December 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

SIP Phone Integration
by James Mbuthia 06 Dec '10

06 Dec '10

Hi guys, Am new to Freeswitch and am looking for info that can help me develop a web-based SIP Phone I am developing a web-based SIP Phone based on PHP/MySQL and OpenSIPS. I have developed the component of the SIP phone responsible for the 3way handshake and SDP offer. My challenge is to now integrate the application to a rtp stack which will enable the app to pick up audio and transmit it over the internet. Ultimately I want to connect the app to the PSTN using a media server such as Freeswitch or Asterisk. I want to use the Speex codec. I wanted to know whether Freeswitch has a rtp stack integrated with speex and whether there any tutorials on how integration to a softphone can be done. Any pointers or ideas from you would be very helpful and highly appreaciated. regards, James Mbuthia

4 7

managed code in CentOS
by srinivasula reddy 06 Dec '10

06 Dec '10

HI All, We are trying to migrate from windows to linux, if we are trying to build managed code in centos, we are facing few issues, /home/hydadmin/freeSwitch/ freeswitch-1.0.6/src/mod/languages/mod_managed/mod_managed.cpp:211: error: aggregate 'MonoAssemblyName name' has incomplete type and cannot be defined make[1]: *** [mod_managed.lo] Error 1 we enabled the mod_managed module in modules_conf.xml, i have installed mono2.8.1, following instruction given by freeswitch, they are asking us to apply mono28.patch, we could not able to get that patch, if mono28.patch is the problem for the above error? if that is the error where will we get that patch? if any other reason please let us know. THanks Srinivas

3 2

mod_khomp
by Geovani Ricardo Wiedenhoft 06 Dec '10

06 Dec '10

The Endpoint and the documentation for the Khomp boards (mod_khomp) are available on the official FreeSWTICH branch. Also, the documents are available on the wiki. http://wiki.freeswitch.org/wiki/Khomp The version is compatible with all Khomp boards (SPX series): - FXS - FXO - E1 - R2, R2 DTMF and OpenCAS in Hardware, ISDN (User, Network), OpenCCS, LineSide, CAS_EL7, E1LC - GSM with send/receive SMS (boards and usb devices) - Passive record (R2, ISDN, OpenCAS, OpenCCS and FXO) - kommuter Visit our site to more information about Khomp products: http://www.khomp.com.br Thank you. :) Khomp development team

1 0

Problem with SIP 180 + FreeTDM
by Juan Antonio Ibañez Santorum 06 Dec '10

06 Dec '10

Hello boys! I have one problem after updating from openzap to freetdm. Now, whe FS receives 'Proceeding' from PRI sends a SIP 180 which makes the phone to play a fake ring tone. After some seconds, when FS receives 'Alerting' from PRI, FS sends a SIP 183 with the real ring tone. Is there any way to avoid first SIP 180? If the calle is busy, I can hear a ringing tone between SIP 180 and SIP 183 and a busy tone after SIP 183. Same if calle is a mobile phone and it is out of signal. Regards

3 9

Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 54, Issue 34
by srinivasula reddy 06 Dec '10

06 Dec '10

HI David, Thanks for the replay, when we trying to build mono_managed using linux in CentOS We are getting the follwing error. /home/hydadmin/freeSwitch/freeswitch-1.0.6/src/mod/languages/mod_managed/mod_managed.cpp: In function 'switch_status_t loadRuntime()': /home/hydadmin/freeSwitch/freeswitch-1.0.6/src/mod/languages/mod_managed/mod_managed.cpp:211: error: aggregate 'MonoAssemblyName name' has incomplete type and cannot be defined make[1]: *** [mod_managed.lo] Error 1 I enable the module mod_managed in modules.conf.xml, Can you please let me know if you have any idea? Thanks Sriniavas 2010/12/5 <freeswitch-users-request(a)lists.freeswitch.org> > 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: Error loading mod_managed in CentOS (David Brazier) > 2. Re: building mod_managed fails (David Brazier) > 3. Re: Problem with SIP 180 + FreeTDM (Juan Antonio Iba?ez Santorum) > 4. GIT/SVN Repo Maint. (Brian West) > 5. Possible Loopback / Originate Race? (Jeroen C. van Gelderen) > > > ---------- Forwarded message ---------- > From: David Brazier <davidjbrazier(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Date: Sat, 4 Dec 2010 00:39:25 +0000 > Subject: Re: [Freeswitch-users] Error loading mod_managed in CentOS > Make sure you uncomment "languages/mod_managed" is uncommented in > modules.conf and follow the Mono 2.8 instructions on > > http://wiki.freeswitch.org/wiki/Mod_managed > > Don't try mod_mono & mod_mono_managed - that is old stuff. > > David > > > > > ---------- Forwarded message ---------- > From: David Brazier <davidjbrazier(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Date: Sat, 4 Dec 2010 00:43:43 +0000 > Subject: Re: [Freeswitch-users] building mod_managed fails > On Wed, Dec 1, 2010 at 5:16 PM, David Brazier <davidjbrazier(a)gmail.com> > wrote: > > I'll try to investigate why the swig output wasn't right. > > I found out why - you need SWIG 2.0 - SWIG 1.3 that I had doesn't do it > right. > > > > > ---------- Forwarded message ---------- > From: "Juan Antonio Ibañez Santorum" <juanito1982(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Date: Sat, 4 Dec 2010 14:24:06 +0100 > Subject: Re: [Freeswitch-users] Problem with SIP 180 + FreeTDM > You are true. Last git version works as expected. > > Thank you very much > > 2010/12/2 David Yat Sin <dyatsin(a)sangoma.com> > >> Hi Juan Antonio, >> >> Can you pull the latest Freeswitch git. This should be fixed now. >> >> >> David >> >> *David Yat Sin, BEng* >> Software Engineer >> Sangoma Technologies >> 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada >> >> t. +1 800 388 2475 x119 >> t. +1 905 474 1990 x119 >> f. +1 905 474 9223 [image: SANGOMA]<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_ca…> >> Products<http://sangoma.com/products?utm_source=signature&utm_medium=email&utm_campa…>| >> Solutions<http://sangoma.com/solutions?utm_source=signature&utm_medium=email&utm_camp…>| >> Events<http://sangoma.com/about_us/events?utm_source=signature&utm_medium=email&ut…>| >> Contact<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_ca…>| >> Wiki<http://wiki.sangoma.com/?utm_source=signature&utm_medium=email&utm_campaign…>| >> Facebook<http://www.facebook.com/pages/Sangoma-VoIP-Cards/43578453335?utm_source=sig…>| >> Twitter<http://www.twitter.com/sangoma?utm_source=signature&utm_medium=email&utm_ca…> >> >> On 11/24/2010 1:38 PM, Juan Antonio Ibañez Santorum wrote: >> >> Any news? >> >> 2010/11/18 David Yat Sin <dyatsin(a)sangoma.com> >> >>> Hi Juan Antonio, >>> We are looking into this and will get back to you in a day or two. >>> >>> David >>> >>> *David Yat Sin, BEng* >>> Software Engineer >>> Sangoma Technologies >>> 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada >>> >>> t. +1 800 388 2475 x119 >>> t. +1 905 474 1990 x119 >>> f. +1 905 474 9223 [image: SANGOMA]<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_ca…> >>> Products<http://sangoma.com/products?utm_source=signature&utm_medium=email&utm_campa…>| >>> Solutions<http://sangoma.com/solutions?utm_source=signature&utm_medium=email&utm_camp…>| >>> Events<http://sangoma.com/about_us/events?utm_source=signature&utm_medium=email&ut…>| >>> Contact<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_ca…>| >>> Wiki<http://wiki.sangoma.com/?utm_source=signature&utm_medium=email&utm_campaign…>| >>> Facebook<http://www.facebook.com/pages/Sangoma-VoIP-Cards/43578453335?utm_source=sig…>| >>> Twitter<http://www.twitter.com/sangoma?utm_source=signature&utm_medium=email&utm_ca…> >>> >>> On 11/18/2010 3:50 PM, Juan Antonio Ibañez Santorum wrote: >>> >>> Hello boys! >>> >>> I have one problem after updating from openzap to freetdm. Now, whe >>> FS receives 'Proceeding' from PRI sends a SIP 180 which makes the phone to >>> play a fake ring tone. After some seconds, when FS receives 'Alerting' from >>> PRI, FS sends a SIP 183 with the real ring tone. >>> >>> Is there any way to avoid first SIP 180? If the calle is busy, I can >>> hear a ringing tone between SIP 180 and SIP 183 and a busy tone after SIP >>> 183. Same if calle is a mobile phone and it is out of signal. >>> >>> Regards >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershtt… >>> >>> >>> _______________________________________________ >>> 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 listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershtt… >> >> >> _______________________________________________ >> 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: Brian West <brian(a)freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Date: Sat, 4 Dec 2010 17:35:02 -0600 > Subject: [Freeswitch-users] GIT/SVN Repo Maint. > I'm going to be doing some work on the repo to move it this weekend so > please be patient if you can't access it while I'm moving it to Dallas > Sunday December 5th 2010. > > Thanks, > Brian > > > > > > ---------- Forwarded message ---------- > From: "Jeroen C. van Gelderen" <slim(a)thegreek.com> > To: "'FreeSWITCH Users Help'" <freeswitch-users(a)lists.freeswitch.org> > Date: Sun, 5 Dec 2010 02:23:33 -0500 > Subject: [Freeswitch-users] Possible Loopback / Originate Race? > > Hi, > > > > (Filed in JIRA FS-2899. Please let me know if posting this on the list as > well is frowned upon.) > > > > I’m using switch_limit in a loopback extension with the following > problematic dialplan fragment (full dialplan context at end of message): > > > > <extension name="limit_exceeded"> > > <condition field="destination_number" expression="^limit_exceeded$"> > > > > <action application="hangup" data="NORMAL_CIRCUIT_CONGESTION"/> > > > > </condition> > > </extension> > > > > This works fine in MOST cases with the expected result: > > switch_ivr_originate.c:3448 Originate Resulted in Error Cause: 34 > [NORMAL_CIRCUIT_CONGESTION] > > > > Every once in a while though this very same extension will magically yield: > > switch_ivr_originate.c:3448 Originate Resulted in Error Cause: 19 > [NO_ANSWER] > > > > Looking at the log snippets pasted below the correct result (34) appears > when this sequence of events occurs (first snippet): > > > > mod_loopback.c:425 Hangup loopback/18884447693-a [CS_CONSUME_MEDIA] > [NORMAL_CIRCUIT_CONGESTION] > > switch_ivr_originate.c:3448 Originate Resulted in Error Cause: 34 > [NORMAL_CIRCUIT_CONGESTION] > > > > The incorrect answer (19) appears whenever this sequence shows in the logs: > > > > switch_ivr_originate.c:3448 Originate Resulted in Error Cause: 19 > [NO_ANSWER] > > mod_loopback.c:425 Hangup loopback/18884447701-a [CS_CONSUME_MEDIA] > [NORMAL_CIRCUIT_CONGESTION] > > > > Pouring over my logs this ALWAYS is the case. Whenever the wrong error > cause appears the switch_ivr_originate call precedes the mod_loopback call. > Conversely whenever the correct error cause is given the call sequence is > reversed. > > > > I can reproduce this at will by rapidly invoking the loopback extension in > the dialplan pasted at the bottom of this mail. > > > > Any help is appreciated. > > > > Cheers, > > -Slim > > > > ----8<----8<----8<----8<----8<----8<----8<---- > > EXECUTE loopback/18884447693-b hangup(NORMAL_CIRCUIT_CONGESTION) > > 2010-12-05 01:27:13.318020 [DEBUG] switch_channel.c:2455 > (loopback/18884447693-b) Callstate Change EARLY -> HANGUP > > 2010-12-05 01:27:13.318020 [NOTICE] mod_dptools.c:906 Hangup > loopback/18884447693-b [CS_EXECUTE] [NORMAL_CIRCUIT_CONGESTION] > > > > 2010-12-05 01:27:13.319023 [DEBUG] switch_channel.c:2471 Send signal > loopback/18884447693-b [KILL] > > 2010-12-05 01:27:13.319023 [DEBUG] mod_loopback.c:468 > loopback/18884447693-b CHANNEL KILL > > 2010-12-05 01:27:13.319023 [DEBUG] switch_core_session.c:1083 Send signal > loopback/18884447693-b [BREAK] > > 2010-12-05 01:27:13.319023 [DEBUG] mod_loopback.c:468 > loopback/18884447693-b CHANNEL KILL > > > > 2010-12-05 01:27:13.320024 [DEBUG] switch_core_session.c:1993 > loopback/18884447693-b skip receive message [APPLICATION_EXEC_COMPLETE] > (chann > > el is hungup already) > > > > 2010-12-05 01:27:13.320024 [DEBUG] switch_core_state_machine.c:366 > (loopback/18884447693-b) State EXECUTE going to sleep > > 2010-12-05 01:27:13.320024 [DEBUG] switch_core_state_machine.c:320 > (loopback/18884447693-b) Running State Change CS_HANGUP > > 2010-12-05 01:27:13.321011 [DEBUG] switch_core_state_machine.c:557 > (loopback/18884447693-b) State HANGUP > > > > > > 2010-12-05 01:27:13.321983 [DEBUG] mod_loopback.c:414 > loopback/18884447693-b CHANNEL HANGUP > > 2010-12-05 01:27:13.321983 [DEBUG] switch_channel.c:2455 > (loopback/18884447693-a) Callstate Change EARLY -> HANGUP > > > > 2010-12-05 01:27:13.321983 [NOTICE] mod_loopback.c:425 Hangup > loopback/18884447693-a [CS_CONSUME_MEDIA] [NORMAL_CIRCUIT_CONGESTION] > > > > 2010-12-05 01:27:13.322995 [DEBUG] switch_ivr_originate.c:3448 Originate > Resulted in Error Cause: 34 [NORMAL_CIRCUIT_CONGESTION] > > > > ----8<----8<----8<----8<----8<----8<----8<---- > > > > EXECUTE loopback/18884447701-b hangup(NORMAL_CIRCUIT_CONGESTION) > > 2010-12-05 01:27:14.364448 [DEBUG] switch_channel.c:2455 > (loopback/18884447701-b) Callstate Change EARLY -> HANGUP > > 2010-12-05 01:27:14.365470 [NOTICE] mod_dptools.c:906 Hangup > loopback/18884447701-b [CS_EXECUTE] [NORMAL_CIRCUIT_CONGESTION] > > > > 2010-12-05 01:27:14.365470 [DEBUG] switch_channel.c:2471 Send signal > loopback/18884447701-b [KILL] > > 2010-12-05 01:27:14.365470 [DEBUG] mod_loopback.c:468 > loopback/18884447701-b CHANNEL KILL > > 2010-12-05 01:27:14.365470 [DEBUG] switch_core_session.c:1083 Send signal > loopback/18884447701-b [BREAK] > > 2010-12-05 01:27:14.366485 [DEBUG] mod_loopback.c:468 > loopback/18884447701-b CHANNEL KILL > > > > 2010-12-05 01:27:14.366485 [DEBUG] switch_core_session.c:1993 > loopback/18884447701-b skip receive message [APPLICATION_EXEC_COMPLETE] > (chann > > el is hungup already) > > > > 2010-12-05 01:27:14.366485 [DEBUG] switch_core_state_machine.c:366 > (loopback/18884447701-b) State EXECUTE going to sleep > > 2010-12-05 01:27:14.367508 [DEBUG] switch_core_state_machine.c:320 > (loopback/18884447701-b) Running State Change CS_HANGUP > > 2010-12-05 01:27:14.369585 [DEBUG] switch_core_state_machine.c:557 > (loopback/18884447701-b) State HANGUP > > > > 2010-12-05 01:27:14.369585 [DEBUG] mod_loopback.c:414 > loopback/18884447701-b CHANNEL HANGUP > > 2010-12-05 01:27:14.369585 [DEBUG] switch_channel.c:2455 > (loopback/18884447701-a) Callstate Change EARLY -> HANGUP > > > > 2010-12-05 01:27:14.369585 [DEBUG] switch_ivr_originate.c:3448 Originate > Resulted in Error Cause: 19 [NO_ANSWER] > > > > 2010-12-05 01:27:14.369585 [NOTICE] mod_loopback.c:425 Hangup > loopback/18884447701-a [CS_CONSUME_MEDIA] [NORMAL_CIRCUIT_CONGESTION] > > > > ----8<----8<----8<----8<----8<----8<----8<---- > > > > > > <context name="tech-outbound"> > > > > <extension name="tech-outbound"> > > <condition field="destination_number" expression="^1888(\d{7})$"> > > > > <action application="info"/> > > <action application="set" data="loopback_bowout=false"/> > > <action application="limit" data="db outgoing goip 1" /> > > <action application="bridge" > data="{ignore_early_media=true}sofia/internal/${ > destination_number}@192.168.3.11:5060<http://destination_number%7D%40192.168.3.11%3A5060> > "/> > > > > </condition> > > </extension> > > > > <extension name="limit_exceeded"> > > <condition field="destination_number" expression="^limit_exceeded$"> > > > > <action application="<nothing> OR <answer> OR <pre_answer>"/>  > > > > <action application="hangup" data="NORMAL_CIRCUIT_CONGESTION"/> > > > > </condition> > > </extension> > > > > </context> > > _______________________________________________ > 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 > > -- Srinivasula Reddy K

2 1

Asterisk SCF vs FreeSWITCH
by Carlos Romero 05 Dec '10

05 Dec '10

hello what is the difference between Asterisk SCF vs FreeSWITCH? https://wiki.asterisk.org/wiki/display/TOP/Asterisk+SCF+Home

4 4

event from LUA to phones
by Tóth Róbert 05 Dec '10

05 Dec '10

Hi I would like send an event from LUA to a phone, with this header: Event: x-gs-screen (This message makes the Grandstream phone refresh its Idle-Screen) I tried this: eout = freeswitch.Event("NOTIFY"); eout.addHeader(eout, "profile", "mellek-belso"); eout.addHeader(eout, "to-uri", "sip:50@192.168.1.10"); eout.addHeader(eout, "from-uri", "sip:50@192.168.1.10"); eout.addHeader(eout, "event-string", "x-gs-screen"); eout.fire(eout); and played with it a lot, e.g. add more headers, change the IPs, ... but the SIP message does not arrive to the phone, altough there are not any error message. How can i do this? Thanks

1 0

Possible Loopback / Originate Race?
by Jeroen C. van Gelderen 05 Dec '10

05 Dec '10

Hi, (Filed in JIRA FS-2899. Please let me know if posting this on the list as well is frowned upon.) I'm using switch_limit in a loopback extension with the following problematic dialplan fragment (full dialplan context at end of message): <extension name="limit_exceeded"> <condition field="destination_number" expression="^limit_exceeded$"> <action application="hangup" data="NORMAL_CIRCUIT_CONGESTION"/> </condition> </extension> This works fine in MOST cases with the expected result: switch_ivr_originate.c:3448 Originate Resulted in Error Cause: 34 [NORMAL_CIRCUIT_CONGESTION] Every once in a while though this very same extension will magically yield: switch_ivr_originate.c:3448 Originate Resulted in Error Cause: 19 [NO_ANSWER] Looking at the log snippets pasted below the correct result (34) appears when this sequence of events occurs (first snippet): mod_loopback.c:425 Hangup loopback/18884447693-a [CS_CONSUME_MEDIA] [NORMAL_CIRCUIT_CONGESTION] switch_ivr_originate.c:3448 Originate Resulted in Error Cause: 34 [NORMAL_CIRCUIT_CONGESTION] The incorrect answer (19) appears whenever this sequence shows in the logs: switch_ivr_originate.c:3448 Originate Resulted in Error Cause: 19 [NO_ANSWER] mod_loopback.c:425 Hangup loopback/18884447701-a [CS_CONSUME_MEDIA] [NORMAL_CIRCUIT_CONGESTION] Pouring over my logs this ALWAYS is the case. Whenever the wrong error cause appears the switch_ivr_originate call precedes the mod_loopback call. Conversely whenever the correct error cause is given the call sequence is reversed. I can reproduce this at will by rapidly invoking the loopback extension in the dialplan pasted at the bottom of this mail. Any help is appreciated. Cheers, -Slim ----8<----8<----8<----8<----8<----8<----8<---- EXECUTE loopback/18884447693-b hangup(NORMAL_CIRCUIT_CONGESTION) 2010-12-05 01:27:13.318020 [DEBUG] switch_channel.c:2455 (loopback/18884447693-b) Callstate Change EARLY -> HANGUP 2010-12-05 01:27:13.318020 [NOTICE] mod_dptools.c:906 Hangup loopback/18884447693-b [CS_EXECUTE] [NORMAL_CIRCUIT_CONGESTION] 2010-12-05 01:27:13.319023 [DEBUG] switch_channel.c:2471 Send signal loopback/18884447693-b [KILL] 2010-12-05 01:27:13.319023 [DEBUG] mod_loopback.c:468 loopback/18884447693-b CHANNEL KILL 2010-12-05 01:27:13.319023 [DEBUG] switch_core_session.c:1083 Send signal loopback/18884447693-b [BREAK] 2010-12-05 01:27:13.319023 [DEBUG] mod_loopback.c:468 loopback/18884447693-b CHANNEL KILL 2010-12-05 01:27:13.320024 [DEBUG] switch_core_session.c:1993 loopback/18884447693-b skip receive message [APPLICATION_EXEC_COMPLETE] (chann el is hungup already) 2010-12-05 01:27:13.320024 [DEBUG] switch_core_state_machine.c:366 (loopback/18884447693-b) State EXECUTE going to sleep 2010-12-05 01:27:13.320024 [DEBUG] switch_core_state_machine.c:320 (loopback/18884447693-b) Running State Change CS_HANGUP 2010-12-05 01:27:13.321011 [DEBUG] switch_core_state_machine.c:557 (loopback/18884447693-b) State HANGUP 2010-12-05 01:27:13.321983 [DEBUG] mod_loopback.c:414 loopback/18884447693-b CHANNEL HANGUP 2010-12-05 01:27:13.321983 [DEBUG] switch_channel.c:2455 (loopback/18884447693-a) Callstate Change EARLY -> HANGUP 2010-12-05 01:27:13.321983 [NOTICE] mod_loopback.c:425 Hangup loopback/18884447693-a [CS_CONSUME_MEDIA] [NORMAL_CIRCUIT_CONGESTION] 2010-12-05 01:27:13.322995 [DEBUG] switch_ivr_originate.c:3448 Originate Resulted in Error Cause: 34 [NORMAL_CIRCUIT_CONGESTION] ----8<----8<----8<----8<----8<----8<----8<---- EXECUTE loopback/18884447701-b hangup(NORMAL_CIRCUIT_CONGESTION) 2010-12-05 01:27:14.364448 [DEBUG] switch_channel.c:2455 (loopback/18884447701-b) Callstate Change EARLY -> HANGUP 2010-12-05 01:27:14.365470 [NOTICE] mod_dptools.c:906 Hangup loopback/18884447701-b [CS_EXECUTE] [NORMAL_CIRCUIT_CONGESTION] 2010-12-05 01:27:14.365470 [DEBUG] switch_channel.c:2471 Send signal loopback/18884447701-b [KILL] 2010-12-05 01:27:14.365470 [DEBUG] mod_loopback.c:468 loopback/18884447701-b CHANNEL KILL 2010-12-05 01:27:14.365470 [DEBUG] switch_core_session.c:1083 Send signal loopback/18884447701-b [BREAK] 2010-12-05 01:27:14.366485 [DEBUG] mod_loopback.c:468 loopback/18884447701-b CHANNEL KILL 2010-12-05 01:27:14.366485 [DEBUG] switch_core_session.c:1993 loopback/18884447701-b skip receive message [APPLICATION_EXEC_COMPLETE] (chann el is hungup already) 2010-12-05 01:27:14.366485 [DEBUG] switch_core_state_machine.c:366 (loopback/18884447701-b) State EXECUTE going to sleep 2010-12-05 01:27:14.367508 [DEBUG] switch_core_state_machine.c:320 (loopback/18884447701-b) Running State Change CS_HANGUP 2010-12-05 01:27:14.369585 [DEBUG] switch_core_state_machine.c:557 (loopback/18884447701-b) State HANGUP 2010-12-05 01:27:14.369585 [DEBUG] mod_loopback.c:414 loopback/18884447701-b CHANNEL HANGUP 2010-12-05 01:27:14.369585 [DEBUG] switch_channel.c:2455 (loopback/18884447701-a) Callstate Change EARLY -> HANGUP 2010-12-05 01:27:14.369585 [DEBUG] switch_ivr_originate.c:3448 Originate Resulted in Error Cause: 19 [NO_ANSWER] 2010-12-05 01:27:14.369585 [NOTICE] mod_loopback.c:425 Hangup loopback/18884447701-a [CS_CONSUME_MEDIA] [NORMAL_CIRCUIT_CONGESTION] ----8<----8<----8<----8<----8<----8<----8<---- <context name="tech-outbound"> <extension name="tech-outbound"> <condition field="destination_number" expression="^1888(\d{7})$"> <action application="info"/> <action application="set" data="loopback_bowout=false"/> <action application="limit" data="db outgoing goip 1" /> <action application="bridge" data="{ignore_early_media=true}sofia/internal/${destination_number}(a)192.168. 3.11:5060"/> </condition> </extension> <extension name="limit_exceeded"> <condition field="destination_number" expression="^limit_exceeded$"> <action application="<nothing> OR <answer> OR <pre_answer>"/>  <action application="hangup" data="NORMAL_CIRCUIT_CONGESTION"/> </condition> </extension> </context>

1 0

GIT/SVN Repo Maint.
by Brian West 04 Dec '10

04 Dec '10

I'm going to be doing some work on the repo to move it this weekend so please be patient if you can't access it while I'm moving it to Dallas Sunday December 5th 2010. Thanks, Brian

1 0

building mod_managed fails
by babak yakhchali 04 Dec '10

04 Dec '10

Hi I'm trying to build mod_managed using mono on centos 5.5. I'm building mono from source and when I try with versions above 2.6 it complains about glib.h and with mono 2.4 it results in errors: error CS1110: `System.Linq.EnumerableExtensions.ForEach<T>(this System.Collections.Generic.IEnumerable<T>, System.Action<T>)': Extension methods cannot be declared without a reference to System.Core.dll assembly. Add the assembly reference or remove `this' modifer from the first parameter Loader.cs(110,16): error CS0246: The type or namespace name `HashSet`1' could not be found. Are you missing a using directive or an assembly reference? Log.cs(57,23): error CS1110: `FreeSWITCH.Log.ToLogString(this FreeSWITCH.LogLevel)': Extension methods cannot be declared without a reference to System.Core.dll assembly. Add the assembly reference or remove `this' modifer from the first parameter ManagedSession.cs(75,16): error CS0246: The type or namespace name `Action' could not be found. Are you missing a using directive or an assembly reference? ManagedSession.cs(89,16): error CS0246: The type or namespace name `Func`3' could not be found. Are you missing a using directive or an assembly reference? ManagedSession.cs(91,16): error CS0246: The type or namespace name `Func`2' could not be found. Are you missing a using directive or an assembly reference? PluginManager.cs(76,9): error CS0246: The type or namespace name `Action' could not be found. Are you missing a using directive or an assembly reference? PluginManager.cs(77,44): error CS0246: The type or namespace name `Action' could not be found. Are you missing a using directive or an assembly reference? PluginManager.cs(90,18): error CS0246: The type or namespace name `Func`1' could not be found. Are you missing a using directive or an assembly reference? PluginManager.cs(92,69): error CS0246: The type or namespace name `Func`1' could not be found. Are you missing a using directive or an assembly reference? PluginManager.cs(122,18): error CS0246: The type or namespace name `Func`1' could not be found. Are you missing a using directive or an assembly reference? PluginManager.cs(124,69): error CS0246: The type or namespace name `Func`1' could not be found. Are you missing a using directive or an assembly reference? PluginManager.cs(262,23): error CS0246: The type or namespace name `Func`1' could not be found. Are you missing a using directive or an assembly reference? ScriptPluginManager.cs(236,16): error CS0246: The type or namespace name `Action' could not be found. Are you missing a using directive or an assembly reference? ScriptPluginManager.cs(187,22): error CS0246: The type or namespace name `Action' could not be found. Are you missing a using directive or an assembly reference? ScriptPluginManager.cs(188,37): error CS0246: The type or namespace name `Action' could not be found. Are you missing a using directive or an assembly reference? ScriptPluginManager.cs(218,22): error CS0246: The type or namespace name `Action' could not be found. Are you missing a using directive or an assembly reference? ScriptPluginManager.cs(219,37): error CS0246: The type or namespace name `Action' could not be found. Are you missing a using directive or an assembly reference? Util.cs(42,25): error CS1110: `FreeSWITCH.FSUtil.CreateSwigTypePointer<T>(this System.IntPtr)': Extension methods cannot be declared without a reference to System.Core.dll assembly. Add the assembly reference or remove `this' modifer from the first parameter XmlSearchBinding.cs(81,40): error CS0246: The type or namespace name `Func`2' could not be found. Are you missing a using directive or an assembly reference? Compilation failed: 20 error(s), 0 warnings make[5]: *** [FreeSWITCH.Managed.dll] Error 1 thanks

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