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Re: [Freeswitch-users] IP SIP Trunk errors/issues
by Syed Hussain 29 Jun '10

29 Jun '10

I have tried the options below with no luck. David will tcpdump output suffice ? please advise thanks S Re: [Freeswitch-users] IP SIP Trunk errors/issues Monday, June 28, 2010 8:12 AM From: "Zuhair Raza" <engineerzuhairraza(a)gmail.com> To: freeswitch-users(a)lists.freeswitch.orghi try this one <condition field="network_addr" expression="^x\.x\.x\.x$"> <anti-action application="reject" data="407"/> On Mon, Jun 28, 2010 at 4:16 PM, Syed Hussain <uzairsh(a)yahoo.com> wrote: Hi , I have subscribed to sonovoip a sip trunk provider who uses IP based authentication for SIP services , I am trying to make outbound international calls which are being rejected by the provider for some reason, I am not able to figure out if there is something wrong with the configuration. I am listing out my configuration. Any help would be greatly appreciated . I have also tried #wiki.freeswitch.org/wiki/Provider_Configuration:_SonoVoIP configuration from freeswitch wiki gives errors . The voip configuration works perfectly with asterisk 1.6 My configuration is follows Under /usr/local/freeswitch/conf/sip_profiles/external cat sonovoip.xml <include> <gateway name="sonovoip"> <param name="username" value="anything"/> <param name="password" value="anything"/> <param name="realm" value="x.x.x.x"/>  <param name="register" value="false"/> <param name="caller-id-in-from" value="true"/> <param name="extension-in-contact" value="true"/> </gateway> </include> My dialplan , under /usr/local/freeswitch/conf/dialplan/public.xml <extension name="sonovoip" continue="true"> <condition field="destination_number" expression="^(00.*)$"> <action application="bridge" data="sofia/gateway/sonovoip/$1"/> </condition> </extension> I did change the vars.xml ports  <X-PRE-PROCESS cmd="set" data="internal_auth_calls=true"/> <X-PRE-PROCESS cmd="set" data="internal_sip_port=5080"/> <X-PRE-PROCESS cmd="set" data="internal_tls_port=5081"/> <X-PRE-PROCESS cmd="set" data="internal_ssl_enable=false"/> <X-PRE-PROCESS cmd="set" data="internal_ssl_dir=$${base_dir}/conf/ssl"/>  <X-PRE-PROCESS cmd="set" data="external_auth_calls=false"/> <X-PRE-PROCESS cmd="set" data="external_sip_port=5060"/> <X-PRE-PROCESS cmd="set" data="external_tls_port=5061"/> <X-PRE-PROCESS cmd="set" data="external_ssl_enable=false"/> <X-PRE-PROCESS cmd="set" data="external_ssl_dir=$${base_dir}/conf/ssl"/> The debug #pastebin.com/4MFhSWm9 Please advise if any other information is required. Thanks S _______________________________________________ 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 -- Regards, Zuhair Raza

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Re: [Freeswitch-users] FreeTDM and DAHDI+OSLEC/MG2 anyone?
by Jeroen C. van Gelderen 29 Jun '10

29 Jun '10

Hi Mark, I'm curious to see if you'd had any luck with DAHDI built-in echo cancellers such as MG2? (I figured since MG2 has been built into DAHDI for a while it would be slightly more likely to work than OSLEC. In my case neither MG2 nor OSLEC works.) In order to determine whether echo cancellation is in use with DAHDI you can check the output of lsdahdi: [root@elastix ~]# lsdahdi ### Span 1: XBUS-00/XPD-00 "Xorcom XPD #00/00: FXS" (MASTER) 1 FXS FXOKS (In use) (SWEC: OSLEC) (EC: OSLEC) 2 FXS FXOKS (In use) (SWEC: OSLEC) This shows FXS port 1 in off-hook with OSLEC in-use. FXS port 2 has OSLEC configured but is on-hook. If I were to run FreeSwitch the channel would be silent in this configuration. HTH, -Slim -- Jeroen C. "Slim" van Gelderen Olympic Sports Data Services Email: jeroeng(a)thegreek.com Phone: +1 876 953 6182 x128 -----Original Message----- From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Mark Maly Sent: Monday, June 28, 2010 13:45 To: freeswitch-users(a)lists.freeswitch.org Subject: Re: [Freeswitch-users] FreeTDM and DAHDI+OSLEC/MG2 anyone? I'm attempting to use Freeswitch/DAHDHI and OSLEC, as well. I agree in commenting out the "echocancellation" line in zt.conf. The channels seem to work OK but the echo cancellation does not seem to work (I can see the kernel drivers for it, but I still have a great deal of echo). Mark -----Original Message----- From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Jeroen C. van Gelderen Sent: Saturday, June 26, 2010 7:57 PM To: freeswitch-users(a)lists.freeswitch.org Subject: [Freeswitch-users] FreeTDM and DAHDI+OSLEC/MG2 anyone? Hi everybody, I have a feeling I must be missing the obvious... I've been trying to get echo canceling to work with Freeswitch/FreeTDM/DAHDI but I have been failing miserably for days. When I enable echo cancellation (MG2 or OSLEC) on a DAHDI/FreeTDM channel (FXS or FXO) that channel goes silent. In absence of better suggestions I was hoping to replicate someone's "known working" configuration. Is anyone successfully running the following combination: - any x86 hardware - any recent flavor Linux - any recent version of DAHDI using FXO ports - FreeSwitch/FreeTDM trunk from git - OSLEC echo canceller (or MG2) - (BONUS:) Xorcom Astribank with FXO/FXS ports. I addition to much Googling I've tried most permutations of: - CentOS 5.4, 5.5, Ubuntu 10.04 LTS, Elastix - x86 and x64 single and multicore CPUs. - Non-PAE kernels for 32-bit installs. - DADHI from SVN (or -in case of Elastix- the built-in 2.2.0.2) - MG2 and OSLEC echo cancellers - Freeswitch from Git trunk In each case everything configures fine to the point that Asterisk 1.6 will function with echo cancellation enabled. So we know that DAHDI layer works. But Freeswitch channels go silent when echo cancellation is enabled. Puzzled, -Slim -- Jeroen C. "Slim" van Gelderen _______________________________________________ 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 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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ESL install error
by Steve O 29 Jun '10

29 Jun '10

Hello all, I'm trying to add phpmod-install into my Freeswitch installation on 2 different servers (was failing on the first server, so trying another). Here's a snippet of the error I'm encountering on both servers: g++ -I/usr/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable -I/usr/include/php5 -I/usr/include/php5/main -I/usr/include/php5/TSRM -I/usr/include/php5/Zend -I/usr/include/php5/ext -I/usr/include/php5/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: In function ‘void _wrap_ESLevent_event_set(int, zval*, zval**, zval*, int)’: esl_wrap.cpp:1047: error: format not a string literal and no format arguments esl_wrap.cpp: In function ‘void _wrap_ESLevent_event_get(int, zval*, zval**, zval*, int)’: esl_wrap.cpp:1073: error: format not a string literal and no format arguments esl_wrap.cpp: In function ‘void _wrap_ESLevent_serialized_string_set(int, zval*, zval**, zval*, int)’: . . . make[1]: *** [esl_wrap.o] Error 1 make[1]: Leaving directory `/usr/src/freeswitch/libs/esl/php' make: *** [phpmod] Error 2 I searched the list for posts related to the [esl_wrap.o] Error 1, followed all the advice that helped others, but am still having the problem. A couple things about my situation I'm considering that might be unique, and the problem (pretty much guessing at this point), is the OS and architectures I'm running. One server's (the 1st server, referenced above) running FreeBSD 8.0 on a 32 bit processor, and the other server's running Ubuntu 10.04 on a 64 bit processor. Thoughts? What other information can I provide to help troubleshoot? Thanks, Steve

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FreeTDM and DAHDI+OSLEC/MG2 anyone?
by Jeroen C. van Gelderen 29 Jun '10

29 Jun '10

Hi everybody, I have a feeling I must be missing the obvious... I've been trying to get echo canceling to work with Freeswitch/FreeTDM/DAHDI but I have been failing miserably for days. When I enable echo cancellation (MG2 or OSLEC) on a DAHDI/FreeTDM channel (FXS or FXO) that channel goes silent. In absence of better suggestions I was hoping to replicate someone's "known working" configuration. Is anyone successfully running the following combination: - any x86 hardware - any recent flavor Linux - any recent version of DAHDI using FXO ports - FreeSwitch/FreeTDM trunk from git - OSLEC echo canceller (or MG2) - (BONUS:) Xorcom Astribank with FXO/FXS ports. I addition to much Googling I've tried most permutations of: - CentOS 5.4, 5.5, Ubuntu 10.04 LTS, Elastix - x86 and x64 single and multicore CPUs. - Non-PAE kernels for 32-bit installs. - DADHI from SVN (or -in case of Elastix- the built-in 2.2.0.2) - MG2 and OSLEC echo cancellers - Freeswitch from Git trunk In each case everything configures fine to the point that Asterisk 1.6 will function with echo cancellation enabled. So we know that DAHDI layer works. But Freeswitch channels go silent when echo cancellation is enabled. Puzzled, -Slim -- Jeroen C. "Slim" van Gelderen

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How to Access To Bridged Seesion
by afshin afzali 28 Jun '10

28 Jun '10

Hi FreeSWITCH, What is the preferred method for obtaining bridged session from the current one? I've just found the signal_bond variable. BEST, -- afshin

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Question about Lua in diaplan
by DJB International 28 Jun '10

28 Jun '10

I was trying to make lua to bridge in dialplan, but somehow it did not behave like XML dialplan: If I have this in XML: <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="bridge" data="sofia/gateway/1.2.3.4/$1" /> <action application="bridge" data="sofia/gateway/5.6.7.8/$1" /> then, the second action will not be called if the first one hangs up the channel unless the first action failed. But, if I have this in Lua: session:execute("set","hangup_after_bridge=true") session:execute("set","continue_on_fail=true") for k, v in pairs(split(dst_gw_ip_list, ';')) do session:execute("bridge","sofia/gateway/" .. v .. "/" .. dest_out .. "") end then, the second action is also called. How do I make the lua script to behave like XML? Thank you, Dorn B.

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IP SIP Trunk errors/issues
by Syed Hussain 28 Jun '10

28 Jun '10

Hi , I have subscribed to sonovoip a sip trunk provider who uses IP based authentication for SIP services , I am trying to make outbound international calls which are being rejected by the provider for some reason, I am not able to figure out if there is something wrong with the configuration. I am listing out my configuration. Any help would be greatly appreciated . I have also tried #wiki.freeswitch.org/wiki/Provider_Configuration:_SonoVoIP configuration from freeswitch wiki gives errors . The voip configuration works perfectly with asterisk 1.6 My configuration is follows Under /usr/local/freeswitch/conf/sip_profiles/external cat sonovoip.xml <include> <gateway name="sonovoip"> <param name="username" value="anything"/> <param name="password" value="anything"/> <param name="realm" value="x.x.x.x"/>  <param name="register" value="false"/> <param name="caller-id-in-from" value="true"/> <param name="extension-in-contact" value="true"/> </gateway> </include> My dialplan , under /usr/local/freeswitch/conf/dialplan/public.xml <extension name="sonovoip" continue="true"> <condition field="destination_number" expression="^(00.*)$"> <action application="bridge" data="sofia/gateway/sonovoip/$1"/> </condition> </extension> I did change the vars.xml ports  <X-PRE-PROCESS cmd="set" data="internal_auth_calls=true"/> <X-PRE-PROCESS cmd="set" data="internal_sip_port=5080"/> <X-PRE-PROCESS cmd="set" data="internal_tls_port=5081"/> <X-PRE-PROCESS cmd="set" data="internal_ssl_enable=false"/> <X-PRE-PROCESS cmd="set" data="internal_ssl_dir=$${base_dir}/conf/ssl"/>  <X-PRE-PROCESS cmd="set" data="external_auth_calls=false"/> <X-PRE-PROCESS cmd="set" data="external_sip_port=5060"/> <X-PRE-PROCESS cmd="set" data="external_tls_port=5061"/> <X-PRE-PROCESS cmd="set" data="external_ssl_enable=false"/> <X-PRE-PROCESS cmd="set" data="external_ssl_dir=$${base_dir}/conf/ssl"/> The debug #pastebin.com/4MFhSWm9 Please advise if any other information is required. Thanks S

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Javascript: inconsistent session.ready() result when originating a call
by Marc Kellem 28 Jun '10

28 Jun '10

I'm using Javascript to originate a new outbound call. The Session ready() method returns inconsistent results when the call is answered. When it returns false, the session's causecode value is 0. Am I doing something wrong here, or could it be a bug? A simple test script and log output is below. I'm executing the script from the console using jsrun. I've noticed that switch_core_state_machine.c logs a state change when ready() returns true. I'm not sure if this is significant, but it might indicate a race condition. ---- test script ---- new_session = new Session( "{ignore_early_media=true}sofia/internal/1000%192.168.1.10" ); if ( new_session.ready() ) { console_log( "info", "Call connected.\n" ); // play audio.... } else { // the causecode is 0 when this fails on a successful call attempt console_log("err", "Call not connected. Cause: " + new_session.cause + "[" + new_session.causecode + "]\n" ); } ---- call connected log ---- [NOTICE] sofia.c:4851 Channel [sofia/internal/1000] has been answered [DEBUG] switch_ivr_originate.c:3273 Originate Resulted in Success: [sofia/internal/1000] [DEBUG] mod_spidermonkey.c:2866 (sofia/internal/1000) State Change CS_CONSUME_MEDIA -> CS_SOFT_EXECUTE [DEBUG] switch_core_session.c:1027 Send signal sofia/internal/1000 [BREAK] [DEBUG] switch_core_state_machine.c:314 (sofia/internal/1000) Running State Change CS_SOFT_EXECUTE [INFO] originate.js:1 Call connected. [DEBUG] switch_core_state_machine.c:354 (sofia/internal/1000) State SOFT_EXECUTE [DEBUG] mod_sofia.c:544 SOFIA SOFT_EXECUTE [DEBUG] switch_core_state_machine.c:200 sofia/internal/1000 Standard SOFT_EXECUTE [DEBUG] switch_core_state_machine.c:354 (sofia/internal/1000) State SOFT_EXECUTE going to sleep ---- call not connected log ---- [NOTICE] sofia.c:4851 Channel [sofia/internal/1000] has been answered [DEBUG] switch_ivr_originate.c:3273 Originate Resulted in Success: [sofia/internal/1000] [DEBUG] mod_spidermonkey.c:2866 (sofia/internal/1000) State Change CS_CONSUME_MEDIA -> CS_SOFT_EXECUTE [DEBUG] switch_core_session.c:1027 Send signal sofia/internal/1000 [BREAK] [ERR] originate.js:1 Call not connected. Cause: NONE(0) [DEBUG] switch_channel.c:2261 (sofia/internal/1000) Callstate Change ACTIVE -> HANGUP [NOTICE] mod_spidermonkey.c:3066 Hangup sofia/internal/1000 [CS_SOFT_EXECUTE] [NORMAL_CLEARING]

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Use lua to replace a .wav file
by Raymond Chan 28 Jun '10

28 Jun '10

Hi, I am using freeswitch as call announcement purpose. Admin user can call in and record a sound clip and then updated the announcement sound clip. I created IVR menu and use the recording lua to record the sound clip (to record as temp...uuid...wav file). It is ok. However, I do not know how to replace the announcement sound clip after the user press a key in ivr to confirm update. I try to use lua to do it, but it seems that it do not have file replacement function. Am I correct? Thanks Raymond

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Invalid Application hash
by Lars Zeb 28 Jun '10

28 Jun '10

Recently I've been getting an Invalid Application hash error. Any ideas on what might be causing this. I have not changed anything in the configuration for a long while. I am using git-c58efca 2010-06-24 11-52-49 -0500 on Centos 5.3. Pastebin is http://pastebin.freeswitch.org/13255. Thanks Lars

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