February 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] external_sip_ip not refreshed after ip address change.
by Anurag Srivastava 08 Feb '12

08 Feb '12

I have a question about nat behavior in freeswitch. Basically my external calls to freeswitch are getting disconnected after 30 seconds of two way audio when my external ip address changes. I have dhcp from my ISP and am using external_sip_ip and external_rtp_ip as stun:<stun_server>. When my IP changes I see that external_sip_ip does not get refreshed but external_rtp_ip does. I am not allowed to enable upnp/nat-pmp on my router. Apparently it is a known issue that external_sip_ip is read just at load time and not refreshed even if it is specified in stun format Is there a fix to this problem? There is always the option of restarting profile when ddclient notes an ip change. Is there something inbuilt into FS. It is already finding that ip address has changed as reflected in external_rtp_ip which does use stun and gets the right ip address.

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Polycom BLF
by DJB International 08 Feb '12

08 Feb '12

Does anyone ever experience with BLF for DND on polycom expansion module? The BLF is working fine when the extension is busy, but when the extension has DND on, it did not change the LED light. Any sugestion? Thank you.

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Re: [Freeswitch-users] Fs ignoring voicemail to email
by Balamurugan Mahendran 08 Feb '12

08 Feb '12

I did, also I have no issue sending email outside FS. Thanks, Bala On Tue, Feb 7, 2012 at 8:31 PM, Balamurugan Mahendran < balamurugan(a)adaptavant.com> wrote: > I did, also I have no issue sending email outside FS. > > Thanks, > Bala > > On Tue, Feb 7, 2012 at 8:09 PM, John <freeswitch(a)earthspike.net> wrote: > >> I updated the wiki mod_voicemail page a few days ago with some >> instructions for debugging email from freeswitch. Have you tried those? >> >> >> On 07/02/12 13:47, Balamurugan Mahendran wrote: >> >> All, >> >> I am having the same issue, its not sending email(extension --> lua >> script) getting *"Segmentation fault"* >> >> xml : >> >> <include> >> <user id="503"> >> <params> >> <param name="password" value="1234"/> >> <param name="vm-password" value="503"/> >> <param name="vm-mailto" value="b2m(a)a-cti.com"/> >> <param name="vm-enabled" value="true"/> >> <param name="vm-skip-instructions" value="true"/> >> <param name="vm-disk-quota" value="3600"/> >> <param name="vm-email-all-messages" value="true"/> >> <param name="max-login-attempts" value="2"/> >> <param name="digit-timeout" value="10000"/> >> <param name="max-record-len" value="900"/> >> <param name="tone-spec" value="%(1000, 0, 640)"/> >> <param name="record-silence-threshold" value="2000"/> >> <param name="record-silence-hits" value="3"/> >> </params> >> <variables> >> <variable name="toll_allow" value="domestic,international,local"/> >> <variable name="accountcode" value="503"/> >> <variable name="user_context" value="default"/> >> <variable name="effective_caller_id_name" value="Extension 503"/> >> <variable name="effective_caller_id_number" value="503"/> >> <variable name="outbound_caller_id_name" >> value="$${outbound_caller_name}"/> >> <variable name="outbound_caller_id_number" >> value="$${outbound_caller_id}"/> >> <variable name="callgroup" value="techsupport"/> >> </variables> >> </user> >> </include> >> >> dialplan : >> >> <include> >> <extension name="503"> >> <condition field="destination_number" expression="^(503)$"> >> <action application="lua" data="vmbox.lua >> ${effective_caller_id_number}"/> >> </condition> >> </extension> >> </include> >> >> >> Lua : >> >> caller=503; >> freeswitch.consoleLog("info","From :"..caller); >> session:set_tts_parms("flite", "slt"); >> session:speak("Welcome To Voice Mail !. You Can Leave Your Message >> Here."); >> path="/usr/local/freeswitch/recordings/"; >> prompt=caller..".mp3"; >> recpath=path..prompt; >> freeswitch.consoleLog("info","record path="..recpath); >> session:recordFile(recpath,30,10,10); >> session:speak("Thank you."); >> >> freeswitch.consoleLog("info","testing"); >> freeswitch.email("b2m(a)a-cti.com", >> "saraswathi.devaraj(a)a-cti.com", >> "subject: Voicemail from 801\n", >> "Hello,\n\nYou've got a voicemail, click the attachment >> to listen to it.", >> "/usr/local/freeswitch/recordings/503.mp3", >> "", >> ""); >> freeswitch.consoleLog("info","hai"); >> >> >> >> switchconf : >> >> >> <param name="mailer-app" value="/usr/sbin/sendmail"/> >> <param name="mailer-app-args" value="-t"/> >> >> >> Thanks for your help!! >> >> Thanks, >> Bala >> >> >> >> On Sat, Feb 4, 2012 at 12:31 AM, John <freeswitch(a)earthspike.net> wrote: >> >>> On 03/02/12 18:00, Thomas Hoellriegel wrote: >>> > >>> > Its works fine!! >>> Good news! >>> >> >> >> _________________________________________________________________________ >> 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 >> >> >

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Can't get Paging to work on SPA962
by Mario G 08 Feb '12

08 Feb '12

I have not been able to get paging to work on SPA962s for a year now, I ran traces searched the web, etc. No luck. I found a post from 2009 that said it works but no description on how. Here is what happens: If I connect my old SPA9000 PBX up and power it on, then turn it off, FreeSwitch paging works! But, if a phone is powered off, FreeSwitch paging no longer works. Power on/off SPA9000 and all is well. Anyone out there have it working that could tell me the secret? I am using *11 to page to trigger the rtp_multicast_page extension. Thanks in advance! MarioG

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Help with attended transfer / att_xfer
by Brett Wilson 08 Feb '12

08 Feb '12

Sorry if I double posted this. I did not see the message in the digest so I am reposting. Hey guys, I need a pointer. I'm trying to get some kind of functionality where after a attended transfer is completed, ie. The ouside caller is connected to the 2nd phone after the 1st phone has hung up or chosen to continue the transfer. I would like a beep to sound for the 2nd phone or for both legs, just to let the recipient of the transfer know that it has gone through and they can start talking. Any ideas? Thanks Brett Wilson Launch 3 Ventures, LLC 134 Myer Street Hackensack, NJ 07601 Phone: 877.878.9134 Fax: 646.536.3866 Email: Brett.Wilson(a)launch3.net AOL IM: Brett.Wilson(a)launch3.net <blocked::http://www.launch3.net/> www.Launch3.net <http://www.launch3telecom.com/> www.Launch3telecom.com *************************** <http://launch3telecom.blogspot.com/> Description: Description: Blogger-logo <http://www.facebook.com/pages/Launch-3-Telecom/165610820178720> Description: Description: FaceBook-Logo <https://twitter.com/launch3telecom> Description: Description: Twitter-Logo

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I have 4 different ethernet ports with different ip's
by Josue Diaz Cruz 08 Feb '12

08 Feb '12

I have a server with 4 ETH. Each one have a different IP In 2 cases i fixed the ip address, In the other 2 cases i have 2 cable connections in bridge mode so i receive the public ip in dhcp mode. eth0 local address 192.168.1.100 eth1 local address 192.168.2.100 eth2 public address dhcp (cable) eth3 public address dhcp (cable) I was thinking to build 4 different profiles for each eth port, but i came to know that if we change the place of colocation it will not work properly because the dhcp ip. It is any way to attach a profile to the eth port, not the ip? Or if you know a better way to manage this case. I would like to receive traffic from the 2 public ip's and route it both like if i have just one profile. Josue Diaz Cruz Departamento Tecnico y Soporte <mailto:jdiaz@coinfru.com> jdiaz(a)coinfru.com C/ Balsicas 3 Alquerias | 30580 | Murcia <http://www.coinfru.com/> www.coinfru.com

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How to make calls from Browser?
by Balamurugan Mahendran 08 Feb '12

08 Feb '12

Hi all, I am trying to make calls from browser(chrome, safari, FireFox, Opera, IE) is there any known open source tool available already? Thanks, Bala

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Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 68, Issue 22
by Piyush Sharma 08 Feb '12

08 Feb '12

I am new to FreeSWITCH, I am telling you how I did it, In autoload config there is a file cdr_csv.conf.xml there are few queries written, I made my own according to my cdr requirement, After that, I wrote a function to insert into sql in mod_cdr_csv.c and called it from my_on_reporting (written function in mod_cdr_cdv.c), and it made my cdr to mysql. On Mon, 2012-02-06 at 01:05 +0300, 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: sip_gateway dialplan variable (Hynek Cihlar) > 2. Re: vm_delete (Darcy) > 3. Server with 2 NICS (Luis Daniel Lucio Quiroz) > 4. how to store cdr data to sql dB (Sherif Omran) > 5. Re: how to store cdr data to sql dB (Avi Marcus) > MHTML Document attachment > > -------- Forwarded Message -------- > > From: Hynek Cihlar <hynek.cihlar(a)gmail.com> > > Reply-To: FreeSWITCH Users Help > > <freeswitch-users(a)lists.freeswitch.org> > > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > > Subject: Re: [Freeswitch-users] sip_gateway dialplan variable > > Date: Sun, 5 Feb 2012 19:41:26 +0100 > > > > Yes! Thanks for the tip! > > > > Hynek > > > > > > > > On Sun, Feb 5, 2012 at 11:12 AM, Avi Marcus <avi(a)avimarcus.net> > > wrote: > > At worst, you can create a variable in the gateway settings > > so it populates a channel variable. I couldn't find the > > syntax on the wiki, though.. should make that easy to find. > > We need a real gateway page.. > > > > -Avi > > > > > > On Wed, Feb 1, 2012 at 9:29 AM, Hynek Cihlar > > <hynek.cihlar(a)gmail.com> wrote: > > Hello Mike and thanks for the reply. > > > > > > Unfortunately sip_req_host only gives me the sip > > host. My setup is probably not quite usual, I have a > > multihome system with many gateways, each gateway > > name assigned to a unique id. The unique id is > > simply a random string identifying the gateway in > > the whole system. The unique id is also persisted in > > the DB, based on the id additional call processing > > takes place. My initial idea was to get the gateway > > name in the dialplan and pass it to an ESL app. But > > it really looks like it is not possible to retrieve > > the gateway name at all, sadly. > > > > Hynek > > > > > > > > On Tue, Jan 24, 2012 at 8:42 AM, Michael Lutz > > <mytemike72(a)gmail.com> wrote: > > Hi Hynek, > > > > > > {sip_req_host} should give your own gateway > > (local), and {network_addr} should give the > > remote gateway. > > > > > > Otherwise writing a cdr and look will help > > you with this. > > > > > > Regards, > > Mike > > > > Op 24 jan. 2012 om 08:27 heeft Hynek Cihlar > > <hynek.cihlar(a)gmail.com> het volgende > > geschreven: > > > > > > > > > In one of the examples on wiki I found > > > ${sip_gateway}, but that doesn't seem to > > > work. Also, there is ${sip_gateway_name}, > > > but that works on outbound calls only. > > > > > > Hynek > > > > > > > > > > > > On Mon, Jan 23, 2012 at 7:03 PM, Hynek > > > Cihlar <hynek.cihlar(a)gmail.com> wrote: > > > Dear, is there a dialplan variable > > > that would hold the gateway name > > > an incoming call came through? > > > Searching through the source code > > > and wiki didn't yield any positive > > > result. > > > > > > > > > Thanks! > > > > > > > > > Hynek > > > > > > > > > > > > _________________________________________________________________________ > > > 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 > > > > > > > _________________________________________________________________________ > > 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 > > > > > > > > _________________________________________________________________________ > > 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 > > > > > > > > > > _________________________________________________________________________ > > 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 > > > > > > > MHTML Document attachment > > -------- Forwarded Message -------- > > From: Darcy <darcy(a)primrose.ws> > > Reply-To: FreeSWITCH Users Help > > <freeswitch-users(a)lists.freeswitch.org> > > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > > Subject: Re: [Freeswitch-users] vm_delete > > Date: Sun, 5 Feb 2012 14:47:32 -0500 > > > > I did a work around, because I am running this from a cgi, after I issue > > the command to the fs to delete the email, I actually delete the file with > > the cgi, not the best way, but it works until I find a seemless fix from fs. > > > > Darcy > > > > -----Original Message----- > > From: cogs66 > > Sent: Sunday, February 05, 2012 1:21 PM > > To: freeswitch-users(a)lists.freeswitch.org > > Subject: Re: [Freeswitch-users] vm_delete > > > > Hi > > > > I am experiencing the same but not been able to find a resolve to the issue > > yet. > > > > Thanks > > > > Andy > > > > -- > > View this message in context: > > http://freeswitch-users.2379917.n2.nabble.com/vm-delete-tp7255354p7256341.h… > > Sent from the freeswitch-users mailing list archive at Nabble.com. > > > > _________________________________________________________________________ > > 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 > > > > > > > > > MHTML Document attachment > > -------- Forwarded Message -------- > > From: Luis Daniel Lucio Quiroz <luis.daniel.lucio(a)gmail.com> > > Reply-To: FreeSWITCH Users Help > > <freeswitch-users(a)lists.freeswitch.org> > > To: freeswitch-users(a)lists.freeswitch.org > > Subject: [Freeswitch-users] Server with 2 NICS > > Date: Sun, 5 Feb 2012 09:07:01 -0600 > > > > Helo :) > > > > I'm new to FS but not to linux. I've sucessfully install FS into a > > centos 6.2. But I have an issue. > > > > Server has 2 nics: > > venet0: 10.9.19.5/16 > > eth0: <public IP> > > > > when starting fs, it listen SIP (5060/udp) using venet0. Phones can > > register and they talk each other. However, we are using an external > > GW and that fails. Doing a depth analisys I've realized that the > > reason is that because venet0 is being used, SIP payload has venet0 > > within but server IP is eth0's so this is tipical mitchach. > > > > I've sucessfully configure in vars.xml the local_ip4 (or whatever it > > writes) to use public IP. but after that no phones are registering. > > (before anyone says, yes phones points to public IP of course). Nmap > > reports that SIP (5060/udp) is listen by eth0 but when I got phone > > SIP, i only get Registering request without an answer. > > > > I really need to make this listen in eth0. What suggestion do you > > have. Please void telling me to turn of venet0 (not possible for > > politics). > > > > TIA > > > > > MHTML Document attachment > > -------- Forwarded Message -------- > > From: Sherif Omran <sherifomran2000(a)yahoo.com> > > Reply-To: FreeSWITCH Users Help > > <freeswitch-users(a)lists.freeswitch.org> > > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > > Subject: [Freeswitch-users] how to store cdr data to sql dB > > Date: Sun, 5 Feb 2012 14:00:10 -0800 (PST) > > > > Hi guys, > > > > any body knows how to store cdr data into mysql dB through xml-curl? > > > > could you please help > > > > thank you > > kind regards > > > > Sherif > > > > > MHTML Document attachment > > -------- Forwarded Message -------- > > From: Avi Marcus <avi(a)avimarcus.net> > > Reply-To: FreeSWITCH Users Help > > <freeswitch-users(a)lists.freeswitch.org> > > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > > Subject: Re: [Freeswitch-users] how to store cdr data to sql dB > > Date: Mon, 6 Feb 2012 00:05:18 +0200 > > > > You must > > mean http://wiki.freeswitch.org/wiki/Mod_xml_cdr or http://wiki.freeswitch.org/wiki/Xmlcdrd > > xml_cdr "merely" posts the CDR data in XML form to a web page. You > > can then have you python/perl/ruby/php/node.js/lua script pick out > > the columns you want and save it to the mysql db. > > If you want a php sample, fusionpbx has the open source code to save > > to mysql, sqlite, or pg, via PDO. > > > > > > -Avi > > > > > > On Mon, Feb 6, 2012 at 12:00 AM, Sherif Omran > > <sherifomran2000(a)yahoo.com> wrote: > > Hi guys, > > > > any body knows how to store cdr data into mysql dB through > > xml-curl? > > > > could you please help > > > > thank you > > kind regards > > > > Sherif > > > > > > > > _________________________________________________________________________ > > 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 -- Piyush Sharma <piyush.sharma(a)coraltele.com>

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Unexpected BYE's after 100 calls
by Boudewyn Ligthart 08 Feb '12

08 Feb '12

Hi, We have been testing with Freeswitch (hacked-20120106T151802Z). There are two profiles, allowing calls through using ACL, the core is running with ODBC on MySQL, all running on ubuntu, virtualized in XEN. When testing with SIPP and standard uac towards Freeswitch there are no issue's with the SIP messaging, SIPP on another machine acts as uas. So call flows SIPP --> Freeswitch --> SIPP. When testing with SIPP (using same routing) streaming audio after about 100 calls (5 seconds in) there are unexpected BYE's coming from Freeswitch (I have even lowered the CPS count). I do not see any error's in the logs, just the BYE's. What could be causing this issue? Thanks, Bou

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error on outgoing calls
by Charles Bujold 08 Feb '12

08 Feb '12

Freeswitch was working properly and today our outgoing calls started to get rejected. Here is the log of an outgoing call. Can somebody point to the error and how to fix it. Thanks CJB 2012-02-07 19:33:13.052955 [DEBUG] switch_core_session.c:875 Send signal sofia/external/150XXXXXXX [BREAK] 2012-02-07 19:33:13.052955 [DEBUG] switch_core_session.c:875 Send signal sofia/external/150XXXXXXX [BREAK] 2012-02-07 19:33:13.052955 [DEBUG] sofia.c:5508 Channel sofia/external/1506XXXXXXX entering state [terminated][403] 2012-02-07 19:33:13.052955 [DEBUG] switch_channel.c:2852 (sofia/external/1506XXXXXX) Callstate Change RINGING -> HANGUP 2012-02-07 19:33:13.052955 [NOTICE] sofia.c:6272 Hangup sofia/external/1506XXXXXXX [CS_CONSUME_MEDIA] [CALL_REJECTED] 2012-02-07 19:33:13.052955 [DEBUG] switch_ivr_originate.c:3364 Originate Resulted in Error Cause: 21 [CALL_REJECTED] 2012-02-07 19:33:13.052955 [DEBUG] switch_channel.c:2875 Send signal sofia/external/150XXXXXXXX [KILL] 2012-02-07 19:33:13.052955 [INFO] mod_dptools.c:2916 Originate Failed. Cause: CALL_REJECTED

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