December 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

freeswitch repo
by Hristo Benev 10 Dec '12

10 Dec '12

When updating I have following error: Package freeswitch-application-httapi-1.2.5.1-1.el6.i386.rpm is not signed Is that normal? Thanks

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Avaya SIP Phones?
by Sean Devoy 10 Dec '12

10 Dec '12

Hi, We have a potential customer asking us to give them a bid with AVAYA phones!!! Does AVAYA have any true SIP phones? Do they work with FreeSwitch? Thanks, Sean

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FreeTDM ftmod pri tap error
by Gopalakrishnan N 10 Dec '12

10 Dec '12

Hi, Am trying to install ftmod pri tap by following this link http://wiki.freeswitch.org/wiki/FreeTDM#Tapping and enabled --with-pritap while doing ./configure, and while doing make I get error. Can someone help me on this. Do I need to update any patch for tap? My installation log is attached here. Regards. Gopal.

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recovery_profile_name - any explanation?
by Cal Leeming [Simplicity Media Ltd] 09 Dec '12

09 Dec '12

Hello, I've just been trawling Google for an explanation of what exactly the variable '*recovery_profile_name' means, but no luck.* * * *It seems to relate to the following change from 4 months ago;* https://github.com/freeSwitch/freeswitch/commit/2a8841ab666ec23ceb5688587bb… The commit message states 'change mod_sofia to use new core based recovery engine'. But, it is unclear when this variable should be relied upon, and under what circumstances it will be used under. Any clarifications would be good, I will of course put in time to update the wiki too. Thanks Cal

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Different dtmf type for different gateways in single profile
by Muhammad Naseer Bhatti 09 Dec '12

09 Dec '12

Is there a way to set different dtmf type for different gateways in same profile? I am originating from esl, call is bridged with audio file. (Dialer setup). Some of the providers send dtmf inband and others rfc2833. I am currently checking in my lua script for a custom param, if present then execute start_dtmf to listen for inband dtmf. While this breaks the rest of dtmf. Is there a better way to deal with this or do I have to end up with different profiles each with different dtmf-type. -- Sent via a mobile device.

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Re: [Freeswitch-users] xml_curl result not found
by Jeff Geddes 08 Dec '12

08 Dec '12

thanks guys Avi, your suggestion makes perfect sense, that's what I'll do jeff On Sat, Dec 8, 2012 at 6:19 PM, < 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. Problem freeswitch and asterisk mediagateway bridge no direct > connection (Thomas Hoellriegel) > 2. Re: Problem freeswitch and asterisk mediagateway bridge no > direct connection (covici(a)ccs.covici.com) > 3. xml_curl result not found (Jeff Geddes) > 4. Re: xml_curl result not found (Avi Marcus) > 5. Re: xml_curl result not found (Cal Leeming [Simplicity Media Ltd]) > 6. Re: IVR menu delay - 1 second between events > (Cal Leeming [Simplicity Media Ltd]) > 7. recovery_profile_name - any explanation? > (Cal Leeming [Simplicity Media Ltd]) > > > ---------- Forwarded message ---------- > From: Thomas Hoellriegel <admin(a)blindi.net> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Sat, 8 Dec 2012 16:25:31 +0100 (CET) > Subject: [Freeswitch-users] Problem freeswitch and asterisk mediagateway > bridge no direct connection > Hi, > i have setup a voicechat (social network for handycap users). > Fs and asterisk works on the same host: > eth0 mainip fs > eth0:0 asteriskip > I connecting freeswitch (siprouter line managemnt), asterisk > (mediagateway), > then i connect from a sip provider > Fs answer the call, and forward to bridge to asterisk fine. then i make a > direct connection to fs: (no sipprovider) > via sipuri: "dorf(a)dorf.blindi.net" > fs answer the calls, but the bridge don.t works to asterisk. > the bridge hangup before asterisk answer the call from fs > > the problem is only to connect fs from the world. Why works only fine from > a sip provider? > must i edit the internal profile to bridge calls to another instance? > > Can you help please? thanks. > > > > i connection > > --------------- > Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: > http://www.blindi.net/callback > homepage: http://www.blindi.net > blinde-misc mailingliste für blinde. anmeldung unter: > http://www.blindi.net/mailman/**listinfo/blinde-misc<http://www.blindi.net/mailman/listinfo/blinde-misc> > > > ---------- Forwarded message ---------- > From: covici(a)ccs.covici.com > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Sat, 08 Dec 2012 12:02:11 -0500 > Subject: Re: [Freeswitch-users] Problem freeswitch and asterisk > mediagateway bridge no direct connection > Why use asterisk at all? What is it doing? > > Thomas Hoellriegel <admin(a)blindi.net> wrote: > > > Hi, > > i have setup a voicechat (social network for handycap users). > > Fs and asterisk works on the same host: > > eth0 mainip fs > > eth0:0 asteriskip > > I connecting freeswitch (siprouter line managemnt), asterisk > > (mediagateway), > > then i connect from a sip provider > > Fs answer the call, and forward to bridge to asterisk fine. then i > > make a direct connection to fs: (no sipprovider) > > via sipuri: "dorf(a)dorf.blindi.net" > > fs answer the calls, but the bridge don.t works to asterisk. > > the bridge hangup before asterisk answer the call from fs > > > > the problem is only to connect fs from the world. Why works only fine > > from a sip provider? > > must i edit the internal profile to bridge calls to another instance? > > > > Can you help please? thanks. > > > > > > > > i connection > > > > --------------- > > Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: > > http://www.blindi.net/callback > > homepage: http://www.blindi.net > > blinde-misc mailingliste für blinde. anmeldung unter: > > http://www.blindi.net/mailman/listinfo/blinde-misc > > _________________________________________________________________________ > > 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 > > -- > Your life is like a penny. You're going to lose it. The question is: > How do > you spend it? > > John Covici > covici(a)ccs.covici.com > > > > > ---------- Forwarded message ---------- > From: Jeff Geddes <geddes.jeffrey(a)gmail.com> > To: freeswitch-users(a)lists.freeswitch.org > Cc: > Date: Sat, 8 Dec 2012 13:32:45 -0400 > Subject: [Freeswitch-users] xml_curl result not found > I'm using xml_curl to generate a dialplan and send back result status="not > found" if something throws an error. I don't want it to fail back to the > in-memory dialplan if there is an error. Is there a way to do this? > > thanks > jeff > > > ---------- Forwarded message ---------- > From: Avi Marcus <avi(a)avimarcus.net> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Sat, 8 Dec 2012 20:10:29 +0200 > Subject: Re: [Freeswitch-users] xml_curl result not found > Why send a not found? Send a hangup with a failure reason, and that will > be the end of processing. > > -Avi > > > On Sat, Dec 8, 2012 at 7:32 PM, Jeff Geddes <geddes.jeffrey(a)gmail.com>wrote: > >> I'm using xml_curl to generate a dialplan and send back result >> status="not found" if something throws an error. I don't want it to fail >> back to the in-memory dialplan if there is an error. Is there a way to do >> this? >> >> thanks >> jeff >> >> _________________________________________________________________________ >> 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: Sat, 8 Dec 2012 20:48:33 +0000 > Subject: Re: [Freeswitch-users] xml_curl result not found > Exclude any dialplan sections from your config, and thus it won't have > anything to fall back onto. > > However, if any other modules are configured to look for configs (??) then > you may need to use the approach Avi mentioned, i.e. sending back a hangup > with failure reason. > > Cal > > On Sat, Dec 8, 2012 at 5:32 PM, Jeff Geddes <geddes.jeffrey(a)gmail.com>wrote: > >> I'm using xml_curl to generate a dialplan and send back result >> status="not found" if something throws an error. I don't want it to fail >> back to the in-memory dialplan if there is an error. Is there a way to do >> this? >> >> thanks >> jeff >> >> _________________________________________________________________________ >> 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: Sat, 8 Dec 2012 21:09:54 +0000 > Subject: Re: [Freeswitch-users] IVR menu delay - 1 second between events > I've raised a JIRA ticket for this; > http://jira.freeswitch.org/browse/FS-4924 > > Cal > > On Fri, Dec 7, 2012 at 8:08 PM, Cal Leeming [Simplicity Media Ltd] < > cal.leeming(a)simplicitymedialtd.co.uk> wrote: > >> Just trawled through the source and found this; >> >> src/switch_ivr_menu.c: >> if (status == SWITCH_STATUS_SUCCESS) { >> status = switch_ivr_sleep(session, 1000, SWITCH_FALSE, NULL); >> } >> >> So, I changed 1000 to 10, the delay has now disappeared and it seems to >> work perfectly. >> > > > > ---------- 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: Sat, 8 Dec 2012 22:18:37 +0000 > Subject: [Freeswitch-users] recovery_profile_name - any explanation? > Hello, > > I've just been trawling Google for an explanation of what exactly the > variable '*recovery_profile_name' means, but no luck.* > * > * > *It seems to relate to the following change from 4 months ago;* > > https://github.com/freeSwitch/freeswitch/commit/2a8841ab666ec23ceb5688587bb… > > The commit message states 'change mod_sofia to use new core based > recovery engine'. > > But, it is unclear when this variable should be relied upon, and under > what circumstances it will be used under. > > Any clarifications would be good, I will of course put in time to update > the wiki too. > > Thanks > > Cal > > _______________________________________________ > 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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xml_curl result not found
by Jeff Geddes 08 Dec '12

08 Dec '12

I'm using xml_curl to generate a dialplan and send back result status="not found" if something throws an error. I don't want it to fail back to the in-memory dialplan if there is an error. Is there a way to do this? thanks jeff

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having problem using java with esl library
by bawajan 08 Dec '12

08 Dec '12

I am new to freeswitch and am using java app with esl library to make outbound calls and am getting the error is "Create additional event dispatch thread 2". After that in the freeswitch logs its printing - Event 0 Blocking and its keeps printing. 2012-12-05 13:46:02.179956 [WARNING] switch_event.c:607 Create additional event dispatch thread 2 2012-12-05 13:46:19.919951 [WARNING] switch_event.c:607 Create additional event dispatch thread 3 2012-12-05 13:46:46.159954 [WARNING] switch_event.c:607 Create additional event dispatch thread 4 2012-12-05 13:47:03.679956 [WARNING] switch_event.c:607 Create additional event dispatch thread 5 2012-12-05 13:47:24.999950 [WARNING] switch_event.c:607 Create additional event dispatch thread 6 2012-12-05 13:47:40.299950 [WARNING] switch_event.c:607 Create additional event dispatch thread 7 2012-12-05 13:47:54.879950 [WARNING] switch_event.c:348 Event Thread 0 is blocking 2012-12-05 13:47:55.079953 [WARNING] switch_event.c:348 Event Thread 0 is blocking 2012-12-05 13:47:55.139957 [WARNING] switch_event.c:348 Event Thread 0 is blocking 2012-12-05 13:47:55.639950 [WARNING] switch_event.c:348 Event Thread 0 is blocking 2012-12-05 13:47:55.819960 [WARNING] switch_event.c:348 Event Thread 0 is blocking 2012-12-05 13:47:55.859950 [WARNING] switch_event.c:348 Event Thread 0 is blocking 2012-12-05 13:47:55.939953 [WARNING] switch_event.c:348 Event Thread 0 is blocking Kindly let me know where am doing mistake and how to resolve it. Thanks in Advance, Bawajan -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/having-problem-using-java-wit… Sent from the freeswitch-users mailing list archive at Nabble.com.

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POLYCOM 335 registration w/ Freeswitch
by Sean Devoy 07 Dec '12

07 Dec '12

I am new to Polycom phones. We have always used CISCO 5xx Series. I have been trying to use the WEB interface. The example in the WIKI uses tftp through DHCP option 60. MY home router/gateway's DHCP does not support option 60. I am unable to get the Polycom 335 to register, it keeps getting "Unauthorized". I have double checked by having my Cisco phone register with this information and it works fine. I must be missing something. On the Polycom Web Interface I have the following values: SIP TAB: SERVERS: Outbound Proxy is blank SERVER 1: Address: <my server's FQDN > Port: 5060 Transport: UDPOnly Expires:3600 Register: 1 Retry Timeout: 0 Retry Max Count: 3 Line Seize Timeout: 30 SERVER 2: <same as server 1 for now except port 5062) LINE TAB: LINE 1: Display Name: 228 Address: 228 Auth UserID:228 Auth Password: <actual password here> Label: 228 Type: Private 3rd Party Name: blank Number of Line Keys: 1 Calls per line: 24 SERVER 1: Address: <my server's FQDN > Port: 5060 Transport: UDPOnly Expires:3600 Register: 1 Retry Timeout: 0 Retry Max Count: 3 Line Seize Timeout: 30 SERVER 2: <same as server 1 for now) Everything else is the defaults. The resulting sofia trace messages: ------------------------------------------------------------------------ recv 546 bytes from udp/[71.127.152.57]:1025 at 22:11:38.210138: ------------------------------------------------------------------------ REGISTER sip:fs_bfis.bizfocused.com:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.40.47;branch=z9hG4bKa1ca8525B629017C From: "228" <sip:228@fs_bfis.bizfocused.com>;tag=DDAB2D48-F3771D9B To: <sip:228@fs_bfis.bizfocused.com> CSeq: 1 REGISTER Call-ID: ca6990df-b25553e-44d1bb49(a)10.10.40.47 Contact: <sip:228@10.10.40.47>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_335-UA/3.3.3.0069 Accept-Language: en Max-Forwards: 70 Expires: 3600 Content-Length: 0 ------------------------------------------------------------------------ send 703 bytes to udp/[71.127.152.57]:5060 at 22:11:38.211063: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.10.40.47;branch=z9hG4bKa1ca8525B629017C;received=71.127.152.57 From: "228" <sip:228@fs_bfis.bizfocused.com>;tag=DDAB2D48-F3771D9B To: <sip:228@fs_bfis.bizfocused.com>;tag=37Hyv1SjceDre Call-ID: ca6990df-b25553e-44d1bb49(a)10.10.40.47 CSeq: 1 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120712T080314Z~435f28cefb+unclean~20120 712T101002Z Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: precondition, path, replaces WWW-Authenticate: Digest realm="fs_bfis.bizfocused.com", nonce="f9a005f6-d33c-425e-9264-525c669833ef", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ recv 546 bytes from udp/[71.127.152.57]:1025 at 22:11:38.712339: ------------------------------------------------------------------------ REGISTER sip:fs_bfis.bizfocused.com:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.40.47;branch=z9hG4bKa1ca8525B629017C From: "228" <sip:228@fs_bfis.bizfocused.com>;tag=DDAB2D48-F3771D9B To: <sip:228@fs_bfis.bizfocused.com> CSeq: 1 REGISTER Call-ID: ca6990df-b25553e-44d1bb49(a)10.10.40.47 Contact: <sip:228@10.10.40.47>;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER" User-Agent: PolycomSoundPointIP-SPIP_335-UA/3.3.3.0069 Accept-Language: en Max-Forwards: 70 Expires: 3600 Content-Length: 0 ------------------------------------------------------------------------ send 703 bytes to udp/[71.127.152.57]:5060 at 22:11:38.712479: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.10.40.47;branch=z9hG4bKa1ca8525B629017C;received=71.127.152.57 From: "228" <sip:228@fs_bfis.bizfocused.com>;tag=DDAB2D48-F3771D9B To: <sip:228@fs_bfis.bizfocused.com>;tag=37Hyv1SjceDre Call-ID: ca6990df-b25553e-44d1bb49(a)10.10.40.47 CSeq: 1 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120712T080314Z~435f28cefb+unclean~20120 712T101002Z Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: precondition, path, replaces WWW-Authenticate: Digest realm="fs_bfis.bizfocused.com", nonce="f9a005f6-d33c-425e-9264-525c669833ef", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ Any ideas?

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pgsql as dsn
by Frank Carmickle 07 Dec '12

07 Dec '12

Hi all I'm trying to use the new pg support. Here's what I get. 2012-12-06 20:00:50.078511 [CONSOLE] switch_loadable_module.c:1834 mod_db unloaded. 2012-12-06 20:00:50.078511 [CRIT] switch_core_sqldb.c:464 Failure! PGSQL NOT AVAILABLE! Can't connect to DSN host=my_hostname.tld dbname=freeswitch user=freeswitch password='stringofupperlowerandnumbers' options='-c client_min_messages=NOTICE' application_name='freeswitch' 2012-12-06 20:00:50.078511 [INFO] switch_time.c:1165 Timezone reloaded 530 definitions 2012-12-06 20:00:50.078511 [CONSOLE] switch_loadable_module.c:1348 Successfully Loaded [mod_db] 2012-12-06 20:00:50.078511 [NOTICE] switch_loadable_module.c:254 Adding Application 'db' 2012-12-06 20:00:50.078511 [NOTICE] switch_loadable_module.c:254 Adding Application 'group' 2012-12-06 20:00:50.078511 [NOTICE] switch_loadable_module.c:298 Adding API Function 'db' 2012-12-06 20:00:50.078511 [NOTICE] switch_loadable_module.c:298 Adding API Function 'group' 2012-12-06 20:00:50.078511 [NOTICE] switch_loadable_module.c:473 Adding Limit interface 'db' Doing a tcpdump on the interface where this traffic should be showing up shows no attempts. This connection will need to be ipv6. Is there something special I need to do to make this work? Thanks for any help you can give. --FC

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