'm > hoping it might open up a good discussion. > > I would be really interested to hear other peoples experiences with their > DID providers, as much detail as possible. > > There are several providers we've looked at, but they are either reselling > others services, or do not deliver the service quality we are looking for. > > The criteria I've been looking for is; > > * Able to provide either US or UK numbers > * Exposes an API that allows control of purchasing and routing > * Guarantees premium call quality, good latency, and inbound CLI. > * Does NOT operate on a market/exchange basis > * Trading as a telecoms provider for more than 5 years > * Has a reputable name within the community or industry > > So far, I have only found the following providers that meet this criteria; > > * Hello Telecom - UK numbers only, ITSPA registered, no charges for > minutes, rental or concurrent channels. No minimum charges. Excellent call > quality/latency, no CLI problems, and almost no downtime in the last 6 > months. However, they do not offer a sane purchasing API which makes > automation difficult, and interop with FS has been somewhat painful. > > * Simwood - UK numbers only, ITSPA registered. Slightly more expensive but > still very fair. Currently at inter-op stage, so can't comment further - > but they seem very good so far. > > * Voxbone - Most numbers, ITSPA registered. Extremely high minimum spend, > difficult to justify unless you already have existing traffic. > > * BT - UK numbers only (I think), ITSPA registered (of course lol).. Over > 9000 pages of inter-op and around £10k in set up fees. > > Thanks in advance > > Cal > > > ---------- Forwarded message ---------- > From: Victor Chukalovskiy <victor.chukalovskiy(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Tue, 29 Jan 2013 15:21:44 -0500 > Subject: [Freeswitch-users] Rejecting undesired codecs in bypass media mode > Hello, > > *Call scenario:* call comes in with undesired codec only in SDP. (e.g. > PCMA) > > In my sip profile I have: > inbound-bypass-media="true" > inbound-codec-negotiation="greedy" > inbound-codec-prefs="PCMU,G729" > > *What happens:* regardless of the codec in SDP, Freeswitch executes > dialplan and sends the call to the B-leg. > > It also happens regardless of wether I use "late" or "early" negotiation > > *What I would like to see:* > FS rejecting the call right away if SDP does not match > inbound-codec-prefs. Should not even hit the dial-plan. > > Any clue on how to achieve this? > > Many thanks, > -Victor > > > ---------- Forwarded message ---------- > From: Avi Marcus <avi(a)avimarcus.net> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Tue, 29 Jan 2013 22:55:11 +0200 > Subject: Re: [Freeswitch-users] DID providers - any thoughts? (jan 2013) > Shouldn't go without mentioning ukddi.com... > It's free, since it's subsidized by incoming traffic compensation. He > claims that's more profitable than termination service is, easily allowing > him to keep the business running. > I've not seen the API, I think it has to be specially activated. > > He's very responsive to emails. > > -Avi > > > On Tue, Jan 29, 2013 at 10:13 PM, Cal Leeming [Simplicity Media Ltd] < > cal.leeming(a)simplicitymedialtd.co.uk> wrote: > >> Hello everyone, >> >> I've been recently reviewing several DID providers, and looking for some >> feedback. >> >> I appreciate this isn't totally on topic for FreeSWITCH, but over the >> last few months I've found the community feedback to be invaluable - so I'm >> hoping it might open up a good discussion. >> >> I would be really interested to hear other peoples experiences with their >> DID providers, as much detail as possible. >> >> There are several providers we've looked at, but they are either >> reselling others services, or do not deliver the service quality we are >> looking for. >> >> The criteria I've been looking for is; >> >> * Able to provide either US or UK numbers >> * Exposes an API that allows control of purchasing and routing >> * Guarantees premium call quality, good latency, and inbound CLI. >> * Does NOT operate on a market/exchange basis >> * Trading as a telecoms provider for more than 5 years >> * Has a reputable name within the community or industry >> >> So far, I have only found the following providers that meet this criteria; >> >> * Hello Telecom - UK numbers only, ITSPA registered, no charges for >> minutes, rental or concurrent channels. No minimum charges. Excellent call >> quality/latency, no CLI problems, and almost no downtime in the last 6 >> months. However, they do not offer a sane purchasing API which makes >> automation difficult, and interop with FS has been somewhat painful. >> >> * Simwood - UK numbers only, ITSPA registered. Slightly more expensive >> but still very fair. Currently at inter-op stage, so can't comment further >> - but they seem very good so far. >> >> * Voxbone - Most numbers, ITSPA registered. Extremely high minimum spend, >> difficult to justify unless you already have existing traffic. >> >> * BT - UK numbers only (I think), ITSPA registered (of course lol).. Over >> 9000 pages of inter-op and around £10k in set up fees. >> >> Thanks in advance >> >> Cal >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > _______________________________________________ > 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 > > -- Luke Milbourne Email: luke.milbourne(a)gmail.com Tel: 07857154817

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