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Re: [Freeswitch-users] Prolems with DTMF
by Rudolf Denert 02 Jun '09

02 Jun '09

Hello, I'm not sure which one is it. But I think I send the digits in RFC 2833. All devicves are supporting RFC 2833. The equipment: (VoATM) Allied Data Copperjet 1614 (ISDN) Siemens Euroset 5020 phone (MGCP) Thomson SpeedTouch 780WL Siemens Euroset 5020 phone (SIP) AVM Fritz!Box 7170 Siemens Euroset 5020 phone If necessary I can send my configuration. Greetz ----- Ursprüngliche Mail ----- Von: "Michael Collins" <msc(a)freeswitch.org> An: freeswitch-users(a)lists.freeswitch.org Gesendet: Dienstag, 2. Juni 2009 20:18:41 GMT +01:00 Amsterdam/Berlin/Bern/Rom/Stockholm/Wien Betreff: Re: [Freeswitch-users] Prolems with DTMF On Tue, Jun 2, 2009 at 9:18 AM, Rudolf Denert < rdenert(a)tng.de > wrote: Hello, I have problems with sending DTMF. The freeswitch server recieves the digits from my telefon (SIP account) if I activate DTMF with: <action application="start_dtmf"/> in my dialplan. The line <--param name="dtmf-type" value="rfc2833"/--> in sofia.conf.xml is active, too. But than I have the problem that the other phone doesn't work. It is a VoATM device. The curious thing is that I see the digits in the logfile whiche were sent from the phone. In the first example I saw nothing. Does anybody have an idea??? Are you trying to send digits inband or RFC2833? Unless there's a compelling reason not to, we recommend 2833. What is the equipment on the far end looking for? _______________________________________________ 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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Prolems with DTMF
by Rudolf Denert 02 Jun '09

02 Jun '09

Hello, I have problems with sending DTMF. The freeswitch server recieves the digits from my telefon (SIP account) if I activate DTMF with: <action application="start_dtmf"/> in my dialplan. The line <--param name="dtmf-type" value="rfc2833"/--> in sofia.conf.xml is active, too. But than I have the problem that the other phone doesn't work. It is a VoATM device. The curious thing is that I see the digits in the logfile whiche were sent from the phone. In the first example I saw nothing. Does anybody have an idea??? Greetz

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effective_caller_id_number on bridge dialstring
by Yuriy Ivzhenko 02 Jun '09

02 Jun '09

Hello all. I have test the lcr overriding the Caller ID functionality. It return dialstring, that contains 'effective_caller_id_number' variable. But that variable has no effect. I try test configuration <extension name="Test_calls"> <condition field="destination_number" expression="^(1001)$">  <action application="bridge" data="[effective_caller_id_number=9999]sofia/internal/sip:1001@192.168.2.43:5060"/> <action application="hangup"/> </condition> </extension> There is no result. (caller id number not changed) But If I uncomment the set line, then the caller_id_number changes. I found the similar bug (http://jira.freeswitch.org/browse/MODAPP-122) and his status - fixed. ....may be i not consider something? I use svn trunk 13544. Yuriy

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Outbound socket question
by Nik Middleton 02 Jun '09

02 Jun '09

Hi Guys, I'm going some work with outbound socket, and have a few questions. When each call is answered, I get a connection to my server socket. Is it right to assume that this connection will remain for the duration of the call? If so, do I still need to pass the UUID when I call an application such as playfile? Regards

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always got 0 messages when retrieving voicemail from web
by dujinfang 02 Jun '09

02 Jun '09

I thought it is a problem, made a jira: http://jira.freeswitch.org/browse/XML-2

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always got 0 messages when retrieving voicemail from web
by dujinfang 02 Jun '09

02 Jun '09

sorry forgot to mention I'm on FreeSWITCH Version 1.0.trunk (13524M)

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always got 0 messages when retrieving voicemail from web
by dujinfang 02 Jun '09

02 Jun '09

Hi, I always got 0 messages when using web. Finally I added some debug information in the code and get this: 2009-06-02 22:20:51 [INFO] mod_voicemail.c:3883 voicemail_api_function() port:[8080] 2009-06-02 22:20:51 [INFO] mod_voicemail.c:3884 voicemail_api_function() uri:[/domains/192.168.1.16/api/voicemail] 2009-06-02 22:20:51 [INFO] mod_voicemail.c:3885 voicemail_api_function() user:[] 2009-06-02 22:20:51 [INFO] mod_voicemail.c:3886 voicemail_api_function() domain:[192.168.1.16] 2009-06-02 22:20:51 [INFO] mod_voicemail.c:3887 voicemail_api_function() path_info:[web] It seems the freeswitch-user header not set by xml_rpc and user = switch_event_get_header(stream->param_event, "freeswitch-user"); cannot get the user. Any idea? Thanks.

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calls appear to be dropping ... from landlines
by Dale Trub 02 Jun '09

02 Jun '09

We're running FreeSwitch as part of a teleconferencing service, inside a telcom (so no internet latency/NAT issues) and using g.729 We are receiving some complaints of dropped calls, including from landlines. This means they join the conference, and x minutes in they simply drop. I know that cellphones tend to drop calls frequently, but landlines are pretty reliable, and we're hearing it a lot. From the FreeSwitch side of things, it just looks like those callers hung up (but then dialed back in just a moment later). I'm attaching two different snippets of the FS log files where these issues are occurring. Does anyone have any recommendations about how to troubleshoot this? Any known issues/patches in FS that could be biting us? Is there some SIP logging we can do to debug? Are there any paid contractors avail who would have the expertise to look into this? Any help appreciated ... this is a major issue for us! Thanks much, -Dale daletrub(a)gmail.com

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Can Freeswitch + LAMP run on 128MB RAM?
by Fred-145 02 Jun '09

02 Jun '09

Hello Atcom's IP01 unit (www.atcom.cn) can be expanded to have 128MB RAM and 1GB NAND flash. Before I go ahead and check, would someone know if a minimal Linux + LAMP server* + Freeswitch can run OK with this amount of memory? Thank you. * I think I'll trade MySQL with Firebird, to avoid buying a license for my commercial application -- View this message in context: http://www.nabble.com/Can-Freeswitch-%2B-LAMP-run-on-128MB-RAM--tp23807353p… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Re: [Freeswitch-users] Make current fails (build 13537)
by freeswitch＠davidnicol.otherinbox.com 02 Jun '09

02 Jun '09

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