June 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Problem with attendant transfer
by Raymond Chandler 08 Jun '09

08 Jun '09

Durk de Beer wrote: > > Hello all, > > I have observed an issue on using Freeswitch and some SIP-phones. Ok > the problem is this. Some phones > which phones? -Ray

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Inbound using FS
by Rex_Alex 08 Jun '09

08 Jun '09

Hello, I have installed FS, and tested outbound successfully. Now I am just trying to do the inbound testing. I got the Inbound DID. Please suggest me what changes should I make and where? Thanks, Rex -- View this message in context: http://n2.nabble.com/Inbound-using-FS-tp3012286p3012286.html Sent from the freeswitch-users mailing list archive at Nabble.com.

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DTMF Problems
by Rudolf Denert 08 Jun '09

08 Jun '09

Hello! Is there a possibility to "detect" or "scan" which DTMF mode is sent by the calling CPE so that I can establish logical interrogation in my configuration? Greetz

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Taking long at startup
by Klaus Teller 08 Jun '09

08 Jun '09

Hi, Freeswitch is taking quiet some time to start. Is is normal these days? it didn't used to be the case few months ago. Is there anything i can turn off to start faster? Thanks, Klaus. -- GRATIS für alle GMX-Mitglieder: Die maxdome Movie-FLAT! Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01

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Re: [Freeswitch-users] Problem with attendant transfer
by Brian West 08 Jun '09

08 Jun '09

This makes no sense.... Can you try to explain it more? I do attended transfers with sip phones every day without a single problem. Maybe i'm missing what you're talking about. /b On Jun 8, 2009, at 2:37 AM, Durk de Beer wrote: > Hello all, > I have observed an issue on using Freeswitch and some SIP-phones. Ok > the problem is this. Some phones, when attempting an attendant > transfer, put the recipient of the transfer on hold. This results in > Freeswitch starting to stream MOH music to the phone put on hold, if > implemented. When now the original phone is pasing the transfer, > Freeswitch is not going to process this transfer because the > recipient end of it is on hold. It is however terminating the > connections it has with the phone initiating the transfer. This > means that the recipient of the transfer is never coming of hold > again until it terminates the call. > Is there a way to detect this behaviour, so I can get the recipient > of hold before Freeswitch is processing the transfer?. > Kind regards > Durk Brian West brian(a)freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com

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Problem with attendant transfer
by Durk de Beer 08 Jun '09

08 Jun '09

Hello all, I have observed an issue on using Freeswitch and some SIP-phones. Ok the problem is this. Some phones, when attempting an attendant transfer, put the recipient of the transfer on hold. This results in Freeswitch starting to stream MOH music to the phone put on hold, if implemented. When now the original phone is pasing the transfer, Freeswitch is not going to process this transfer because the recipient end of it is on hold. It is however terminating the connections it has with the phone initiating the transfer. This means that the recipient of the transfer is never coming of hold again until it terminates the call. Is there a way to detect this behaviour, so I can get the recipient of hold before Freeswitch is processing the transfer?. Kind regards Durk

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Re: [Freeswitch-users] Insufficient RTP stream
by Rudolf Denert 08 Jun '09

08 Jun '09

Ok, here is the SIP trace. If you need more, just tell me and I will send them. The RTP trace you already have, haven't you? Greetz ----- Ursprüngliche Mail ----- Von: "Brian West" <brian(a)freeswitch.org> An: freeswitch-users(a)lists.freeswitch.org Gesendet: Mittwoch, 3. Juni 2009 17:46:14 GMT +01:00 Amsterdam/Berlin/Bern/Rom/Stockholm/Wien Betreff: Re: [Freeswitch-users] Insufficient RTP stream Without a sip trace its hard to tell. /b On Jun 3, 2009, at 10:29 AM, Rudolf Denert wrote: How can I avoid the problem? It seems that RTP auto adjust generates the error. Maybe I can deactivate RTP auto adjust but I suspect that freeSWITCH doesn't find the right media IP. Do you have any other solution? Greetz Brian West brian(a)freeswitch.org -- Meet us at ClueCon! 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 -- -- Rudolf Denert, Technical Support TNG AG, NGN Projensdorfer Str. 324, D-24106 Kiel, Germany phone: +49 431 7097-10, fax: +49 431 7097-555 mailto: rdenert(a)tng.de http://www.tng.de - Register: Amtsgericht Kiel, HRB 6596 KI Supervisory board (Aufsichtsrat): Stefan Scheuermann (Vorsitz) Executive board (Vorstand): Dr. Sven Willert (Vorsitz), Dr. Thomas Rohwer, Sven Schade, Carsten Tolkmit, Dr.-Ing. Volkmar Hausberg Tax-Id (Steuernr.): 1929112637, VAT-Id (USt-Id): DE225201428 - This e-mail may contain confidential and/or privileged information. If you are not the intended recipient (or have received this e-mail in error) please notify the sender immediately and destroy this e-mail. Any unauthorized copying, disclosure or distribution of the material in this e-mail is strictly forbidden. Diese E-Mail enthält vertrauliche und/oder rechtlich geschuetzte Informationen. Wenn Sie nicht der richtige Adressat sind oder diese E-Mail irrtümlich erhalten haben, informieren Sie bitte sofort den Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie die unbefugte Weitergabe dieser Mail ist nicht gestattet.

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busy tone detect issue
by god.nirvana 08 Jun '09

08 Jun '09

hi all i am new to freeswitch. there are some busy tone detect issues,i hope someone could help me. i installed freeswitch from trunk,openzap,zaptel.... but i found some busy tone isuues my tones.conf: [us] generate-dial => v=-7;%(1000,0,350,440) detect-dial => 350,440 generate-ring => v=-7;%(2000,4000,440,480) detect-ring => 440,480 generate-busy => v=-7;%(500,500,450,340) detect-busy =>450,340 generate-attn => v=0;%(100,100,1400,2060,2450,2600) detect-attn => 1400,2060,2450,2600 generate-callwaiting-sas => v=0;%(300,0,440) detect-callwaiting-sas => 440 generate-callwaiting-cas => v=0;%(80,0,2750,2130) detect-callwaiting-cas => 2750,2130 detect-fail1 => 913.8 detect-fail2 => 1370.6 detect-fail3 => 776.7 openzap.conf.xml : <configuration name="openzap.conf" description="OpenZAP Configuration"> <settings> <param name="debug" value="0"/>    </settings> <pri_spans>  <param name="q921loglevel" value="alert"/> <param name="q931loglevel" value="alert"/> <param name="mode" value="user"/> <param name="dialect" value="5ess"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="q921loglevel" value="alert"/> <param name="q931loglevel" value="alert"/> <param name="mode" value="user"/> <param name="dialect" value="5ess"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> </pri_spans>  <analog_spans>    <param name="tonegroup" value="us"/> <param name="digit-timeout" value="2000"/> <param name="max-digits" value="11"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="enable-callerid" value="true"/>        <param name="tonegroup" value="us"/> <param name="digit-timeout" value="2000"/> <param name="max-digits" value="11"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="enable-callerid" value="true"/>        <param name="tonegroup" value="us"/> <param name="digit-timeout" value="2000"/> <param name="max-digits" value="11"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="enable-callerid" value="true"/>        <param name="tonegroup" value="us"/> <param name="digit-timeout" value="2000"/> <param name="max-digits" value="11"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> <param name="enable-callerid" value="true"/>     </analog_spans> </configuration> when i call the pstn phone from a ip phone,if the pstn call hangup first,the ip phone will hear the busy tone,but the system does not handle the busytone ,the channel does not erase. so i have to add <action application="tone_detect" data="busy 450,340 w +25000 hangup 'normal_clearing' 3"/> in the dialplan.and it works.the channel erased. but in the conference case,pstn phone call in,hangup. all participants hear the tone,"do ~,do~~".freeswitch doest handle it. so i change the conference dialplan. <extension name="incoming-fxo-channel-1"> <condition field="source" expression="mod_openzap"> <action application="tone_detect" data="busy 450,340 w +25000 hangup 'normal_clearing' 3"/> <action application="bridge" data="sofia/maqian/6789(a)njpbx.vicp.net"/> </condition> </extension> restart freeswitch,try again,freeswitch not handle the hangup tone still,all participants hear the tone. how to solve it?could some one help me ??? thx! BR M.Q 2009-06-04 god.nirvana

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How to receive a call from another SIP proxy??
by Brad Tuan 08 Jun '09

08 Jun '09

As title, How to receive a call from another SIP proxy??

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How to reject a call without answering
by Klaus Teller 07 Jun '09

07 Jun '09

Hi, Going through the socket api, how can i reject a call without having to answer it first? I tried sending a hangup command with cause set either to NO_ANSWER or NORMAL_CLEARING. In both cases, Freeswitch does create another socket to deliver the very same call. More precisely, when a call comes in i send a connect command. Then after some few seconds, i then send the following hangup command: SendMsg 6debb41e-05a6-4f8a-9003-9f755630519f call-command: hangup hangup-cause: NO_ANSWER Thanks for any feedback. Klaus. -- GRATIS für alle GMX-Mitglieder: Die maxdome Movie-FLAT! Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01

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