- FreeSWITCH-users - FreeSWITCH Mailing Lists

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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FreeSWITCH Community Conference Call Today
by Michael Collins 30 Jan '13

30 Jan '13

Hello folks, Our conference call starts in less than an hour. The agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2013_01_30 Today we are going to do a refresher on event socket and ESL, specifically on the commands you can do when controlling a phone call. We are finding more and more people are using event socket and ESL and some of the same questions are coming up so we thought it would be a good idea to review a few things when it comes to using the event socket. Talk to you soon! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

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Patton FXO configuration on Freeswitch - more details
by Alessio 30 Jan '13

30 Jan '13

Hi all! We're trying to configure a Patton FXO in our Freeswitch. We have an internal PBX - Freeswitch 1.3.6 installed from svn master with (demo) setup on IP 10.0.1.190 - Patton 4112 with simple config on IP 10.0.1.10 Find more info below. So any help would be great! We set it up that the Patton registers with the PBX and we get the error: --------------------- 2013-01-25 16:44:39.433873 [WARNING] sofia_reg.c:2469 Can't find user [fxo0_3_0(a)10.0.1.190] You must define a domain called '10.0.1.190' in your directory and add a user with the id="fxo0_3_0" attribute and you must configure your device to use the proper domain in it's authentication credentials. --------------------- We defined a gateway in conf/sip_profiles/external/fxo.xml: --------------------- <include> <gateway name="10.0.1.190">--> <param name="username" value="fxo0_3_0"/> <param name="password" value="fxo0_3_0"/> <param name="register" value="false"/> </gateway> </include> --------------------- And the relevant parts of the Patton config: --------------------- # define auth authentication-service authenticate username fxo0_3_0 password gtatFAePDQFL0btl3cbsaA== encrypted # patton registers location-service LOCATION_FXO_3_0 domain 1 10.0.1.190 5060 identity-group group_fxo0_3_0 authentication outbound authenticate 1 authentication-service authenticate username fxo0_3_0 identity fxo0_3_0 authentication outbound authenticate 1 authentication-service authenticate username fxo0_3_0 registration outbound registrar 10.0.1.190 5060 lifetime 180 register auto retry-timeout on-system-error 10 retry-timeout on-client-error 10 retry-timeout on-server-error 10 call outbound use profile tone-set IT use profile voip default use profile sip default preferred-transport-protocol tcp traffic-class local-default invite-transaction-timeout 32 non-invite-transaction-timeout 32 penalty-box call inbound use profile tone-set IT use profile voip default use profile sip default context sip-gateway ASISTAR_3_0 interface fxo0_3_0 bind interface eth0 context router port 5060 context sip-gateway ASISTAR_3_0 bind location-service LOCATION_FXO_3_0 no shutdown --------------------- Output from 'sofia status' --------------------- external::10.0.1.10 gateway sip:fxo0_3_0@10.0.1.10 NOREG Thanks, Alessio

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FSClient question - messaging
by Dominic Raferd 30 Jan '13

30 Jan '13

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Re: [Freeswitch-users] DID providers - any thoughts? (jan 2013)
by Avi Marcus 30 Jan '13

30 Jan '13

Didww has great coverage. I think that's the only positive thing I can say about them. I yanked my numbers from them. -Avi (This message was painstakingly thumbed out on my mobile, so apologies for brevity and errors.) On Jan 30, 2013 3:14 AM, "Ashley Breeden" <ash(a)url.net.au> wrote: I find DIDww not too bad for ad-hoc DID's all over the world. Not sure if it meets your exact criteria. Check out www.didww.com Ash. On 30/01/2013, at 12:00 PM, Patrick Lists <freeswitch-list(a)puzzled.xs4all.nl> wrote: > On 01/30/20...

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voices in the recordings are out of sync
by Yungwei Chen 30 Jan '13

30 Jan '13

Hi, I found one issue that voices are always out of sync in the recordings. I am running freeswitch-1.2.5.1-1 on CentOS 5 (i386), which was installed from yum. I am having trouble installing the latest version from source due to an error: Autoconf version 2.62 or higher is required. It would be nice if someone can reproduce this issue against HEAD. Thanks. Here're the steps to reproduce it. The idea is to call a phone number and then bridge to another phone number while the entire session is being recorded. 1. In dialplan/public.xml, make sure you have a dialplan to handle any 10 digit phone numbers. <extension name="public_extensions"> <condition field="destination_number" expression="^\d{10}$"> <action application="transfer" data="main XML default"/> </condition> </extension> 2. In dialplan/default/main.xml, make sure you have an extension to handle the call in the default context. <extension name="test"> <condition field="${destination_number}" expression="^main$"> <action application="set" data="RECORD_STEREO=false"/> <action application="answer" /> <action application="record_session" data="/tmp/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/> <action application="bridge" data="{ignore_early_media=false}[leg_timeout=60]sofia/gateway/gw1/1234567890"/> </condition> </extension> 3. In sip_profiles/external/gateways.xml, make sure you have a gateway that allows you to make an outbound call. <include> <gateway name="gw1"> <param name="username" value=""/> <param name="password" value=""/> <param name="realm" value=""/> <param name="from-domain" value=""/> <param name="extension" value=""/> <param name="expire-seconds" value="60"/> <param name="register" value="false"/> <param name="retry-seconds" value="60"/> </gateway> </include> 4. make a call to one of the allowed 10-digit phone numbers in your environment. 5. Once the call is answered, the caller shall start to count from 1 to 60 with some pause after each number. 6. The callee shall repeat each number he/she heard from the caller. 7. You should be able to hear that 2 voices in the recoridng (/tmp/rec.wav) are out of sync.

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SNOM Phones - BEARERCAPABILITY_NOTAUTH error
by Sirish Masur Mohan 30 Jan '13

30 Jan '13

Hello All, I think I am missing something obvious here, but I am having trouble trying to make calls between two SNOM phones. I have an open ACL for registration and I keep getting BEARERCAPABILITY_NOTAUTH error when I try to make calls. Would appreciate any inputs on how to debug this! Logs are at : http://pastebin.freeswitch.org/20504 With regards, Sirish

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