November 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

mod_dingaling for Google Voice - delay issues?
by Avi Marcus 19 Nov '10

19 Nov '10

Hi, has anyone else experienced about 2.5 second delays when using mod_dingaling for google voice? I used my gizmo callback for google voice, though the same freeswitch server to an echo test and noticed about 1/2 a second delay. (I am in Israel, that could explain it.) But when using mod_dingaling, it's 2.5 seconds, so it's not a generic google voice issue. Either it's a mod_dingaling issue, or a google jingle issue. Any ideas on how to fix this? Thanks, Avi

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sofia: call does not exist
by Christian Löschenkohl 19 Nov '10

19 Nov '10

hello the sofia stack responds to an outgoing invite with this message. the problem seems to be the following behavior from the endpoint (ip pbx) - 1st invite To: <sip:0123456789@sipserver> - server sends Proxy Authentication Required + nonce - pbx sends To: <sip:0123456789@sipserver>;tag=Sp1m3ayr748rH so the two sip dialoges do not match and the server respons with "Call Does Not Exist". is there a way to handle this (maybe broken) behavior from this sip client. possibly per user with an NDLB config option br -- Ing. Christian Löschenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl(a)xpirio.com

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Receiving calls from incoming DID's
by Neil J. Macvicar 19 Nov '10

19 Nov '10

Hello, I have recently compiled and installed freeswitch, and it is running with no problems. I can make and receive calls via my defined gateway with my provider, VoipTalk. My problem lies in that I have several DIDs with Voiptalk, and I have the option of routing them directly to my IP address as a SIP call. When I do this and call one of the numbers, not much happens. So I put sofia into debug sip mode, and observed the same message repeating :- ------------------------------------------------------------------------ send 866 bytes to udp/[217.14.138.126]:5060 at 22:22:15.236893: ------------------------------------------------------------------------ SIP/2.0 407 Proxy Authentication Required 217.14.138.126 being one of my providers' machines. What i'd like to know is how can I get freeswitch to stop asking for authentication to my provider, without putting a blacket "allow" in the domain ACLS. Voiptalk have a large number of IP addresses that they use to send the outbound calls, so listing them all would be a chore. Thanks in advance, --Neil.

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intercept call with pin
by budi wibowo 19 Nov '10

19 Nov '10

hi any idea to intercept outgoing call with pin? before or after press dialed digit, ivr will be played and ask for pin. i already googling but most of pin usage is for conference. thx and regards budi

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IVR menu failure/timeout action
by Raymond Chan 19 Nov '10

19 Nov '10

Hi all, I am experiencing an IVR problem. Except the top menu, all other sub menu will not hang up after reached max invalid input or timeout limit. It will go back up upper menu. I want it to load the exit-sound and then hang the call when it reached max invalid input or timeout limit. Do you tell how to configure? <menu name="m800_ivr_submenu-m_ch" greet-long="C:/FreeSWITCH/recordings/800-ivr-2nd-m.wav" greet-short="C:/FreeSWITCH/recordings/greet-short-m.wav" invalid-sound="ivr/ivr-that_was_an_invalid_entry-m.wav" exit-sound="voicemail/vm-goodbye-m.wav" timeout="5000" max-failures="2"> <entry action="menu-exec-app" digits="1" param="playback C:/FreeSWITCH/recordings/800-ivr-announcement-m_ch.wav"/> <entry action="menu-exec-app" digits="2" param="lua endcall-m.lua"/> </menu> Thanks Raymond

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error while loading mod_spidermonkey
by ovvenkat 19 Nov '10

19 Nov '10

Hi , I am getting following error while loading mod_spidermonkey freeswitch@internal> load mod_spidermonkey +OK Reloading XML -ERR [module load file routine returned an error] 2010-11-19 13:32:04.754650 [INFO] mod_pocketsphinx.c:482 PocketSphinx Reloaded freeswitch@internal> 2010-11-19 13:32:04.754650 [INFO] mod_enum.c:808 ENUM Reloaded 2010-11-19 13:32:04.754650 [INFO] switch_time.c:950 Timezone reloaded 530 definitions 2010-11-19 13:32:04.755651 [CRIT] switch_loadable_module.c:928 Error Loading module /usr/local/freeswitch/mod/mod_spidermonkey.so **/usr/lib/libnssutil3.so: undefined symbol: PR_GetDirectorySeparator** -- Regards Venkatesan OV.

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problem with voicemail
by joy this 19 Nov '10

19 Nov '10

Dear all: I am confused with voicemail for several days. I hope someone could help me please. I would like to send a voicemail with audio attachment by my application. My FS is Version 1.0.head, on Windows XP. The following are my configurations: switch.conf.xml: <param name="mailer-app" value="C:\FreeSWITCH\Mailer.exe"/> <param name="mailer-app-args" value="MyArgs -t"/> \directory\default\1001.xml <param name="vm-password" value="1001"/> <param name="vm-mailto" value="thisjoy(a)gmail.com" /> <param name="vm-email-all-messages" value="true"/> <param name="vm-attach-file" value="true" /> Other settings are as default. I register with 1000 and call 1001 (not registered) for sending a voicemail. When I left a message, my Mailer.exe only receives the arguments: -f 1001(a)192.168.1.1 MyArgs –t thisjoy(a)gmail.com. FS shows "The process tried to write to a nonexistent pipe", and the mail actually sends to me without the attachment (I have test my Mailer.exe, it could attach a file without FS). The audio file exists in "C:\FreeSWITCH\storage\voicemail\default\192.168.1.1\1001". I saw “2010-11-12 18:50:18.171875 [DEBUG] switch_utils.c:631 Emailed file [C:\DOCUME~1\123\LOCALS~1\Temp\mail.12895590120029] to [thisjoy(a)gmail.xn--com]-jb7a in the log file, but the file “mail.12895590120029” never appears. I found that the directory “C:\DOCUME~1\123\LOCALS~1\Temp” is as the sqlite temporary path in the “freeswitch\libs\sqlite\www\pragma.tcl”, although I do not know if this causes the mail failure. Could anyone teach me how to attach the audio file with the mail? Sincerely yours, thisjoy.

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outgoing dialplan
by budi wibowo 19 Nov '10

19 Nov '10

hi i have 2 routes for terminating usa raffic 1. using galk 2. using my softswitch in my dialplan i define gtalk and my softswitch for outgoing call, the problem is if i failed with gtalk, it can not route to softswitch. any config to use alternate route? thx budi

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Not able to compile ESL php module
by ovvenkat 18 Nov '10

18 Nov '10

Hi , I have all the php requisites tools . When I am trying to compile ESL with php module I am getting error make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x" CFLAGS="-I/usr/local/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes" CXXFLAGS="-I/usr/local/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable" CXX_CFLAGS="" -C php make[1]: Entering directory `/usr/local/src/freeswitch/libs/esl/php' g++ -I/usr/local/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable -I/usr/include/php -I/usr/include/php/main -I/usr/include/php/TSRM -I/usr/include/php/Zend -I/usr/include/php/ext -I/usr/include/php/ext/date/lib -Wno-unused-label -Wno-unused-function -c esl_wrap.cpp -o esl_wrap.o cc1plus: warnings being treated as errors esl_wrap.cpp:2583: error: deprecated conversion from string constant to ‘char*’ make[1]: *** [esl_wrap.o] Error 1 make[1]: Leaving directory `/usr/local/src/freeswitch/libs/esl/php' make: *** [phpmod] Error 2 I am using FC - 13 and Latest Git version of FreeSwitch. Any one plz tell me, why I am getting this error. -- If you have come to help me, you are wasting your time. If you have come to because your liberation is bound up in mine, we can work together. Regards Venkatesan OV.

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WG: fusionPBX / Dialplans get crazy
by Isti 18 Nov '10

18 Nov '10

Hi, hier is an outbound call, which was sent through the provider "PBX-Network" The number was 0049-721-6019044 ---------------------------------------------------------------------------- Dialplan: sofia/internal/120(a)188.72.247.73 Regex (PASS) [pbxnetwork.0049151152157159160162163170172172173174175176177178179d] destination_number(00497216019044) =~ /^(\+|00)(49[151|152|157|159|160|162|163|170|172|172|173|174|175|176|177|178 |179])(\d+)/ break=on-false Dialplan: sofia/internal/120(a)188.72.247.73 Action set(index=00) Dialplan: sofia/internal/120(a)188.72.247.73 Action set(vorwahl=497) Dialplan: sofia/internal/120(a)188.72.247.73 Action set(nummer=216019044) Dialplan: sofia/internal/120(a)188.72.247.73 Action bridge(sofia/gateway/pbxnetwork/00497216019044) 2010-11-16 01:19:37.604828 [DEBUG] switch_core_state_machine.c:114 (sofia/internal/120(a)188.72.247.73) State Change CS_ROUTING -> CS_EXECUTE 2010-11-16 01:19:37.604828 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/120(a)188.72.247.73 [BREAK] 2010-11-16 01:19:37.604828 [DEBUG] switch_core_state_machine.c:484 (sofia/internal/120(a)188.72.247.73) State ROUTING going to sleep 2010-11-16 01:19:37.604828 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/120(a)188.72.247.73) Running State Change CS_EXECUTE 2010-11-16 01:19:37.604828 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/120(a)188.72.247.73) State EXECUTE 2010-11-16 01:19:37.604828 [DEBUG] mod_sofia.c:173 sofia/internal/120(a)188.72.247.73 SOFIA EXECUTE 2010-11-16 01:19:37.604828 [DEBUG] switch_core_state_machine.c:151 sofia/internal/120(a)188.72.247.73 Standard EXECUTE EXECUTE sofia/internal/120(a)188.72.247.73 set(use_profile=default) 2010-11-16 01:19:37.604828 [DEBUG] mod_dptools.c:748 sofia/internal/120(a)188.72.247.73 SET [use_profile]=[default] EXECUTE sofia/internal/120(a)188.72.247.73 set_user(default(a)188.72.247.73) EXECUTE sofia/internal/120(a)188.72.247.73 db(insert/188.72.247.73-spymap/01795191600/8b4f07ee-37b6-4849-adb8-b9dada887 83e) EXECUTE sofia/internal/120(a)188.72.247.73 db(insert/188.72.247.73-last_dial/01795191600/00497216019044) EXECUTE sofia/internal/120(a)188.72.247.73 db(insert/188.72.247.73-last_dial/global/8b4f07ee-37b6-4849-adb8-b9dada88783 e) EXECUTE sofia/internal/120(a)188.72.247.73 set(index=00) 2010-11-16 01:19:37.686045 [DEBUG] mod_dptools.c:748 sofia/internal/120(a)188.72.247.73 SET [index]=[00] EXECUTE sofia/internal/120(a)188.72.247.73 set(vorwahl=497) 2010-11-16 01:19:37.686045 [DEBUG] mod_dptools.c:748 sofia/internal/120(a)188.72.247.73 SET [vorwahl]=[497] EXECUTE sofia/internal/120(a)188.72.247.73 set(nummer=216019044) 2010-11-16 01:19:37.686045 [DEBUG] mod_dptools.c:748 sofia/internal/120(a)188.72.247.73 SET [nummer]=[216019044] EXECUTE sofia/internal/120(a)188.72.247.73 bridge(sofia/gateway/pbxnetwork/00497216019044) ---------------------------------------------------------------------------- This is actually false. Hier are the dialplans: ------------------------------------------ Dcalling.0036203050607033600300901030406030659078091230334541495455567567898 941207074767778799797986078860798689d 996 true ^(\+|00)(36[20|30|50|60|70|]|336[003|009|01|03|04|0603|0659|078|09|12|303|34 5|41|49|54|55|567|56|78|989]|41[20|70|74|76|77|78|79979|79|86078|86079|86|89 ])(\d+) $ Dcalling pbxnetwork.0049151152157159160162163170172172173174175176177178179d 997 true ^(\+|00)(49[151|152|157|159|160|162|163|170|172|172|173|174|175|176|177|178| 179])(\d+)$ pbxnetwork pbxnetwork.0151152157159160162163170172172173174175176177178179d 998 true ^(0[151|152|157|159|160|162|163|170|172|172|173|174|175|176|177|178|179])(\d +)$ pbxnetwork Sipgate_2.d 999 true ^(\d+) $ Sipgate_2 ------------------------------------------------------- 1. The first provider Dcalling filters the international mobile calls. 2. the 2nd (PBX-Network) filters the german mobile calls which start with "+" or "00", in the internation format. 3. the 3rd (PBX-Network) filters the german mobile calls which start with "0", in the national format. 4. the 4th one, Sipgate catchs all the rest. In this case 0049-721-601-9044 Sipgate have had to cought the call. Because no otfer filter matches. According to the log PBX-Network was activated. But why? Also Dcalling handles the number with 00. and no other filter handles the area code "721", only Sipgate. Why PBX-Network? Can you tell what is wrong. As the dial plans were set up was everything Ok. I have then deleted a dialplan and everything went wrong. In case you have an idea what to do, thank you very much for it. Could you tell me how to set up the dialplans to get "cleaner" forwarding. With kind regards, I.

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