July 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Central FreeSWITCH nodes management with Mod_XML_Curl vs Mod_Lua
by Eugene Azuka 18 Jul '12

18 Jul '12

Thank you FreeSWITCH experts: Michael Collins, Kristian Kielhofner, Vik Killa, Gabriel Gunderson. Thanks you all, i appreciate your comments. Gabriel Gunderson, Anyone who has practical experience of FreeSWITCH and able to reply and give meaningful comment/solution to help request, to me that person an experts. So you are an expert. :) Regards Eugene On Tue, Jul 17, 2012 at 11:41 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. how to get leg_delay_start to work for bridge enterprise (Mario > G) > 2. Re: Username in SUBSCRIBE Request URI (Duvid Rottenberg) > 3. Central FreeSWITCH nodes management with Mod_XML_Curl vs > Mod_Lua (Eugene Azuka) > 4. Re: Central FreeSWITCH nodes management with Mod_XML_Curl vs > Mod_Lua (Vik Killa) > 5. Re: Central FreeSWITCH nodes management with Mod_XML_Curl vs > Mod_Lua (Michael Collins) > 6. Looking for dial plan examples for FS and SPA3102 router > (Todd Bailey) > 7. Re: Central FreeSWITCH nodes management with Mod_XML_Curl vs > Mod_Lua (Gabriel Gunderson) > > > ---------- Forwarded message ---------- > From: Mario G <mario_fs(a)mgtech.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Tue, 17 Jul 2012 09:54:51 -0700 > Subject: [Freeswitch-users] how to get leg_delay_start to work for bridge > enterprise > I am try to delay the second target by 20 seconds. I used [..] and {..} > but no dice. The wiki has them both for enterprise, can someone shed light > on what's wrong, thanks. Main is supposed to ring, 20 secs later the second > target is added while main keeps going, all timeout after a total of 43 > secs. > Mario G > > <action application="bridge" data= > "<originate_timeout=43,origination_caller_id_number=${caller_id_number},origination_caller_id_name=${caller_id_name}>${group_call(main@ > ${domain_name}+E)}:_:{leg_delay_start=20}sofia/gateway/${dial_gateway}/19161234567" > /> > > > ---------- Forwarded message ---------- > From: Duvid Rottenberg <adrottenberg(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Tue, 17 Jul 2012 14:01:20 -0400 > Subject: Re: [Freeswitch-users] Username in SUBSCRIBE Request URI > Apparently this is not currently possible. I have submitted a patch on > JIRA to add a new parameter to the subscriptions section named > user-in-register, when set to true the username will be included in the > request-uri. If/when this patch is approved I will update the wiki. > > On Fri, Jul 13, 2012 at 1:34 PM, Duvid Rottenberg <adrottenberg(a)gmail.com>wrote: > >> I am using embedded freeswitch as a softphone client and I am trying to >> subscribe to call-info on the server, (see config below) but the server is >> responding with a 481 Call/Transaction Does not exist. >> I compared the freeswitch SIP messages with SIP messages sent by a >> polycom phone for this feature and I noticed that freeswitch doesn't send >> the username in the request line. I think that this is causing the 481 >> response. >> >> Polycom Version: >> SUBSCRIBE sip:user@server:5060;transport=udp SIP/2.0 >> Freeswitch: >> SUBSCRIBE sip:server:5060;transport=udp SIP/2.0 >> >> Below is my gateway configuration >> <gateway name='1'> >> <param name='realm' value='SIP_SERVER' /> >> <param name='username' value='SIP_USERNAME' /> >> <param name='password' value='SIP_PASSWORD' /> >> <param name='register' value='true' /> >> <param name='extension' value='SIP_USERNAME' /> >> <param name='extension-in-contact' value='true' /> >> <subscriptions name='x'> >> <subscription event='call-info'> >> <param name='retry-seconds' value='40'/> >> <param name='expire-seconds' value='3600'/> >> <param name='content-type' value=''/> >> </subscription> >> </subscriptions> >> </gateway> >> >> Is there any way to tell freeswitch to include the username in the >> request line? >> >> Thank You, >> Duvid Rottenberg >> > > > > ---------- Forwarded message ---------- > From: Eugene Azuka <eugeneazuka(a)gmail.com> > To: freeswitch-users(a)lists.freeswitch.org > Cc: > Date: Tue, 17 Jul 2012 17:33:56 +0100 > Subject: [Freeswitch-users] Central FreeSWITCH nodes management with > Mod_XML_Curl vs Mod_Lua > Hi FreeSWITCH experts, > > My question is as regards performance and which is the best option. > > I am trying to scale and manage multiple nodes of FreeSWITCH centrally > from my external web server/site using Mod_XML_Curl, But someone more > experience than me says otherwise that Mod_XML_Curl may not be the best > option: Here is his statement below: > > *" The thing I don't like about mod_xml_curl is that to scale it you have > to scale the web server with more listeners. If you run out of listeners > your ability to answer new calls fails. If the web server crashes your > ability to handle calls is gone.* > * > * > *Instead, I use Lua which is embedded into FreeSWITCH as the XML handler. > The Lua reads the info from the database and hands it off to FreeSWITCH. > This approach is not dependent on a service that can fail, or that can run > out of listeners. "* > > > What do you experts think about this comment above? > > Wouldn't Lua reading directly from database still face some of the issues > i may be running from using mod_xml_curl, like database crashing, slow > reading from database? > > Unlike using mod_xml_curl whereby i can optimise my code to make use of > memcached or radis to reduce some calls to database, can i still do such > with mod_lua? > > > So what do experts think, should i go with Mod_Lua approach just as he > said above or should i continue with using Mod_XML_Curl? > > You opinion will be appreciated. > > Regards > > Eugene > > > > > > > ---------- Forwarded message ---------- > From: Vik Killa <vipkilla(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Tue, 17 Jul 2012 15:30:53 -0400 > Subject: Re: [Freeswitch-users] Central FreeSWITCH nodes management with > Mod_XML_Curl vs Mod_Lua > I dont understand what they meant by 'listeners' > AFAIK if the web server process is running it will always return the XML > > > > > ---------- Forwarded message ---------- > From: Michael Collins <msc(a)freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Tue, 17 Jul 2012 13:56:11 -0700 > Subject: Re: [Freeswitch-users] Central FreeSWITCH nodes management with > Mod_XML_Curl vs Mod_Lua > > > On Tue, Jul 17, 2012 at 12:30 PM, Vik Killa <vipkilla(a)gmail.com> wrote: > >> I dont understand what they meant by 'listeners' >> AFAIK if the web server process is running it will always return the XML >> > > Last time I checked, there were one or two websites out there that could > handle thousands of requests per second. The web has given birth to tools > that allow for many, many concurrent requests. The LAMP stack on a beefy > machine can do quite a lot of traffic. Furthermore, MySQL/Postgres/et al > all have backup/redundancy/HA options built in, as does Apache/HTTP. It > seems to me that you could scale farther and have more redundancy using > these time-tested tools. Just my $0.02. > > -MC > > P.S. - Don't get me wrong - I really like Lua. I just don't know if it's > really a "better" solution to this problem. > > > ---------- Forwarded message ---------- > From: Todd Bailey <toddb(a)toddbailey.net> > To: freeswitch <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Tue, 17 Jul 2012 15:37:51 -0700 > Subject: [Freeswitch-users] Looking for dial plan examples for FS and > SPA3102 router > Hi All, > > > I'm having issues getting a dial plan to work on FS and a Cisco SPA 3102 > router. > > when I dial 0, 1 or 9 plus a 10 number, I get to the router's dial tone > but I have to reenter the number I want to connect to. > > The expected action is to only need to enter the number to dial one > > Can some one provide dial plan and/or other config file example on how > to resolve this issue? > > here is what I have so far: > > /usr/local/freeswitch/conf/dialplan/default.xml > >  > > <extension name="external"> > > <condition field="destination_number" > expression="^(1{0,1,9}\d{10})$"> > > <action application="set" > data="effective_caller_id_number=12223334444"/> >  >  >  >  > <action application="bridge" > data="sofia/internal/$1@192.168.1.5:5061" /> > </condition> > </extension> > > > /usr/local/freeswitch/conf/dialplan/default/00_spa3102.xml > > <include> > <extension name="To PSTN"> > <condition field="destination_number" expression="(.*)"> > <action application="bridge" > data="sofia/internal/${destination_number}@192.168.1.5:5061" /> > </condition> > </extension> > </include> > > > > > > > > ---------- Forwarded message ---------- > From: Gabriel Gunderson <gabe(a)gundy.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Tue, 17 Jul 2012 16:40:41 -0600 > Subject: Re: [Freeswitch-users] Central FreeSWITCH nodes management with > Mod_XML_Curl vs Mod_Lua > Comments inline: > > On Tue, Jul 17, 2012 at 10:33 AM, Eugene Azuka <eugeneazuka(a)gmail.com> > wrote: > > " The thing I don't like about mod_xml_curl is that to scale it you have > to > > scale the web server with more listeners. If you run out of listeners > your > > ability to answer new calls fails. If the web server crashes your > ability to > > handle calls is gone. > > Scaling web servers is an easy, well known and a common problem to > solve. Yes, you have to make sure you have the resources, but this it > true of any technology out there. > > > > Instead, I use Lua which is embedded into FreeSWITCH as the XML handler. > The > > Lua reads the info from the database and hands it off to FreeSWITCH. This > > approach is not dependent on a service that can fail, or that can run > out of > > listeners. " > > This is a great approach, but I don't think the logic is sound. Lua > (if it's doing anything fancy) will run out of resources that it > depends on -- memory, database connections, sockets etc. > > If one accepts that they'll have to build out to *really* scale > something, they'll also come to appreciate the fact that they can move > the HTTP stack to another box when needed (or 50 other boxen if > required). The Lua (or any other embedded language) is pretty well > tied to that same box. > > In the end, it probably doesn't matter. If you end up scaling to the > ends of the Earth, you'll have to rewrite it anyway. Just give thanks > that you've found FreeSWITCH and it was flexible enough to give you > the amazing configurability needed to build it in anyway you like :) > > > > What do you experts think about this comment above? > > There are experts on this list now?! Awesome ;) > > > > Wouldn't Lua reading directly from database still face some of the > issues i > > may be running from using mod_xml_curl, like database crashing, slow > reading > > from database? > > Yep, see above. > > > > Unlike using mod_xml_curl whereby i can optimise my code to make use of > > memcached or radis to reduce some calls to database, can i still do such > > with mod_lua? > > If you want and Lua supports it. > > > > So what do experts think, should i go with Mod_Lua approach just as he > said > > above or should i continue with using Mod_XML_Curl? > > Your choice! Happy hacking! > > > Best, > Gabe > > > > _______________________________________________ > 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 > >

3 3

Need help in IVR
by Piyush Sharma 18 Jul '12

18 Jul '12

Hello Everyone, I am trying to create an IVR with menu and sub menus, but getting following error when I make the call, 2012-07-17 11:05:37.755034 [ERR] switch_ivr_menu.c:870 Unable to build xml menu 2012-07-17 11:05:37.755034 [ERR] switch_ivr_menu.c:870 Unable to build xml menu 2012-07-17 11:05:37.755034 [ERR] mod_dptools.c:1774 Unable to create menu I am new to this, so there might be little or a big error. My XML contents in ... conf/ivr_menus is <include> <menu name="ps" greet-long="phrase:ps_main" greet-short="phrase:ps_short" exit-sound="ivr/good_bye.wav" invalid-sound="ivr/ivr-not_available.wav" tts-engine="flite" tts-voice="rms" timeout="3000" max-failures="2" phrase_lang="en" digit-len="1"> <entry action="menu-sub" digits="1" param="ps1"/> <entry action="menu-top" digits="9"/> </menu> <menu name="ps1" greet-long="phrase:ps1_main" greet-short="phrase:ps1_short" exit-sound="ivr/good_bye.wav" invalid-sound="ivr/ivr-not_available.wav" timeout="3000" max-failure="3"> <entry action="menu-top" digit="*"/> </menu> </include> and in .../en/demo is <include> <macro name="ps_main" pause="100"> <input pattern="(.*)"> <match> <action function="play-file" data="ivr/1_welcome_to_SGRH.wav"/> <action function="play-file" data="silence_stream://500"/> <action function="play-file" data="ivr/2_Press1_for_Eng_Hindi_k_liye_2_Dabayen.wav"/> </match> </input> </macro> <macro name="ps_short" pause="100"> <input pattern="(.*)"> <match> <action function="play-file" data="silence_stream://1000"/> <action function="play-file" data="ivr/no_input_recd_try_again.wav"/> </match> </input> </macro> <macro name="ps1_main" pause="100"> <input pattern="(.*)"> <match> <action function="play-file" data="ivr/3_for_Appointment_1_for_Admission_2_for_Lab_reports_3.wav"/> <action function="play-file" data="ivr/4_Press_1_Lab_reports_2_for_Histopatho.wav"/> </match> </input> </macro> <macro name="ps1_short" pause="100"> <input pattern="(.*)"> <match> <action function="play-file" data="silence_stream://1000"/> <action function="play-file" data="ivr/eng_no_input_try_again.wav"/> </match> </input> </macro> </include> Thanks everyone, thanks for this support forum !!! -- Regards, Piyush Sharma.

4 8

Strange 400 with some clients
by Luis Daniel Lucio Quiroz 18 Jul '12

18 Jul '12

Using FS 1.0, I'm having problems registering some devices, Android 4.0.4, Linphone 3.4.0 under Wifi works Android 2.3.4, Linphone 3.4.0 under Wifi works it fails here iPhone IOS 5.1.1, Linphone 3.5.2 under Wifi fails iPod IOS 5.1.1, Linphone 3.5.2 under Wifi fails As far as i seen im getting error 400 when IOS is trying to register here is failling dump REGISTER sip:corporate.xxxx.com SIP/2.0 Via: SIP/2.0/UDP 189.146.85.112:10649;rport;branch=z9hG4bK842453785 From: <sip:107@corporate.xxxx.com>;tag=1058009119 To: <sip:107@corporate.xxxx.com> Call-ID: 508811206 CSeq: 19 REGISTER Contact: <sip:107@189.146.85.112:10649:10649;line=06e1d79779fb5b1> Max-Forwards: 70 User-Agent: Linphone/3.5.2 (eXosip2/3.6.0) Expires: 600 Content-Length: 0 SIP/2.0 400 Bad Contact Header Via: SIP/2.0/UDP 189.146.85.112:10649;rport=10649;branch=z9hG4bK842453785 From: <sip:107@corporate.xxxx.com>;tag=1058009119 To: <sip:107@corporate.xxxx.com>;tag=1FvU4KjDH0ggg Call-ID: 508811206 CSeq: 19 REGISTER Content-Length: 0 here is workig request REGISTER sip:corporate.xxxx.com SIP/2.0 Via: SIP/2.0/UDP 189.146.85.112:5060;rport;branch=z9hG4bK86735004 From: <sip:104@corporate.xxxx.com>;tag=134885940 To: <sip:104@corporate.xxxx.com> Call-ID: 591954255 CSeq: 2 REGISTER Contact: <sip:104@189.146.85.112:5060;line=26e5d923f2f5be6> Authorization: Digest username="104", realm="corporate.xxxx.com", nonce="f418e392-b7d1-43e9-98d5-b841134ad19c", uri="sip:corporate.xxxx.com", response="f599fa83cb2660595d61bb3cadba6c5b", algorithm=MD5, cnonce="0a4f113b", qop=auth, nc=00000001 Max-Forwards: 70 User-Agent: Linphone/3.4.0 (eXosip2/unknown) Expires: 3600 Content-Length: 0 SIP/2.0 200 OK Via: SIP/2.0/UDP 189.146.85.112:5060;rport=5060;branch=z9hG4bK86735004 From: <sip:104@corporate.xxxx.com>;tag=134885940 To: <sip:104@corporate.xxxx.com>;tag=Xc0Hp99F0S0Um Call-ID: 591954255 CSeq: 2 REGISTER Contact: <sip:104@189.146.85.112:5060;line=26e5d923f2f5be6>;expires=3600 Date: Wed, 18 Jul 2012 02:54:14 GMT User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-261505a 2012-02-13 13-50-21 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Length: 0 Just wondering how to workarround this regards LD

3 3

Central FreeSWITCH nodes management with Mod_XML_Curl vs Mod_Lua
by Eugene Azuka 18 Jul '12

18 Jul '12

Hi FreeSWITCH experts, My question is as regards performance and which is the best option. I am trying to scale and manage multiple nodes of FreeSWITCH centrally from my external web server/site using Mod_XML_Curl, But someone more experience than me says otherwise that Mod_XML_Curl may not be the best option: Here is his statement below: *" The thing I don't like about mod_xml_curl is that to scale it you have to scale the web server with more listeners. If you run out of listeners your ability to answer new calls fails. If the web server crashes your ability to handle calls is gone.* * * *Instead, I use Lua which is embedded into FreeSWITCH as the XML handler. The Lua reads the info from the database and hands it off to FreeSWITCH. This approach is not dependent on a service that can fail, or that can run out of listeners. "* What do you experts think about this comment above? Wouldn't Lua reading directly from database still face some of the issues i may be running from using mod_xml_curl, like database crashing, slow reading from database? Unlike using mod_xml_curl whereby i can optimise my code to make use of memcached or radis to reduce some calls to database, can i still do such with mod_lua? So what do experts think, should i go with Mod_Lua approach just as he said above or should i continue with using Mod_XML_Curl? You opinion will be appreciated. Regards Eugene

5 4

Username in SUBSCRIBE Request URI
by Duvid Rottenberg 17 Jul '12

17 Jul '12

I am using embedded freeswitch as a softphone client and I am trying to subscribe to call-info on the server, (see config below) but the server is responding with a 481 Call/Transaction Does not exist. I compared the freeswitch SIP messages with SIP messages sent by a polycom phone for this feature and I noticed that freeswitch doesn't send the username in the request line. I think that this is causing the 481 response. Polycom Version: SUBSCRIBE sip:user@server:5060;transport=udp SIP/2.0 Freeswitch: SUBSCRIBE sip:server:5060;transport=udp SIP/2.0 Below is my gateway configuration <gateway name='1'> <param name='realm' value='SIP_SERVER' /> <param name='username' value='SIP_USERNAME' /> <param name='password' value='SIP_PASSWORD' /> <param name='register' value='true' /> <param name='extension' value='SIP_USERNAME' /> <param name='extension-in-contact' value='true' /> <subscriptions name='x'> <subscription event='call-info'> <param name='retry-seconds' value='40'/> <param name='expire-seconds' value='3600'/> <param name='content-type' value=''/> </subscription> </subscriptions> </gateway> Is there any way to tell freeswitch to include the username in the request line? Thank You, Duvid Rottenberg

1 1

how to get leg_delay_start to work for bridge enterprise
by Mario G 17 Jul '12

17 Jul '12

I am try to delay the second target by 20 seconds. I used [..] and {..} but no dice. The wiki has them both for enterprise, can someone shed light on what's wrong, thanks. Main is supposed to ring, 20 secs later the second target is added while main keeps going, all timeout after a total of 43 secs. Mario G <action application="bridge" data="<originate_timeout=43,origination_caller_id_number=${caller_id_number},origination_caller_id_name=${caller_id_name}>${group_call(main@${domain_name}+E)}:_:{leg_delay_start=20}sofia/gateway/${dial_gateway}/19161234567"/>

1 0

Need French-Candian translator
by Michael Collins 17 Jul '12

17 Jul '12

Bonjour! We could use some assistance with a few sounds that need to be translated from English to French. These will be prompts that are recorded and made available to the community at large. Please email me off list if you are able to assist with translating about 20 or so phrases. Thanks, Michael

1 0

setting max-forwards?
by Brett Wilson 17 Jul '12

17 Jul '12

Hello, Can anyone tell me how and where to set max-forwards? I can't find much of anything on the net. I get exchange routing errors and sometimes calls will not forward to an extension. Also, On another install, emailing the voicemail does not work. The settings seem all ok, the log says emailing, but I did some snooping with filesystem monitoring and the script to send the email is never touched by freeswitch in any way. Where should I start diagnosing this? ******************************************* Brett Wilson IT Department Launch 3 Ventures, LLC 134 Myer Street Hackensack, NJ 07601 Phone: 877.878.9134 Fax: 646.536.3866 Email: <mailto:Brett.Wilson@launch3.net> Brett.Wilson(a)launch3.net AOL IM: <mailto:Brett.Wilson@launch3.net> Brett.Wilson(a)launch3.net <http://www.launch3.net/> www.Launch3.net <http://www.launch3telecom.com/> www.Launch3telecom.com ******************************************* <http://launch3telecom.blogspot.com/> Description: Description: Description: Blogger-logo <http://www.facebook.com/pages/Launch-3-Telecom/165610820178720> Description: Description: Description: FaceBook-Logo <https://twitter.com/launch3telecom> Description: Description: Description: Twitter-Logo <https://plus.google.com/102713212642017290959> Description: Description: Description: GPlus-Logo

2 1

How can I use fs_path when originating an external call through a gateway?
by Matt Stockton 17 Jul '12

17 Jul '12

Hi all, I am trying to do the following: Originate a call from Freeswitch to an external gateway. I want the SIP signaling to go through a different Freeswitch box that is acting as an outbound proxy. It seems that I should be able to use fs_path to do this, however I am having trouble getting this to work. I have tried the following originate commands, and for each one, the SIP signaling is not going through the other_fs box. Note that the originate is working just fine, its just going directly to the gateway: originate sofia/gateway/my_gateway/+14145551212;fs_path=sip:other_fs &park() originate sofia/gateway/my_gateway/+14145551212;fs_path=sip:other_fs:5060 &park() originate 'sofia/gateway/my_gateway/+14145551212;fs_path=sip:other_fs' &park() originate 'sofia/gateway/my_gateway/+14145551212;fs_path=sip:other_fs:5060' &park() Am I doing something wrong with fs_path? My goal is to have a FS node (the other_fs) act as an outbound proxy to the gateway so that the gateway sees all SIP signaling traffic come from other_fs. Currently, I am using OpenSIPS to act as the outbound proxy but ideally I would just have Freeswitch nodes, so I am trying to remove OpenSIPS from the SIP path. Any help on this is appreciated! - Matt

1 0

Not droping calls after BUSY signal
by Saulius Kibilda 17 Jul '12

17 Jul '12

Hi All I have Dinstar GSM gateway and configured like in this how to -> http://wiki.freeswitch.org/wiki/Dinstar_GSM_gateway_FreeSwitch_HowTo Everything works well except when calling from an IP phone to a mobile phone and mobile phone dropps the call. Freeswitch does not understand the BUSY signal and calls again sending INVITE and making new call to mobile phone. Hare is SIP debug http://pastebin.freeswitch.org/19539 BTW: in Freeswitch 1.1-beta I not observed this problem Dinstar with Asterisk works fine, so problem i think not in Dinstar gateway. Saulius

1 0