- FreeSWITCH-users - FreeSWITCH Mailing Lists

Routing messages in mod_sms chatplan
by Yuriy Ostapchuk 05 Feb '13

05 Feb '13

Hi, I have a setup where I need to route incoming sms messages from mod_gsmopen to different sip or dingaling extensions, depending on sender's phone number and other conditions. Is there a way to do this directly in chatplan without executing an external script? Basically, I'm looking for a functionality similar to 'transfer' application in dialplan. Thanks.

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building mod_opal and v1.2stable
by Gilbert Amar 05 Feb '13

05 Feb '13

Hi, I am trying to build mod_opal in FS v1.2stable, using ptlib 2.10.7 and opal 3.10.7 on a debian Ptlib and opal get build with no error, but when trying mod_opal I get Compiling /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp... quiet_libtool: compile: g++ -g -ggdb -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -I/usr/local/src/freeswitch/libs/stfu -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -DP_64BIT -DPTRACING=1 -D_REENTRANT -D_GNU_SOURCE=1 -fno-exceptions -I/usr/local/include -I/usr/local/include/opal -I/usr/include/SDL -DHAVE_CONFIG_H -c /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp -fPIC -DPIC -o .libs/mod_opal.o /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp: In member function 'switch_status_t FSConnection::read_frame(const OpalMediaType&, switch_frame_t**, switch_io_flag_t)': /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp:1208: error: 'class OpalCall' has no member named 'IsSwitchingT38' /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp: In member function 'switch_status_t FSConnection::write_frame(const OpalMediaType&, const switch_frame_t*, switch_io_flag_t)': /usr/local/src/freeswitch/src/mod/endpoints/mod_opal/mod_opal.cpp:1225: error: 'class OpalCall' has no member named 'IsSwitchingT38' make[3]: *** [mod_opal.lo] Error 1 make[2]: *** [install] Error 1 make[1]: *** [mod_opal-install] Error 1 make: *** [mod_opal-install] Error 2 Has anyone succeded in building mod_opal ? Should I go with mod_h323 instead ? Thanks for your help

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spandsp_start_fax_detect how to use?
by George Assaad 05 Feb '13

05 Feb '13

Hello, I have one DID that I would like to detect the tone 1- if fax goes to extension 1111. 2- If it is is not fax then t goes to extension 2222 How do that using spandsp_start_fax_detect ? I am looking for the Inbound dialplan Thanks in advance, George

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valet park
by Darcy Primrose 05 Feb '13

05 Feb '13

When I place a call in valet apark, and then retrieve it, it stays bridges to the valet_slot Dial plan: <extension name="valet_park_me_2"> <condition field="destination_number" expression="^(6002)$"> <action application="answer"/> <action application="set" data="domain_name=192.168.209.49"/> <action application="valet_park" data="valet_parking_lot 6002"/> <action inline="true" application="set" data="presence_id=6002@${domain_name}"/> </condition> </extension> CALL STATUS AFTER PICKUP: freeswitch@internal> show calls uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,des t,presence_id,presence_data,callstate,callee_name,callee_num,callee_directio n,call_uuid,hostname,sent_callee_name,sent_callee_num,b_uuid,b_direction,b_c reated,b_created_epoch,b_name,b_state,b_cid_name,b_cid_num,b_ip_addr,b_dest, b_presence_id,b_presence_data,b_callstate,b_callee_name,b_callee_num,b_calle e_direction,b_sent_callee_name,b_sent_callee_num,call_created_epoch 54b084a8-6f0c-11e2-9612-1dd513f41a4b,inbound,2013-02-04 16:49:12,1360010952,sofia/internal/101@192.168.209.49,CS_SOFT_EXECUTE,ext101 ,101,192.168.209.23,6001,6001(a)192.168.209.49,,ACTIVE,,,SEND,54b084a8-6f0c-11 e2-9612-1dd513f41a4b,localhost.localdomain,voice2net,6137041248,4f6e82e2-6f0 c-11e2-95e8-1dd513f41a4b,inbound,2013-02-04 16:49:03,1360010943,sofia/external/6137041248@98.158.129.14,CS_EXCHANGE_MEDI A,voice2net,6137041248,98.158.129.14,6001,6001(a)192.168.209.49,,ACTIVE,voice2 net,6137041248,SEND,ext101,101,1360010952 1 total. VERSION: freeswitch@internal> version FreeSWITCH Version 1.3.13b+git~20130119T155523Z~6cb3be7d8b (git 6cb3be7 2013-01-19 15:55:23Z) All aspects of the call work, we want to use the blf with the valet park and this does work, however, because the call does not drop, the valet_park light stays on, this unnecessarily ties up parking spots. Any help or suggestions would be greatly appreciated. Darcy

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Callgroup keeps ringing after answer - changed from shared extensions
by Mario G 04 Feb '13

04 Feb '13

Been at this for 3 weeks now, hope someone can help.... This started after replacing shared extensions with unique users and updating from an early 2011 git. Ring group via "pointer" works fine as it always has. However, answer via callgroup variable intermittently (30-40%) leaves one or mores phone ringing even though the the call was answered at another extension. The darn thing is that I can't reproduce it myself to get a trace. Any help/suggestions greatly appreciated. Phones are SPA962 (UDP) and Bria on iPad (TCP, same problems) and they all stopped ringing when they shared IDs. Mario G default.xml ring-group has (abbreviated to save space)------------------: <group name="home"> <users> <user id="100" type="pointer"/> <user id="101" type="pointer"/> ..... more users </users> </group> <group name="work"> <users> <user id="200" type="pointer"/> <user id="201" type="pointer"/> </users> </group> </groups> Extension----------------------: <user id="101"> User: <params> <param name="password" value="$${default_password}"/> </params> <variables> <variable name="voicemail_domain" value="xxxxxx.com"/> <variable name="toll_allow" value="domestic,domestic10,international,local"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Front Desk"/> <variable name="effective_caller_id_number" value="101"/> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/>  <variable name="callgroup" value="home"/> </variables> </user> Bridge command-----------------: <extension name="xxphone"> <condition field="destination_number" expression="^2920$"> <action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/> <action application="bind_meta_app" data="2 b s record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/>  <action application="execute_extension" data="2925 XML default"/>  <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="ringback=file_string://${xxsounds}hi-long.wav!${sayname}!tone_stream://${us-ring};loops=-1"/>  <action application="bridge" data="{originate_timeout=17}${group_call(home@${domain_name}+A)}"/>

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ESL outbound race condition?
by David Villasmil 04 Feb '13

04 Feb '13

Hello guys, I've writing an app using outbound socket, when the channel is created from FS to my app, I get an event called "SOCKET_DATA", I was expecting a CHANNEL_CALLSTATE or similar. I read somewhere that it is related to a race condition? I'd love some help! Thanks! David

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SIP SRV failover behavior
by Jeff Pyle 04 Feb '13

04 Feb '13

Hello, I use an SRV record as the proxy value for a sofia gateway. This particular record is configured for failover rather than load balancing. In other words, there is only one record per priority. I've verified Freeswitch will advance to the second host if the first becomes unavailable. It seems Freeswitch waits around 10 minutes before it tries the first host again. Is this delay a configurable value? Is there some kind of internal blacklist the failed host occupies before this timer expires? Is this list manageable from the CLI? Besides a 408, are there other response codes that will land a host on this apparent blacklist? - Jeff

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Routing messages in mod_sms chatplan
by Yuriy Ostapchuk 04 Feb '13

04 Feb '13

Hi, I have a setup where I need to route incoming sms messages from mod_gsmopen to different sip or dingaling extensions, depending on sender's phone number and other conditions. Is there a way to do this directly in chatplan without executing an external script? Basically, I'm looking for a functionality similar to 'transfer' application in dialplan. Thanks.

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Dequeue DTMF from FreeTDM Channel
by Subhash 03 Feb '13

03 Feb '13

Hi all, I have a freeswitch configured with freetdm and wanpipe.My problem is with dtmf. When calling from an external pstn number to freeswitch, dtmf seems to be detected properly and enqueued it in the channel consistently. But when call came to same channel again, dtmf which is stored in the buffer for the previous call is used in the current call.So when i am done with previous call i want to dequeue the dtmf in the channel .So the problem seems to be something I've misconfigured with the Sangoma card. Can anyone point me in the right direction? Thanks, Subhash.

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Re: [Freeswitch-users] Choice of codecs for incoming call
by Brian West 02 Feb '13

02 Feb '13

You can't reinvite our 200ok has only pcmu so yes it will 488 Sent from my iPhone On Feb 1, 2013, at 10:56 PM, "Александр Мустафин" < mustafin.aleksandr(a)gmail.com> wrote: Hello, Steven. Thank you for reply. Maybe my DID-provider is buggy, but I've succesfull calls in the other pbx with the same number. Short examples below - second example is work. FS (1st test) ----> INVITE (PCMU,PCMA) <---- 100 Trying <---- 200 OK (PCMU) ----> ACK ----> Re-INVITE (PCMA) <---- 100 Trying <---- 488 Not Acceptable Here CG (2nd test) ----> INVITE (PCMU,PCMA) <---- 100 Trying <---- 200 OK (PCMU, PCMA) ----> ACK ----> Re-INVITE (PCMA) <---- 100 Trying <---- 200 OK (PCMA) ----> ACK 02.02.2013 02:41, Steven Ayre пишет: How can I manage set of codec in inbound calls? <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,G729,GSM"/> <X-PRE-PROCESS cmd="set" data="inbound_codec_prefs=PCMU,PCMA,G729,GSM"/> <param name="inbound-codec-prefs" value="$${inbound_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/> Your previous configuration used the global list for both parameters. global_codec_prefs isn't actually then used, there's a codec-prefs param for convenience only that sets both inbound and outbound at the same time and isn't actually needed if you're setting both. -Steve On 1 February 2013 11:18, Александр Мустафин<mustafin.aleksandr(a)gmail.com> <mustafin.aleksandr(a)gmail.com>wrote: Hello. My FS receive INVITE with 2 codecs in SDP: PCMA and PCMU. After hunting dialplan, this call forwarding to IVR, and I see next message in log: 2013-01-31 01:23:22.388780 [DEBUG] sofia_glue.c:5094 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2013-01-31 01:23:22.388780 [DEBUG] sofia_glue.c:3077 Set Codec sofia/internal/9089080908@PSTN PCMU/8000 20 ms 160 samples 64000 bits 2013-01-31 01:23:22.388780 [DEBUG] switch_core_codec.c:111 sofia/internal/9089080908@PSTN Original read codec set to PCMU:0 And FS send 200 OK to remote side with one codec in SDP: PCMU. My parameters: In internal.xml (profile) <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/> In vars.xml <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMU,PCMA,G729,G722,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,G729,GSM"/> <X-PRE-PROCESS cmd="set" data="inbound_codec_prefs=PCMU,PCMA,G729,GSM"/> This is normal behavior?How can I manage set of codec in inbound calls? Best regards, Alexander. _________________________________________________________________________ Professional FreeSWITCH Consulting Services:consulting@freeswitch.orghttp://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.… FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershtt… _________________________________________________________________________ Professional FreeSWITCH Consulting Services:consulting@freeswitch.orghttp://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.… FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershtt… _________________________________________________________________________ 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

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