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Auto adjust Change port RTP
by MarhBere 26 Sep '11

26 Sep '11

Hi, I have an issue with extension (eyebeam) behind nat. Scenario: The call Incoming From ITSP and is bridged with the a user registered; The RTP arrive to FS from register user, but don't have the auto adjust IP/Port. So only had silence in both points. IF the phone Hold the call and Unhold the call. FS now yet change the RTP to correct private IP/Port and call is now with bidirectional audio. I red that is hapened in old release : http://freeswitch-users.2379917.n2.nabble.com/No-RTP-destination-auto-corre… But I running the last Head trunk. "FreeSWITCH Version 1.0.head (git-8df1872 2011-09-19 13-51-41 -0400)" SO: CentOs 5. 64x Someone know what is wrong? PD: Sorry for my traduction. Thanks, Marcelo Bereterbide. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Auto-adjust-Change-port-RTP-t… Sent from the freeswitch-users mailing list archive at Nabble.com.

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IAX
by f.erfurth 26 Sep '11

26 Sep '11

Hi, I installed FreeSWITCH for BigBlueButton (Conference-System). I want call by using IAX. Is IAX available by default or do I have some configurations for that? I have a dialplan which is responsible for connecting to Conference of BigBlueButton. I should use the context bbb-voip. cu Floh

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Incompelete calls in the log files
by Andy Ayers 26 Sep '11

26 Sep '11

Hi, I have been investigating a number of calls coming in to my switch that don't appear to complete properly. These are incoming calls to a service which records voicemail messgaes. When I look in the logs to see what has happened to these calls the logging information seems to simply come to an abrupt halt at a certain point in the call. In other words there is no information about the call being hung up or the session ended. They all seem to end at the same point in the call and the last few lines in the log file are: 018afe16-e402-11e0-8e40-a5b8fd8e2a79 2011-09-21 04:30:14.848903 [DEBUG] switch_ivr_play_say.c:452 Raw Codec Activated, ready to waste resources! 2011-09-21 04:30:14.849956 [WARNING] switch_core_file.c:176 Sample rate doesn't match 018afe16-e402-11e0-8e40-a5b8fd8e2a79 2011-09-21 04:30:14.849956 [DEBUG] switch_ivr_play_say.c:560 Raw Codec Activated 018afe16-e402-11e0-8e40-a5b8fd8e2a79 2011-09-21 04:30:14.849956 [DEBUG] switch_core_codec.c:122 sofia/external/anonymous@localhost Push codec L16:10 018afe16-e402-11e0-8e40-a5b8fd8e2a79 2011-09-21 04:30:29.028452 [DEBUG] switch_core_io.c:403 Engaging Read Buffer at 320 bytes vs 40 It's also then in most cases followed by a new call coming in: 39cade04-e402-11e0-8e41-a5b8fd8e2a79 2011-09-21 04:31:43.177540 [NOTICE] switch_channel.c:669 New Channel sofia/external/anonymous@localhost [39cade04-e402-11e0-8e41-a5b8fd8e2a79] 39cade04-e402-11e0-8e41-a5b8fd8e2a79 2011-09-21 04:31:43.177540 [DEBUG] switch_core_state_machine.c:314 (sofia/external/anonymous@localhost) Running State Change CS_NEW 39cade04-e402-11e0-8e41-a5b8fd8e2a79 2011-09-21 04:31:43.177540 [DEBUG] switch_core_state_machine.c:320 (sofia/external/anonymous@localhost) State NEW 39cade04-e402-11e0-8e41-a5b8fd8e2a79 2011-09-21 04:31:43.211238 [DEBUG] sofia.c:4153 Channel sofia/external/anonymous@localhost entering state [received][100] I'm guessing the session has simply crashed somehow and freeswitch has been good enough to recover from it. Anyone had a similar experience or can shed any light on what's actually happening and how to prevent it? I'm on version 1.06 Cheers Andy

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G.729
by Fernando Berretta 26 Sep '11

26 Sep '11

Hi, Is there some way to install g729 codec in freepbx for free, in order to "try" it like is provided for asterisk ? Best Regards, Fernando

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Invalid Application reject
by Edmar Cruz 26 Sep '11

26 Sep '11

Hi Guys, I always getting error "Invalid application reject" calling in local extensions and I cannot call locally. But external calls are ok. This are the results. 2011-09-26 11:44:14.564848 [NOTICE] switch_channel.c:669 New Channel sofia/internal/8011105(a)202.6.85.46 [883ec35a-4f52-419a-81ff-fd6d43709868] 2011-09-26 11:44:14.616854 [INFO] mod_dialplan_xml.c:418 Processing 1001->1003 in context default 2011-09-26 11:44:14.868869 [ERR] switch_core_session.c:1731 Invalid Application reject 2011-09-26 11:44:14.868869 [NOTICE] switch_core_session.c:1732 Hangup sofia/internal/1001@xxx [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] 2011-09-26 11:44:15.260894 [NOTICE] switch_core_session.c:1182 Session 1 (sofia/internal/8011105@xxx) Ended 2011-09-26 11:44:15.260894 [NOTICE] switch_core_session.c:1184 Close Channel sofia/internal/8011105@xxx [CS_DESTROY] What maybe the problem? Please help... Thanks -- View this message in context: http://old.nabble.com/Invalid-Application-reject-tp32503926p32503926.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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ACL to fallback digest authentication
by Muhammad Naseer Bhatti 26 Sep '11

26 Sep '11

Hello, I am trying to implement ACL in FreeSWITCH. What I want to do is, check the user's IP in the ACL. If that is present, let the user send INVITE. If ACL fails, fallback to digest authentication and authentication user/pass. Problem is that when I do auth_calls true, I am not able to INVITE. We have actually two scenarios, 1 make call without REGISTER, second REGISTER and make calls. I have set default ACL to deny, but that still not helping. Either one of them works, with the digest or ACL. I don't want to create another profile and make 1 for authentication, and the other without auth. Is there any workaround for this? -B

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Problem compiling
by Steven Lam, KeenSystems B.V. 25 Sep '11

25 Sep '11

Hi, I'm having problems compiling FreeSWITCH on a Gentoo x86_64 machine. I did a git checkout and still have the same problem, the error i see whem compiling is: make "OUR_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /source/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-all ; done )"; echo $mods )" "OUR_CLEAN_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /source/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-clean ; done )"; echo $mods )" "OUR_INSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /source/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-install ; done)"; echo $mods )" "OUR_UNINSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "#" /source/freeswitch/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" "OUR_DISABLED_MODULES=$(tmp_mods="$(grep "#" /source/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-all ; done )"; echo $mods )" "OUR_DISABLED_CLEAN_MODULES=$(tmp_mods="$(grep "#" /source/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-clean ; done )"; echo $mods )" "OUR_DISABLED_INSTALL_MODULES=$(tmp_mods="$(grep "#" /source/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-install ; done)"; echo $mods )" "OUR_DISABLED_UNINSTALL_MODULES=$(tmp_mods="$(grep "#" /source/freeswitch/modules.conf | grep -v "##" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" `test -n "" || echo -s` all-recursive Making all in . cc1: warnings being treated as errors libs/esl/fs_cli.c: In function 'clear_cli': libs/esl/fs_cli.c:109: error: ignoring return value of 'write', declared with attribute warn_unused_result libs/esl/fs_cli.c: In function 'clear_line': libs/esl/fs_cli.c:578: error: ignoring return value of 'write', declared with attribute warn_unused_result libs/esl/fs_cli.c:580: error: ignoring return value of 'write', declared with attribute warn_unused_result libs/esl/fs_cli.c:581: error: ignoring return value of 'write', declared with attribute warn_unused_result libs/esl/fs_cli.c: In function 'redisplay': libs/esl/fs_cli.c:588: error: ignoring return value of 'write', declared with attribute warn_unused_result libs/esl/fs_cli.c:589: error: ignoring return value of 'write', declared with attribute warn_unused_result make[2]: *** [fs_cli-fs_cli.o] Error 1 Is there someone who can help me on this? Thanks! Steven

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Polycom auto answer with microphone muted?
by Yehavi Bourvine 25 Sep '11

25 Sep '11

Hello, I have Polycom phones with which I want to implement Intercom. The users would like it to auto answer with the microphone muted at first. The phones indeed answer but with the mic opened... Anyone know what should be done? I have the following in the dial plan: <action application="set" data="sip_auto_answer=true" /> <action application="set" data="sip_h_Call-Info=<sip:${domain_name}>;answer-after=2" /> <action application="bridge" data="${RingAt}" /> And I have in the phone's config file: <call.autoAnswer call.autoAnswer.micMute="1" /> However, the mic is not muted... Thanks! __Yehavi:

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please help for Failed Registration, setting retry to 1020 seconds
by James zhu 25 Sep '11

25 Sep '11

hi: I am using GSM gateway to regiester to freeswitch, the incoming from gsm gateway is ok, but i can not regiester into freeswitch due to the gateway connection. ===the debug info from fs_CLI ------------------------------------------------------------------------ 2011-09-21 23:14:50.645350 [ERR] sofia_reg.c:1496 gsm Registration Failed with status Request Timeout [408]. failure #33 2011-09-21 23:14:53.318802 [WARNING] sofia_reg.c:386 gsm Failed Registration, setting retry to 1020 seconds. =====sip trace =============== ------------------------------------------------------------------------ send 590 bytes to udp/[172.16.33.11]:5060 at 15:14:50.147741: ------------------------------------------------------------------------ REGISTER sip:172.16.33.11 SIP/2.0 Via: SIP/2.0/UDP 172.16.33.100:5080;rport;branch=z9hG4bKB09cFa6Uvyc1H Max-Forwards: 70 From: <sip:1000@172.16.33.11;transport=udp>;tag=B1ytyyXQS3NKr To: <sip:1000@172.16.33.11;transport=udp> Call-ID: 4809dacc-aeee-4a33-a245-884c6788a4a6 CSeq: 17972965 REGISTER Contact: <sip:gw+gsm@172.16.33.100:5080;transport=udp;gw=gsm> Expires: 30 User-Agent: FreeSWITCH-mod_sofia/1.0.6-svn-exported Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------ send 590 bytes to udp/[172.16.33.11]:5060 at 15:14:46.146854: ------------------------------------------------------------------------ REGISTER sip:172.16.33.11 SIP/2.0 Via: SIP/2.0/UDP 172.16.33.100:5080;rport;branch=z9hG4bKB09cFa6Uvyc1H Max-Forwards: 70 From: <sip:1000@172.16.33.11;transport=udp>;tag=B1ytyyXQS3NKr To: <sip:1000@172.16.33.11;transport=udp> Call-ID: 4809dacc-aeee-4a33-a245-884c6788a4a6 CSeq: 17972965 REGISTER Contact: <sip:gw+gsm@172.16.33.100:5080;transport=udp;gw=gsm> Expires: 30 User-Agent: FreeSWITCH-mod_sofia/1.0.6-svn-exported Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Content-Length: 0 =======gateway confi: ====== <include> <gateway name="gsm">   <param name="username" value="1000"/>   <param name="realm" value="172.16.33.11"/>  <param name="password" value="1234"/>  <param name="expire-seconds" value=""/>  <param name="register" value="false"/>  <param name="sip-trace" value="yes"/> </gateway> </include> freeswitch@internal> sofia status Name Type Data State ================================================================================================= external profile sip:mod_sofia@172.16.33.100:5080 RUNNING (0) external::example.com gateway sip:joeuser@example.com NOREG external::gsm gateway sip:1000@172.16.33.11 FAIL_WAIT internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0) internal profile sip:mod_sofia@172.16.33.100:5060 RUNNING (0) 172.16.33.100 alias internal ALIASED ================================================================================================= somehow, the gateway never sends anything. anyone can give a hit? Best regards, James.zhu Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri<->SIP). website: www.voipviews.com

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Early media playback
by Chris Martineau 24 Sep '11

24 Sep '11

Hi Trying to play a custom announcement/welcome to the a leg on receiving a 183 from the far end. Have tried a straight playback just before the bridge which confirms that it finds the playback file as the recording is heard but as this is a bit nasty FS complains with a codec error and closes the call. Tried execute on media with the same playback details as follows... <action application="set" data="execute_on_media=playback /path/mediafile" Which according to the logs seems to work but I get just silence until I presume the file stops playing which is strange as the file plays fine when using a forced playback? Logs in both cases show the same file being found and used mediafile.PCMA. Also I would like to terminate the playback on answer but cannot find any obvious process to do this. Obviously you could do an execute_on_answer but can't see what I would execute to achieve the desired effect? Many thanks for any help you can offer. Regards Chris

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