- FreeSWITCH-users - FreeSWITCH Mailing Lists

How to disable Re-invite?
by Balamurugan Mahendran 12 Mar '12

12 Mar '12

2012-03-12 11:08:11.137525 [DEBUG] sofia_glue.c:4874 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2012-03-12 11:08:11.137525 [DEBUG] sofia_glue.c:2925 Already using PCMU 2012-03-12 11:08:11.137525 [DEBUG] sofia_glue.c:4988 Set 2833 dtmf send payload to 101 2012-03-12 11:08:11.137525 [DEBUG] sofia_glue.c:3213 Audio params are unchanged for sofia/external/+919884730340. 2012-03-12 11:08:11.137525 [DEBUG] sofia_glue.c:3223 sofia/external/+919884730340 Setting audio receive payload in *Re-INVITE to 0* * * Thanks, Bala

2 1

Audio problems with unimrcp and Vestec ASR
by Peter Steinbach 12 Mar '12

12 Mar '12

I have an Audio problem with unimrcp and Vestec ASR with mrcp v1. This seems to be a similar problem to thread: "ASR from Freeswitch to MS Speech Server [using MRCP Connector] - Audio Problem" What does work: * MRCP command are passed successsfull in both ways on port 1554 (vestec mrcp server) * Grammar is accepted and recognition is started * I can see RTP stream (wireshark) from FS port 400x to the mrcp server * Result (002 no-input-timeout) is sent back to Freeeswicth and is processed by FS successfully. What does NOT work: * RTP stream is always empty (silence), i grepped this on the network with ngrep, RTP data is empty, there is no change in the audio data when I speak. * so voice recognition will always time out (002 no-input-timeout) Here are my configs: unimrcp.conf.xml <configuration name="unimrcp.conf" description="UniMRCP Client"> <settings> <param name="default-tts-profile" value="unimrcpserver-mrcp1"/> <param name="default-asr-profile" value="unimrcpserver-mrcp1"/> <param name="log-level" value="DEBUG"/> <param name="enable-profile-events" value="false"/> <param name="max-connection-count" value="100"/> <param name="offer-new-connection" value="1"/> </settings> <profiles> <X-PRE-PROCESS cmd="include" data="../mrcp_profiles/*.xml"/> </profiles> </configuration> unimrcpserver-mrcp-v1.xml <include>  <profile name="unimrcpserver-mrcp1" version="1"> <param name="server-ip" value="192.168.178.180"/> <param name="server-port" value="1554"/> <param name="resource-location" value=""/> <param name="speechsynth" value="speechsynthesizer"/> <param name="speechrecog" value="speechrecognizer"/> <param name="rtp-ip" value="auto"/> <param name="rtp-port-min" value="4000"/> <param name="rtp-port-max" value="5000"/> <param name="codecs" value="PCMU PCMA L16/96/8000"/> <synthparams> </synthparams>  <recogparams>  </recogparams> </profile> </include> See logs here http://pastebin.freeswitch.org/18570 Any help is appreciated. Best egards Peter -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de

2 3

Question on Multiple Registrations with Polycom 501
by Ira Tessler 12 Mar '12

12 Mar '12

I have a Polycom 501 that is going in and out of registration and creating multiple registrations. Anyone have an ideas why? Thanks! Registrations: ================================================================================================= Call-ID: 6a66a4f8-215d8f26-cf1e7593(a)192.168.4.115 User: 101(a)19036.cmvtele.com Contact: "user" <sip:101@4.59.18.226:5060> Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.7.0134 Status: Registered(UDP)(unknown) EXP(2012-03-12 10:41:31) EXPSECS(14) Host: fs2.cmvvoip.com IP: 4.59.18.226 Port: 5060 Auth-User: 101 Auth-Realm: 19036.cmvtele.com MWI-Account: 101(a)19036.cmvtele.com Call-ID: c388378d-c3835e0b-c4deaab0(a)192.168.4.115 User: 101(a)19036.cmvtele.com Contact: "user" <sip:101@4.59.18.226:5060> Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.7.0134 Status: Registered(UDP)(unknown) EXP(2012-03-12 10:51:24) EXPSECS(607) Host: fs2.cmvvoip.com IP: 4.59.18.226 Port: 5060 Auth-User: 101 Auth-Realm: 19036.cmvtele.com MWI-Account: 101(a)19036.cmvtele.com Call-ID: 5d0c1f2e-69c026bc-dc40a9b9(a)192.168.4.115 User: 101(a)19036.cmvtele.com Contact: "user" <sip:101@4.59.18.226:5060> Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.7.0134 Status: Registered(UDP)(unknown) EXP(2012-03-12 10:52:06) EXPSECS(649) Host: fs2.cmvvoip.com IP: 4.59.18.226 Port: 5060 Auth-User: 101 Auth-Realm: 19036.cmvtele.com MWI-Account: 101(a)19036.cmvtele.com Total items returned: 3 Ira Tessler ConnectMe (732) 490-9007 x2 www.connectmevoice.com

1 0

Re: [Freeswitch-users] bootstrap
by Peter Olsson 12 Mar '12

12 Mar '12

Usually just once. Sometimes a bigger change might be commited to git which requires a re-bootstrap, but that is very rare. Usually you just need "make current", and it will do everything for you. /Peter -----Ursprungligt meddelande----- Från: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Bzzz Skickat: den 12 mars 2012 04:10 Till: FreeSWITCH Users Help Ämne: [Freeswitch-users] bootstrap Hi, is it necessary to re-bootstrap @ each update, or just once for the first compilation? -- BOFH excuse #404: Sysadmin accidentally destroyed pager with a large hammer. _________________________________________________________________________ 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 !DSPAM:4f5da5dd32767885415725!

2 1

Check SRTP encryption on Leg B
by Ivan Ilves 12 Mar '12

12 Mar '12

Dear FreeSWITCH pros, I need to check SRTP encryption on both call legs. Checking SRTP encryption on leg A was easy enough: --- CUT HERE --- <extension name="check_srtp_on_leg_a" continue="true"> <condition field="${entry_context}" expression="^public$" break="on-true"/> <condition field="${sip_has_crypto}" expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$" break="never"> <anti-action application="log" data="INFO No encryption on leg A. Rejecting call after audible warning."/> <anti-action application="sleep" data="2000"/> <anti-action application="playback" data="mamba.mp3"/> <anti-action application="hangup" data="CALL_REJECTED"/> </condition> </extension> --- CUT HERE --- But how do I check SRTP encryption on leg B? I need this to disallow improperly configured devices to receive unencrypted calls. Could you please help me with this? Thanks in advance!!!

1 0

how does eval command works
by piyush singhai 12 Mar '12

12 Mar '12

I want to know the channel state corresponding to chan_uuid. I tried eval but it is not working can anybody tell me how it will work. I followed the wiki http://wiki.freeswitch.org/wiki/Mod_commands --Piyush

1 0

removing caller_id_name
by François Delawarde 12 Mar '12

12 Mar '12

Hi list, I'd like to be able to remove the caller_id_name when sending a call to my Grandstream HT503, because it makes it hard for me to manage in my analog phone directory. It can be changed easily using "effective_caller_id_name", but setting it to empty string doesn't work. Apparently, setting "effective_caller_id_name" to "_undef_" could be an option, but it only removes it in the "From" field, and not from the PAI or RPID headers. My current workaround is to do that and set sip_cid_type to none, so the Grandstream fetches the CID in the From header. Is there any other way to set it to empty string that I might have missed? Thanks, François.

4 12

Re: [Freeswitch-users] Check for freeswitch session exists
by Peter Olsson 12 Mar '12

12 Mar '12

Since ESL is event driven, there is no way to be 100% sure of this. However, the hangup event won't be missed, I guess it will just show up a little bit later (if you continue to read a few more events). So you will just have to handle the "errors" if you try to playback when the channel is hung up already, and then when you receive the hangup event, you know for sure that the call has ended. /Peter Från: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För piyush singhai Skickat: den 12 mars 2012 08:41 Till: FreeSWITCH Users Help Ämne: [Freeswitch-users] Check for freeswitch session exists Hello, I am using ESL lib for making connection with freeswitch from the Application. Some time we miss the hangup event in application and instead make another execute request on the same session, lets say for playback(). Then we receive CHANNEL_EXECUTE_COMPLETE for application 'park' instead of playback(). Is this an indication that the session has already ended (in this case due to hangup by the user)? Otherwise how can we check before executing any new function that the session is still alive at fresswitch through ESL? --Piyush !DSPAM:4f5da6e632765021882877!

1 0

Re: [Freeswitch-users] seg fault on latest git
by Peter Olsson 12 Mar '12

12 Mar '12

Please report to Jira, and supply all needed information.Also make sure to search open issues first, so it doesn't exist already. /Peter -----Ursprungligt meddelande----- Från: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För andy Skickat: den 12 mars 2012 02:44 Till: freeswitch-users(a)lists.freeswitch.org Ämne: [Freeswitch-users] seg fault on latest git Hi all, I just got segmentation faults with a build from git (yesterday). After launching FS with mod_xml_curl, everything seems normal. If I unload mod_xml_curl from fs_cli, the next call crashes the system with a seg fault. I was able to repeat this behavior several times in a row. cheers _________________________________________________________________________ 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 !DSPAM:4f5da5de32761177412446!

1 0

bootstrap
by Bzzz 12 Mar '12

12 Mar '12

Hi, is it necessary to re-bootstrap @ each update, or just once for the first compilation? -- BOFH excuse #404: Sysadmin accidentally destroyed pager with a large hammer.

1 0