October 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

the wiki is down...
by Antonio 18 Oct '12

18 Oct '12

Hi, I can't access the wiki. http://wiki.freeswitch.org ... i was navigating... and one minute later no access... I have tried from another internet conection and the same problema occurs... no access... keeps loading... -- Un cordial saludo / Best regards, _________________________ António Silva E-mail:asilva@wirelessmundi.com

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Best way to pass CRM data from FSClient into Freeswitch?
by roland＠callrica.co.za 18 Oct '12

18 Oct '12

Hi Freeswitch Team, I'm looking for advice on how to pass CRM related information from a FSClient to a Freeswitch server and use that data to customise the voice recording file name and CDR The use case is like this: 1. User starts softphone and dials a contact from his contact list e.g. John Smith 2. Softphone passes CRM Data e.g. "John Smith" with call into Freeswitch 3. Freeswitch switches the call and records the conversation 4. Freeswitch labels the recording filename as "John Smith_SomeDateTime.wav" 5. Freeswitch tags the CDR with "John Smith" and call duration, start time etc... I would like all the voice recordings and CDRs to be stored on the FS Server. Is it possible for the FSClient to embed the CRM data in the SIP message, and the FS server extract it into variables for use in the dial plan? Very new to all this, so any pointers would be much appreciated. Rgds

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503 Response with ISUP content
by Luis Jimenez 17 Oct '12

17 Oct '12

Hello, devs, i have this situation. if FS sends an INVITE to a Nortel switch and the Nortel answer is SIP 503, FS does not send the ACK to the Nortel, any guidance please. Here is a trace of the Nortel answer. SIP/2.0 503 Service Unavailable. Via: SIP/2.0/UDP 67.220.147.141;received=67.220.147.141;branch=z9hG4bK5Q6g5UjNZeQZr;rport=5060. From: "9143345048" <sip:+9143345048@67.220.147.141>;tag=5maaU2Sa3gNHH. To: <sip:+59386510685@190.166.232.141;user=phone >;tag=f1b31366541302012101718172. Call-ID: 37e5bc92-934b-1230-4986-782bcb61be68. CSeq: 34920047 INVITE. Server: CS2000_NGSS/9.0. k: 100rel. Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK,UPDATE. Content-Disposition: signal;handling=optional. c: application/ISUP;version=. Content-Length: 6. ...... Thanks in advance for your help. Luis Jiménez

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Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 76, Issue 108
by Jeferson Rodrigo Almeida 17 Oct '12

17 Oct '12

Hi guys, I'm trying to use the fs_ivrd daemon, to execute a php script, that is called by the dialplan. It works like a charm, if not for the fact that the daemon (fs_ivrd) ends once the script (php) ends. In a system, I'll have to execute this script many times, and the daemon have to stay up. I think that it is wrong, right? I've already tryed to connect direct to a nc, simulating the deamon, and worked fine. I've already executed the script direct from de shell, and there's no sintax errors. *Here is my dialplan extension:* <extension name="outbound-socket"> <condition field="destination_number" expression="^55(522)$"> <action application="set" data="ivr_path=/usr/local/eox/programas/freeswitch/scripts/teste.php"/> <action application="socket" data="127.0.0.1:9099 async full"/> </condition> </extension> *Here is how I call the deamon:* (from the freeswitch bin dir) ./fs_ivrd -h 127.0.0.1 -p 9099 *Here is my PHP script:* #!/usr/local/eox/programas/php/bin/php -q <?php // set a couple of things so we dont kill the system ob_implicit_flush(true); set_time_limit(30); // Open stdin so we can read the data in $in = fopen("php://stdin", "r"); // Connect echo "connect\n\n"; // Answer echo "sendmsg\n"; echo "call-command: execute\n"; echo "execute-app-name: answer\n\n"; // Play a prompt echo "sendmsg\n"; echo "call-command: execute\n"; echo "execute-app-name: log\n"; echo "execute-app-arg: CRIT Testing text...\n\n"; // Wait sleep(2); // Hangup echo "sendmsg\n"; echo "call-command: hangup\n\n"; fclose($in); ?> I'll be grateful for any help. Thanks, and sorry for any error in my english. -- Jeferson Rodrigo Almeida Engenheiro de Computação jefersonparanaense(a)gmail.com

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Sangoma/FreeTDM FXS Loop Current Disconnect
by Brian Wiese 17 Oct '12

17 Oct '12

A stumper of a question... hopefully not for someone here though! I have a Sangoma A200 card with FXS modules and an emergency telephone (Rath Microtech SmartPhone II - http://www.rathmicrotech.com/smiviiise.html) that requires loop current disconnect signaling to indicate that the call has terminated. I tried everything that I could think of and been in touch with Sangoma's support but we cannot figure out what configuration option(s) I need to enable to send loop current disconnect. We tried setting answer-polarity-reverse and hangup-polarity-reverse to true in freetdm.conf.xml but that didn't seem to make any impact at all (based on the name I was guessing it wouldn't, but I'm willing to try anything!). Do you know what I'm missing? Any help/pointers/hints appreciated! Thanks. ~Brian

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Explicit MWI notify only works on first registration
by Tim St. Pierre 17 Oct '12

17 Oct '12

Hello, I'm trying to upgrade to 1.2.3 and I'm running into a few issues with MWI. We use the explicit-mwi-notify, because the voice mail account is usually different than the account the phone registers to. On the Aastra phones we have, subscribing to MWI always uses the same username and domain as the SIP subscription. If there are no registrations, I get a notify and correct status when the phone boots up and registers. If a new message is left or deleted, the update is very fast. Where I run into trouble is if a phone gets rebooted, or a second phone registers. It is fairly common on our system for a user to have two endpoints registered to the same username (which we allow). Is there a trick to making the MWI notify happen on EVERY registration? I have: <param name="forward-unsolicited-mwi-notify" value="false"/> <param name="send-message-query-on-register" value="true"/> in my internal profile. I'm using FreeSWITCH Version 1.2.3+git~20121004T033301Z~94664868a8 (1.2.3; git at commit 94664868a8 on Thu, 04 Oct 2012 03:33:01 Z) Running on FreeBSD 8.3-RELEASE AMD64 Thanks for any suggestions!

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Suspicious URL:RE: Re: SLA/BLF causing 482 loop detected errors on powerpc build
by KEN KELLY-TURNER 17 Oct '12

17 Oct '12

Resolved, the timestamp was what was giving me the trouble. Works flawlessly now From: KEN KELLY-TURNER Sent: October-17-12 1:05 PM To: 'FreeSWITCH Users Help' Subject: RE: Suspicious URL:Re: [Freeswitch-users] Suspicious URL:RE: Suspicious URL:Re: SLA/BLF causing 482 loop detected errors on powerpc build Having looked through it again, the only notable difference I can find between my two freeswitch builds is that my powerpc board is not synced to current time and instead defaults to linux's t=0 (1970-01-01) resulting in the timestamp on the freeswitch messages to have a time 40 years in the past. This may be a stupid question but could that have anything to do with it? Thanks, Ken From: freeswitch-users-bounces(a)lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: October-17-12 12:56 PM To: FreeSWITCH Users Help Subject: Suspicious URL:Re: [Freeswitch-users] Suspicious URL:RE: Suspicious URL:Re: SLA/BLF causing 482 loop detected errors on powerpc build Maybe its not the same? Look at the sip messages and the configs again? On Wed, Oct 17, 2012 at 9:30 AM, KEN KELLY-TURNER <KEN.KELLY-TURNER(a)adtran.com<mailto:KEN.KELLY-TURNER@adtran.com>> wrote: Why would I only be getting that warning on my PPC freeswitch with the same phones and configurations? Thanks, Ken From: freeswitch-users-bounces(a)lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org>] On Behalf Of Anthony Minessale Sent: October-16-12 11:49 AM To: FreeSWITCH Users Help Subject: Suspicious URL:Re: [Freeswitch-users] SLA/BLF causing 482 loop detected errors on powerpc build That is what it does when a line is supposed to be SLA and it does not contain the appearance-index in the call-info header. On Tue, Oct 16, 2012 at 10:20 AM, KEN KELLY-TURNER <KEN.KELLY-TURNER(a)adtran.com<mailto:KEN.KELLY-TURNER@adtran.com>> wrote: Hi Anthony, I upgraded to the git head, the problem persisted. I'm updating my phones to 4.0.3 now but I just noticed an additional warning that is produced on my ppc build when I attempt to make a call to my SLA line: [WARNING] sofia.c:8906 Auto-Fixing Broken SLA [sip:10.10.11.1;appearance-index=1] Any thoughts? And I am a co-op student for adtran, I'm just trying to determine functionality of freeswitch in a linux-powerpc environment. Thanks, Ken From: freeswitch-users-bounces(a)lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org>] On Behalf Of Anthony Minessale Sent: October-16-12 9:34 AM To: FreeSWITCH Users Help Subject: Suspicious URL:Re: [Freeswitch-users] SLA/BLF causing 482 loop detected errors on powerpc build When you are working on debugging stuff its best to use GIT HEAD and not frozen releases. Did you enable the logging on your phone to see what it may have to say? 4.0.x is still a buggy firmware on polycom so you might want to try the last 3.5 or 4.0.3, it may be a simple as a race condition based on the performance of the 2 boxes. We made some changes to accommodate such a situation in GIT HEAD last week. BTW, Couldn't help notice your domain name. Are you doing something on behalf of Adtran or you just happen to have an email addr there? On Mon, Oct 15, 2012 at 2:41 PM, KEN KELLY-TURNER <KEN.KELLY-TURNER(a)adtran.com<mailto:KEN.KELLY-TURNER@adtran.com>> wrote: Hi all, I've cross compiled freeswitch 1.2.3 for ppc and I'm trying to get SLA and BLF working on my polycom phones. I tested both features on a Ubuntu build of freeswitch 1.2.3 and they work perfectly. If I use the same configuration and phones on my powerpc version I get the following error: send 1243 bytes to udp/[10.10.8.81]:5060 at 00:28:17.493372: ------------------------------------------------------------------------ NOTIFY sip:1003@10.10.8.81<mailto:sip%3A1003@10.10.8.81> SIP/2.0 Via: SIP/2.0/UDP 10.10.11.1;rport;branch=z9hG4bKX2crS8D8HQtaB Max-Forwards: 70 From: <sip:1002@10.10.11.1<mailto:sip%3A1002@10.10.11.1>>;tag=0KH1Bj6SVsLd To: "free3" <sip:1003@10.10.11.1<mailto:sip%3A1003@10.10.11.1>>;tag=739B1DFC-3CEB89D3 Call-ID: 4a84c5e6-67b90e1d-2a96ef64(a)10.10.8.81<mailto:4a84c5e6-67b90e1d-2a96ef64@10.10.8.81> CSeq: 4052937957<tel:4052937957> NOTIFY Contact: <sip:1002@10.10.11.1:5060;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.2.3 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: presence Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=3572 Content-Type: application/xpidf+xml Content-Length: 430 <?xml version="1.0"?> <!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd"> <presence> <status> <note>Registered(UDP)</note> </status> <presentity uri="sip:1002@10.10.11.1<mailto:sip%3A1002@10.10.11.1>;method=SUBSCRIBE" /> <atom id="sip:1002@10.10.11.1<mailto:sip%3A1002@10.10.11.1>"> <address uri="sip:mod_sofia@10.10.11.1:5060;user=ip" priority="0.800000"> <status status="open" /> <msnsubstatus substatus="online" /> </address> </atom> </presence> ------------------------------------------------------------------------ recv 395 bytes from udp/[10.10.8.81]:5060 at 00:28:17.513912: ------------------------------------------------------------------------ SIP/2.0 482 Loop Detected Via: SIP/2.0/UDP 10.10.11.1;rport;branch=z9hG4bKX2crS8D8HQtaB From: <sip:1002@10.10.11.1<mailto:sip%3A1002@10.10.11.1>>;tag=0KH1Bj6SVsLd To: "free3" <sip:1003@10.10.11.1<mailto:sip%3A1003@10.10.11.1>>;tag=739B1DFC-3CEB89D3 CSeq: 2147483647<tel:2147483647> NOTIFY Call-ID: 4a84c5e6-67b90e1d-2a96ef64(a)10.10.8.81<mailto:4a84c5e6-67b90e1d-2a96ef64@10.10.8.81> Event: presence User-Agent: PolycomSoundPointIP-SPIP_550-UA/4.0.2.11307_0004f2149566 Accept-Language: en Content-Length: 0 ------------------------------------------------------------------------ Anybody else encounter this or have any suggestions on how to resolve? Thanks, Ken _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org<mailto:consulting@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<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com<mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net<http://irc.freenode.net> #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org<mailto:sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org<mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900<tel:%2B19193869900> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org<mailto:consulting@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<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com<mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net<http://irc.freenode.net> #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org<mailto:sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org<mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900<tel:%2B19193869900> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org<mailto:consulting@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<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com<mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net<http://irc.freenode.net> #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org<mailto:sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org<mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900

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Re: [Freeswitch-users] SLA/BLF causing 482 loop detected errors on powerpc build
by KEN KELLY-TURNER 17 Oct '12

17 Oct '12

Hi Anthony, I upgraded to the git head, the problem persisted. I'm updating my phones to 4.0.3 now but I just noticed an additional warning that is produced on my ppc build when I attempt to make a call to my SLA line: [WARNING] sofia.c:8906 Auto-Fixing Broken SLA [sip:10.10.11.1;appearance-index=1] Any thoughts? And I am a co-op student for adtran, I'm just trying to determine functionality of freeswitch in a linux-powerpc environment. Thanks, Ken From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: October-16-12 9:34 AM To: FreeSWITCH Users Help Subject: Suspicious URL:Re: [Freeswitch-users] SLA/BLF causing 482 loop detected errors on powerpc build When you are working on debugging stuff its best to use GIT HEAD and not frozen releases. Did you enable the logging on your phone to see what it may have to say? 4.0.x is still a buggy firmware on polycom so you might want to try the last 3.5 or 4.0.3, it may be a simple as a race condition based on the performance of the 2 boxes. We made some changes to accommodate such a situation in GIT HEAD last week. BTW, Couldn't help notice your domain name. Are you doing something on behalf of Adtran or you just happen to have an email addr there? On Mon, Oct 15, 2012 at 2:41 PM, KEN KELLY-TURNER <KEN.KELLY-TURNER(a)adtran.com<mailto:KEN.KELLY-TURNER@adtran.com>> wrote: Hi all, I've cross compiled freeswitch 1.2.3 for ppc and I'm trying to get SLA and BLF working on my polycom phones. I tested both features on a Ubuntu build of freeswitch 1.2.3 and they work perfectly. If I use the same configuration and phones on my powerpc version I get the following error: send 1243 bytes to udp/[10.10.8.81]:5060 at 00:28:17.493372: ------------------------------------------------------------------------ NOTIFY sip:1003@10.10.8.81<mailto:sip%3A1003@10.10.8.81> SIP/2.0 Via: SIP/2.0/UDP 10.10.11.1;rport;branch=z9hG4bKX2crS8D8HQtaB Max-Forwards: 70 From: <sip:1002@10.10.11.1<mailto:sip%3A1002@10.10.11.1>>;tag=0KH1Bj6SVsLd To: "free3" <sip:1003@10.10.11.1<mailto:sip%3A1003@10.10.11.1>>;tag=739B1DFC-3CEB89D3 Call-ID: 4a84c5e6-67b90e1d-2a96ef64(a)10.10.8.81<mailto:4a84c5e6-67b90e1d-2a96ef64@10.10.8.81> CSeq: 4052937957<tel:4052937957> NOTIFY Contact: <sip:1002@10.10.11.1:5060;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.2.3 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: presence Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=3572 Content-Type: application/xpidf+xml Content-Length: 430 <?xml version="1.0"?> <!DOCTYPE presence PUBLIC "-//IETF//DTD RFCxxxx XPIDF 1.0//EN" "xpidf.dtd"> <presence> <status> <note>Registered(UDP)</note> </status> <presentity uri="sip:1002@10.10.11.1<mailto:sip%3A1002@10.10.11.1>;method=SUBSCRIBE" /> <atom id="sip:1002@10.10.11.1<mailto:sip%3A1002@10.10.11.1>"> <address uri="sip:mod_sofia@10.10.11.1:5060;user=ip" priority="0.800000"> <status status="open" /> <msnsubstatus substatus="online" /> </address> </atom> </presence> ------------------------------------------------------------------------ recv 395 bytes from udp/[10.10.8.81]:5060 at 00:28:17.513912: ------------------------------------------------------------------------ SIP/2.0 482 Loop Detected Via: SIP/2.0/UDP 10.10.11.1;rport;branch=z9hG4bKX2crS8D8HQtaB From: <sip:1002@10.10.11.1<mailto:sip%3A1002@10.10.11.1>>;tag=0KH1Bj6SVsLd To: "free3" <sip:1003@10.10.11.1<mailto:sip%3A1003@10.10.11.1>>;tag=739B1DFC-3CEB89D3 CSeq: 2147483647<tel:2147483647> NOTIFY Call-ID: 4a84c5e6-67b90e1d-2a96ef64(a)10.10.8.81<mailto:4a84c5e6-67b90e1d-2a96ef64@10.10.8.81> Event: presence User-Agent: PolycomSoundPointIP-SPIP_550-UA/4.0.2.11307_0004f2149566 Accept-Language: en Content-Length: 0 ------------------------------------------------------------------------ Anybody else encounter this or have any suggestions on how to resolve? Thanks, Ken _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org<mailto:consulting@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<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com<mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net<http://irc.freenode.net> #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org<mailto:sip%3A888@conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org<mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900

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Help with Asterisk vs FreeSwitch keywords
by Mario G 17 Oct '12

17 Oct '12

My ITSO sent an asterisk setup but I cannot find an asterisk to freeSwitch cross ref for keywords. Can someone show the equivalent in FreeSwitch? FreeSwitch has realm, from-domain, etc. but I am having problems and want to make sure I code it right. Thanks Mario G srvlookup = yes <- Cannot find in FS domain = itspit.com <- realm? host = _SERVER_ <- realm or proxy? outboundproxy = _SERVER_ <- outbound-proxy register => 1234XXXXID:SUPERSECRET@_SERVER_ <- username/from-user

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FreeSWITCH Conference Call
by Michael Collins 17 Oct '12

17 Oct '12

Hello folks! Our weekly conference call is starting in about one hour. The agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2012_10_17 We look forward to hearing some information from two different Chads: Chad Philips and Chad Engler. The former will be giving us a brief heads up on migrating a project over to github and the latter will be discussing node-esl, a node.js + ESL library. 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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