October 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

originate_disposition
by Madovsky 28 Oct '11

28 Oct '11

Afer hours of test I didn' succeed how to catch the result of originate_disposition -------- I have this extensions in my dialplan <extension name="internal_test"> <condition field="${sip_to_uri}" expression="^(999\d{10})@$${domain}$"> <action inline="true" application="set" data="nibble_account="/> <action inline="true" application="set" data="nibble_rate="/> <action inline="true" application="set" data="dialed_extension=$1"/> </condition> <condition field="${from_admin_services}" expression="^true$" break="never"> <action application="set" data="ignore_early_media=true"/> <action application="set" data="hangup_after_bridge=false"/> <action application="set" data="continue_on_fail=true"/> <action application="bridge" data="{sip_append_audio_sdp='a=rtpmap:0 PCMU/8000',sip_invite_domain=${sip_from_host},nibble_account=,nibble_rate=,origination_caller_id_name=${caller_id_name},origination_caller_id_number=${caller_id_number,}}user/${dialed_extension}"/> </condition> </extension> <extension name="internal_test_tools"> <condition field="${originate_disposition}" expression="^SUBSCRIBER_ABSENT$" break="on-true"> <action application="phrase" data="bad_ob_prepaid_dest"/> <action application="transfer" data="ob_prepaid_call XML features"/> <action application="info"/> </condition> <condition field="${cond(expr(${originate_disposition} != NORMAL_CLEARING) ? YES : NO)}" expression="^YES$"/> <condition field="${cond(expr(${originate_disposition} != SUBSCRIBER_ABSENT) ? YES : NO)}" expression="^YES$"> <action application="ring_ready"/> <action application="sleep" data="2000"/> <action application="answer"/> <action application="sleep" data="1400"/> <action application="voicemail" data="default $${domain} ${dialed_extension}"/> </condition> </extension> ----------- I tried to merge it in one extension without success also. where is the trick ? :) Thanks Franck

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switch_rtp.c - Auto Changing Port
by MarhBere 28 Oct '11

28 Oct '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-1bf97fa 2011-10-11 15-13-23 -0500)" SO: CentOs 5. 64x This happens to me a long time and I can not solve it. 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/switch-rtp-c-Auto-Changing-Po… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Docs Down?
by Peter Blackford 28 Oct '11

28 Oct '11

Is docs.freeswitch.org returning blank pages for everyone or is it just me? Regards, Peter

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Getting NO_ROUTE_DESTINATION
by Miha Zoubek 28 Oct '11

28 Oct '11

Hi, I am new Freeswitch user. I have read freeswitch book. I registered first few phones and make few inbound calls. I am trying to get outgoing calls working but having no luck. I have configured my gw and set up calling plan. When I try to dial number 051357952 I get this: reeswitch(a)localhost.localdomain> 2011-10-26 15:27:25.919562 [NOTICE] switch_channel.c:669 New Channel sofia/internal/1030(a)172.31.1.157 [f2132149-dc40-4601-bd34-c22e5b06b696] 2011-10-26 15:27:25.955550 [INFO] mod_dialplan_xml.c:418 Processing 1030->9051357952 in context default 2011-10-26 15:27:25.960548 [NOTICE] switch_channel.c:669 New Channel sofia/external/051357952 [5ccfd0ae-1bcc-4f19-9ca1-e19f8808e6a4] 2011-10-26 15:27:26.095518 [NOTICE] sofia.c:4789 Hangup sofia/external/051357952 [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION] 2011-10-26 15:27:26.098534 [INFO] mod_dptools.c:2355 Originate Failed. Cause: NO_ROUTE_DESTINATION 2011-10-26 15:27:26.098534 [NOTICE] mod_dptools.c:2418 Hangup sofia/internal/1030(a)172.31.1.157 [CS_EXECUTE] [NO_ROUTE_DESTINATION] 2011-10-26 15:27:26.098534 [NOTICE] switch_core_session.c:1182 Session 49 (sofia/external/051357952) Ended 2011-10-26 15:27:26.098534 [NOTICE] switch_core_session.c:1184 Close Channel sofia/external/051357952 [CS_DESTROY] 2011-10-26 15:27:26.140506 [NOTICE] switch_core_session.c:1182 Session 48 (sofia/internal/1030(a)172.31.1.157) Ended 2011-10-26 15:27:26.140506 [NOTICE] switch_core_session.c:1184 Close Channel sofia/internal/1030(a)172.31.1.157 [CS_DESTROY] GW: <include> <gateway name="IP800">  <param name="username" value="Demo2"/> <param name="password" value="1234"/> <param name="register" value="true"/> <param name="realm" value="172.31.1.200"/> </gateway> </include> This is my dial plan: <include> <extension name="Dial Out Custom Gateway"> <condition field="destination_number" expression="^9(\d{9})$"> <action application="bridge" data="sofia/gateway/IP800/$1"/> </condition> </extension> </include> sofia status Name Type Data State ================================================================================================= external profile sip:mod_sofia@172.31.1.157:5080 RUNNING (0) external::example.com gateway sip:joeuser@example.com NOREG external::IP800 gateway sip:Demo2@172.31.1.200 REGED internal profile sip:mod_sofia@172.31.1.157:5060 RUNNING (0) 172.31.1.157 alias internal ALIASED ================================================================================================= 2 profiles 1 alias Please help me why I am getting not route to destination. Thank you very much for all your help! BR, Miha

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RTP auto adjust
by Ben Naylor 28 Oct '11

28 Oct '11

Hi Freeswitch community I was wondering if someone could help me with the above? The scenario is that I have one sip client connected to Freeswitch over 3G, and one connected via Wifi. When I make a call from the wifi to 3G, I can see in the console debug messages this - 2011-10-28 12:14:42.777087 [INFO] switch_rtp.c:3164 Auto Changing port from 192.168.16.18:60416 to 193.xxx.xxx.123:60416 Success, the audio flows both ways. However, when I then make a call from 3G to Wifi, I do not get this line in the debug messages for the wifi sip client. Why is this? Does the auto change only apply to the originating sip client? The 3G client is passing the correct IP in the SDP details due to there being a SIP ALG router in place to change the details. Any help is greatly appreciated, as always. Kind regards Ben

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Re: [Freeswitch-users] High load on database server
by Madovsky 28 Oct '11

28 Oct '11

but, is anyone experienced that with shared ODBC ? (managing more than 330 concurrent calls) ----- Original Message ----- From: "Madovsky" <infos(a)madovsky.org> To: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> Sent: Thursday, October 27, 2011 12:33 PM Subject: Re: [Freeswitch-users] High load on database server > ha ok, good luck so > > ----- Original Message ----- > From: "Cliff Wells" <cliff(a)develix.com> > To: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> > Sent: Thursday, October 27, 2011 12:26 PM > Subject: Re: [Freeswitch-users] High load on database server > > >> On Thu, 2011-10-27 at 11:55 -0400, Madovsky wrote: >>> I means that depend the quality of your script ;) >> >> Quality isn't the issue here. The script is too simple to be incorrect >> (and it carefully releases the odbc connection back to the pool in a >> hangup handler). I have written poor-quality code plenty of times, but >> 25 years as a programmer usually allows me the luxury of knowing when >> I'm doing it, thanks. >> >> In any case, Lua scripts work fine. Google will tell you that unixODBC >> before 2.3.0 was a bit of a mess, so I expect the issue lies there, but >> again, I emphasize, everything WORKS GREAT (been using it for a couple >> of years now) until you get to very high concurrency (about 330 >> concurrent calls on a single system). If you don't expect to handle >> more than 300 concurrent calls, then you do not need to worry about it. >> At all. >> >> Regards, >> Cliff >> >> >> >> 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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./configure fails on vanilla checkout
by Pete Kelly 28 Oct '11

28 Oct '11

Hi I just checked out freeswitch from Git (git clone git:// git.freeswitch.org/freeswitch.git) did ./bootstrap.sh, commented out the sangoma codec module from modules.conf and ran ./configure. The ./configure fails when it gets to libs/iksemel/configure +11073 It complains about the syntax of the PKG_CHECK_MODULE statement. I commented out the whole if statement and it configures fine. Pete

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FreeTDM and internal extensions and caller_id (user directory?)
by Trever L. Adams 28 Oct '11

28 Oct '11

Hello all, I have a small set of internal extensions working. I have an 8 port DAHDI card. 6 FXS, 2 FXO. I have it so I can call between FXS ports (extensions 1001-1005 right now are configured). This works fine. The problem is that the caller ID information is wrong (the default is FreeSwitch/0000000000). I need to be able to set this on a per port basis. I see that conf/directory is the place to set this up for sip. How/where do I do this for FreeTDM? I have not been able to find any documentation for this. I am afraid I have spent a few hours looking. If there is anything out there, I must be using the wrong search terms. Thanks for any help, Trever

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trouble making conditions for OR
by Trever L. Adams 28 Oct '11

28 Oct '11

Hello everyone, The last few days I am beginning to explore freeswitch. I am having a bit of trouble. I am trying to user the examples near http://wiki.freeswitch.org/wiki/Dialplan_XML#Complex_Condition.2FAction_Rul… in the documentation. If I use '\' as it says, I get an error about </extension> being unexpected. If I use '/' it doesn't work. Below is my dialplan. It should trigger on empty number as well as the strings mentioned. I am pretty sure the regex is accurate. However, instead of running, I get: 2011-10-27 14:38:16.269266 [INFO] mod_dialplan_xml.c:336 Processing private <private>-> in context Incoming-FXO 2011-10-27 14:38:16.269266 [INFO] switch_core_state_machine.c:177 No Route, Aborting The dialplan: <include> <context name="Incoming-FXO"> <extension name="screen"> <condition field="caller_id_number" expression="^$" break="on-true"/> <condition field="caller_id_number" expression=".nonymous.*" break="on-true"/> <condition field="caller_id_number" expression=".nknown.*" break="on-true"/> <condition field="caller_id_number" expression=".*rivate.*" break="on-true"/> <condition field="caller_id_number" expression=".estricted.*" break="on-true"/> <condition field="caller_id_number" expression="PTSN.*" break="on-true"/> <condition field="caller_id_number" expression=".oll.*" break="on-true"/> <condition field="caller_id_name" expression=".nonymous.*" break="on-true"/> <condition field="caller_id_name" expression=".nknown.*" break="on-true"/> <condition field="caller_id_name" expression=".*rivate.*" break="on-true"/> <condition field="caller_id_name" expression=".estricted.*" break="on-true"/> <condition field="caller_id_name" expression="PTSN.*" break="on-true"/> <condition field="caller_id_name" expression=".oll.*" break="on-true"/> <condition field="caller_id_name" expression=".--.*" break="on-true"/> <condition field="caller_id_name" expression=",.*" break="on-true"/> <condition field="caller_id_name" expression="000.*"> <action application="set" data="call_screen_filename=/tmp/${caller_id_number}-name.wav"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="phrase" data="voicemail_record_name"/> <action application="playback" data="tone_stream://%(500, 0, 640)"/> <action application="set" data="playback_terminators=#*0123456789"/> <action application="record" data="${call_screen_filename} 7 200 2"/> <action application="set" data="group_confirm_key=1"/> <action application="set" data="fail_on_single_reject=true"/> <action application="set" data="group_confirm_file=phrase:screen_confirm:${call_screen_filename}"/> <action application="set" data="continue_on_fail=true"/>  <action application="voicemail" data="default $${domain} $1"/> <action application="hangup"/> </condition> </extension> </context> </include> Any help would be greatly appreciated. Thank you, Trever -- "I know who I am. No one else knows who I am. If I was a giraffe, and someone said I was a snake, I'd think, no, actually I'm a giraffe." -- Richard Gere

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Strange echo problems
by Boris Kovalenko 28 Oct '11

28 Oct '11

Hello! I have found a very strange echo problem with different devices. When then user starts speaking there is no echo, but after about 1 minute of talking echo begins. If user stop talking echo disappears and again when user start talking there is echo again after about 1 minute talking. What is this? Codec is G711a, FreeSWITCH Version 1.0.head (git-abc5d7c 2011-10-18 20-48-34 -0400). Any help please. -- Regards, Boris

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