May 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] help with mod_conference stability
by pete＠privateconnect.com 14 May '09

14 May '09

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Audio "clicks" between playback of audio files
by Peter Olsson 14 May '09

14 May '09

I'm implementing an IVR solution for FreeSWITCH, but I have a little problem with audio playback. I'm just calling into application park, and then handle the flow using the event socket. All my audio phrases are .PCMA (8KHz a-law), and I play lots of files after eachother. Between each file I can sometimes here a little "click", even though I'm 100% sure that this is not from my files. My guess is that it might be caused of the fact that no RTP is sent at all when the phrase is not playing. If I merge the files together before playing them it sounds just fine. Is it possible to make FreeSWITCH send silence frames, even when not needed? I know this is a waste of resources, but it will still make the solution sound much better. Regards Peter Olsson

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t38modem + FreeSWITCH problem
by Mailings 14 May '09

14 May '09

Hi, I just tried FreeSWITCH together with t38modem and HylaFAX. But when I bridge the call to the modem I get the following error: 2009-05-14 00:22:38 [ERR] sofia.c:3280 sofia_handle_sip_i_state() Reinvite Codec Error! Does anyone know how to set this up? To PSTN I use a Lancom router which converts T38 to Euro-ISDN and vice versa. The t38modem was started with the following parameters: /usr/bin/t38modem -tt -o /var/log/t38modem.log --no-h323 -u t38modem --sip-register t38modem@freeswitch,password --sip-listen udp\$127.0.0.1:6060 --ptty +/dev/ttyT38-1 --route "modem:.*=sip:<dn>@127.0.0.1" --route "sip:.*=modem:<dn>" I also created a directory entry for FreeSWITCH with the following content: <include> <user id="t38modem"> <params> <param name="password" value="password"/> </params> <variables> <variable name="toll_allow" value="domestic,international,local"/> <variable name="accountcode" value="t38modem"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Fax"/> <variable name="effective_caller_id_number" value="xxxxxxxx"/> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/> <variable name="max_calls" value="1"/> <variable name="callgroup" value="default"/> </variables> </user> </include> The bridging is done in perl: $session->execute("bridge","user/t38modem@" . $VARS{"domain"}); Any help is appreciated. Thanks, Klaus

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Mod_xml_curl doesn't exists Trunk 13288
by Saeed Ahmed 14 May '09

14 May '09

Hi, I just did 'make current' and FS failed to load xml_curl, I looked into mod/ dir but mod_xml_curl.so wasn't there. I even tried to copy it from src directory but somehow FS doesn't load it. What could be the problem? Thanks

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Set nibblebill variables on B-leg
by Yuriy Ivzhenko 14 May '09

14 May '09

Hi All, I want to use mod_nibblebill for billing outgoing calls. My problem is billing begins before other paty answers the call. The WiKi tells: "You need to set the billing variables on your outbound calling leg and NOT on your A-Leg." How i can set nibblebill variables when B-leg connects?

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Suppressing music on hold in conferences
by Jason White 14 May '09

14 May '09

>From the default dialplan: <condition field="destination_number" expression="^9(888|1616|3232)$"> <action application="export" data="hold_music=silence"/> I have just written a similar entry for another conference that I frequently call. If I call the conference from FreeSWITCH via a SIP phone, then place the call on hold, the hold music is suppressed as intended. However, if I call the conference from PortAudio, the above dial-plan logic doesn't have the desired effect - I still get music on hold. The info application shows that the variable is being exported, however. This is under revision 13266. Is this a bug, or is there additional dialplan logic that I need to supply? If it's a bug, could someone add it to the bug list? Thanks.

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Polycom and SRTP
by Brian West 14 May '09

14 May '09

I finally narrowed down what the heck the polycom wanted when the hold operation was taking place. Now Please test snom, polycom and others that use SRTP. I have tested SNOM and Polycom but others that use SRTP will need to test it also. All I can say is DUMB DUMB DUMB DUMB DUMB!!!!! Brian West brian(a)freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com

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mod_event_socket 183 early media
by İbrahim TUNALI 13 May '09

13 May '09

Hi, I am trying to build a SBC but a kind of different described on wiki. I immediately send incoming leg A to a socket app and connect bridge with my socket python application. I will handle all business logic on python side. My dialplan is so simple; <extension name="SBCSocket"> <condition field="destination_number" expression="^1000$"> <action application="socket" data="127.0.0.1:1905 async full"/> </condition> </extension> Freeswitch send calls to parking and 183 early media through mod_sofia. This is unacceptable on my SBC scenario. Because I not receive 183 or 180 from leg B yet. I want to relay early media or ringing when I received from B leg I realize this is not a bug. It is a feature but i need to know is this possible. I would appreciate all suggestions and clues. Regards, ibrahim

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Question about ACL
by Helmut Kuper 13 May '09

13 May '09

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, today I found tha FS isn't authenticating incomming calls from local extensions. I use an ACL and I have auth-calls set to true in default sip profile. <param name="apply-inbound-acl" value="domains"/> FS log shows, that ACL is granting access but FS doesn't authenticate the call. When I disable the ACL in default profile, authentication is performed. FS WIKI doku (http://wiki.freeswitch.org/wiki/Acl#Sofia) chapter "sip_profiles" says this: "The acl behaviour is modfied by auth-calls, accept-blind-reg and accept-blind-auth" I rather think ACL disables those parameters. What I want is an ACL check followed by authentication. I use "FreeSWITCH Version 1.0.trunk (13112M)" Any ideas? regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFKCtih4tZeNddg3dwRAjtYAJsGqPjBp+FVEKZl4GTQNq3TLoh+JwCgiUiZ wJcqu2XOQ7TV+0tBU8REEFc= =M/pp -----END PGP SIGNATURE-----

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RFC: pickup using a two nodes FS PBX
by Helmut Kuper 13 May '09

13 May '09

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I want to allow pickup in a two nodes/servers FS PBX sharing the same database. So phone A on node 1 rings phone B on node 1. phone C on node 2 tries to pickup phone A. mod_limit is working on both nodes sharing the same database. So pickup function in node 2 sees the incomming call from phone A and knows the corresponding uuid. Can I use such functions like intercept in this multi server context? regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFKCnz44tZeNddg3dwRAhCOAKCq7H7OyXvIRivnWjKYw6IaKzQCCgCfQQQ0 rWLR9w1lJ4CrPj+nu0cV6EE= =jQwK -----END PGP SIGNATURE-----

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