July 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Problem of FreeSwitch and T.38 Test
by chi zhang 14 Aug '10

14 Aug '10

Hi,guys. This is sammy. Here i met some problem in T.38 fax test on new module: mod_spandsp(based on the latest src) When simulate one T.38 fax receive process with sipP as below: 1) 1000 calls to 666666, and call is established. 666666 is one extension in public.xml <extension name="666666"> <condition field="destination_number" expression="^(666666)$"> <action application="answer" /> <action application="playback" data="silence_stream://2000"/> <action application="rxfax" data="/tmp/999.tiff"/> <action application="hangup"/> </condition> </extension> 2) 666666 send re-INVITE to 1000 with T.38 SDP. Here re-INVITE retrys 5 times, and FS return " Fax processing not successful - result (48) Disconnected after permitted retries. " i print some logs for libs/spandsp T.30 handle, found in function t30::repeat_last_command(), t30_state_t->state is T30_STATE_R(17) , so FS return " Disconnected after permitted retries ". Does anyone know the reason? Where is the span_log of spandsp lib ? i cannt find it... regards sammy

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mod_xml_curl question.
by Sergey Okhapkin 14 Aug '10

14 Aug '10

I want to use mod_xml_curl to retrieve only dynamic dialplan sections, but to keep a large chunk of dialplan code in a static XML file to minimize the number of http requests. But when mod_xml_curl is loaded, dialplan search/execution starts with xml_curl execution. How to avoid that? I need static xml to be looked up/executed first, and retrieve dialplan context from web server only if the required dialplan context is not defined in static XML file.

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VM Force Password Change
by Peder 14 Aug '10

14 Aug '10

Is there a way to force a password change upon first login to a voicemail account? We set the password to 1234 and would like to force users to change it the first time they login to the account. Peder

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mod_conference dynamic config options
by Tim St. Pierre 14 Aug '10

14 Aug '10

Hello list, I am trying to dynamically set the moh_sound variable in mod_conference, so that each customer can have a specific music on hold for their company conference room. Most of the options are common, so I have a few template profiles in the conference.conf.xml file. I can change the moh_sound variable to suit, but this would require me to create a profile for each room, then reload it. I would prefer to have this pull from a database. I'm using LUA in the dialplan to setup the conference based on the database values, but I can only see options for setting things like PIN, profile, etc. Is there any way I can set this or other settings with a dialplan variable? Failing that, can mod_conference be bound to a lua script that will give it an XML profile each time it is called? Thanks! -Tim

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sofia_contact response via socket
by Oliver Schönbeck 14 Aug '10

14 Aug '10

Hi all, I recently noticed that the response send after a call to sofia_contact via socket connection lags the two \n at the end. Which leads to some Problems parsing the messages correctly. Is this behaviour intended? Im using sofia_contact to check whether a user is currently registered or not so I am in the need to call this function via socket. Is there an alternative? Kind regards Oliver

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Receiving incoming SIP calls
by Alan Dawson 14 Aug '10

14 Aug '10

Hi, I'm new to VOIP, and freeswitch. I'm trying to enable certain internal extensions to receive incoming SIP calls, eg people can dial sip:1001@my.domain and be able to receive those calls. but when they do this I get Rejected by acl "domains". log message. Any help,howtos,faq etc in resolving this appreciated. -- Alan Dawson

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using FIFO to queue calls towards a gateway
by afshin afzali 14 Aug '10

14 Aug '10

Hi FreeSWITCH, Suppose there is a small voice gateway that just has been equipped with a few FXO ports which connected to PSTN's analog lines. I'm going to use FIFO to manage call requests which going through this voice gateway. As I've learned about mod_fifo service, I think that the queue should have statically defined members (one for each port) which in their dial strings target that gateway (parametrized with the PSTN phone number by a session variable) and nothing else! Am I wrong about this? BEST -- afshin

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FreeTDM and Digium analog cards
by Joseph L. Casale 14 Aug '10

14 Aug '10

Does FreeTDM support TDM cards, or just the digital variants? Reading through Sangoma's docs on FreeTDM, it's not clear? Thanks!

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FreeSWITCH Book Has Arrived!
by Michael Collins 12 Aug '10

12 Aug '10

We are very pleased - and relieved - to announce that the FreeSWITCH book has officially been released! You may purchase it online here: https://www.packtpub.com/freeswitch-1-0-6-build-robust-high-performance-tel… There is an eBook special price of US$4.80 - take advantage of this price because we do not know how long it will be available. Tony, Darren, and I would like to thank everyone who proofread the text, offered feedback, and generally gave us encouragement when things got tough. Writing a book is quite an experience and we are better off for the experience. We learned a lot, to be sure, and we rediscovered what a supportive and enthusiastic community we have with FreeSWITCH. We appreciate your continued support of open source projects like FreeSWITCH and the open source telephony developer conference, ClueCon. Keep up the great work! -Michael S. Collins Co-author

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Increase the recording volume
by Samuel Macedo 12 Aug '10

12 Aug '10

Hi, I'm having some problems with the recorded messages - I'm using the uuid_record command and ".wav" extension. I want to increase the volume of the files, they are so low. Is there anything that I can do? Thanks, -- Samuel Macedo

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