March 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

mod_cidlookup and multi-tenant
by Sergey Zhuravlov 06 Mar '13

06 Mar '13

Hi I want to use the module in the multi-domain system all is well, but I want to pass except the numbers in the URL domain name url: http://127.0.0.1:9393/?number=${caller_id_number}&domain=${domain} To in the future, depending on the domain to request a different CRM But substituted the common domain of the settings, not the one that is defined before calling cidlookup <action application="set" data="domain=xxx.xxx.xxxxx.com"/> <action application="set" data="cid_name=${cidlookup(${caller_id_number})}"/> Or another option to have your URL for cidlookup for each domain? Or other ideas? -- WBR, Sergey GTALK/JABBER:4orbit@gmail.com

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[video-media-bug] How to make and load the mod_mp4v2 and mod_fsv modules?
by K K 06 Mar '13

06 Mar '13

Hello, My aim is to record the video from both UAC during a video SIP call. I have checkout the "video-media-bug" branch, because as far as I know there is the source code of the mod_mp4v2 and mod_fsv which can record the video during the communication. I would like to view the recorded video later by some external player, so I guess I should focus on the mod_mp4v2, because the *.fsv is just internal FreeSWITCH format. I am compiling on Windows using the Microsoft Visual C++ 2010 Express. I have firstly compiled the mod_fsv project and replaced the mod_fsv.dll in my FreeSWITCH instalation (1.1.13b). But after running FreeSWITCH I receive the error message box "This application has failed to start because MSVCR100D.dll was not found" and in the console I get the "dll open error" for mod_fsv. Reinstalling the "Visual Studio 2010 C++ Redistributable" package did not help. So I guess the such simple replacement is not working and I will need to compile the whole FreeSWITCH to check how the mod_fsv from "video-media-bug" branch works? So next I have started focus on the mod_mp4v2, but I was not able to find the project file for this module. This module is placed in the "C:\FS_GIT\src\mod\formats" directory, on the Wiki page I have found that it requires the mp4v2 2.0.0 code to be copied into that directory, but still I do not see what project should I compile to obtain the mod_mp4v2.dll for FreeSWITCH? Thank you for any ideas and help.

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switch_core_sqldb.c:579 NATIVE SQL ERR [database is locked]
by Yihui Li 05 Mar '13

05 Mar '13

Hi, I have updated fs to git head today. But the system stopped work and giving out these error message switch_core_sqldb.c:579 NATIVE SQL ERR [database is locked]

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xml cdr - attended transfer cdr - missing aleg uuid
by Lloyd Aloysius 05 Mar '13

05 Mar '13

Hi All I need your help to solve the following issue. I use the following example to explain my issue. User : Marv - Call - User: Dave User : Dave Answer the Call User: Dave - xfer - dial User : Crystal User : Crystal answer the call User : Dave - xfer complete the transfer -- *a_leg cdr* I see one row Dave to Crystal and a_leg_uuid and bridge_uuid. Everything correct here. *b_leg_cdr* I see two records The first record Marv - Dave but *MISSING a_leg_uuid *why the aleg_uuid missing here. I use the following xml_cdr->variables->originating_leg_uuid is comes as empty. Why ? All other scenarios this xml_cdr->variables->originating_leg_uuid works except the attended transfer. Any help is appreciated. Thanks Lloyd * *

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mod_xml_curl docs maintainer
by Cal Leeming [Simplicity Media Ltd] 05 Mar '13

05 Mar '13

Hello, Sadly I have not had any time to work on mod_xml_curl for the last few months, therefore I have unassigned myself as the maintainer for this module documentation. This page could still use some work, so if anyone else decides to pick up on maintaining this page, feel free to hit me up directly if you have any questions or need some guidance. http://wiki.freeswitch.org/wiki/Mod_xml_curl Cal

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Re: [Freeswitch-users] Failing build process - MVC++E2K10 - Why paths have mixed the \ / characters ?
by Peter Olsson 05 Mar '13

05 Mar '13

Sounds quite strange. There is also the "last resort" to use "git clean -fdx" to get it back to the state as it was newly checked out from the git server. However, as I said - using git HEAD probably handles this better already, since it has quite a few improvements for downloading and extracting files. But anyway - good to hear you finally got it up and running! /Peter Från: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För K K Skickat: den 5 mars 2013 15:20 Till: FreeSWITCH Users Help Ämne: Re: [Freeswitch-users] Failing build process - MVC++E2K10 - Why paths have mixed the \ / characters ? Thank you for all help. I still do not understand why it was failing i.e. why the openssl.tar.gz was not automatically downloaded and unpacked during the build process, but I have done it manually, but this time I have copied all the content (not only the crypto directory) and it starts to work. Thank you. !DSPAM:5135fb6a32768157317768!

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aastra connection with freeswitch
by Archana Venugopan 05 Mar '13

05 Mar '13

Hi, Can anyone please tell me how aastra phones retrieve voicemail messages from our freeswitch? And also how aastra phone indicates the missed call and voicemail messages from freeswitch? Thanks Regards, Archana

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Maximum message size
by Kevin Mathy 05 Mar '13

05 Mar '13

Hi List, Another question ;-) , as described in FFT Doc 10.001, the maximum message size is 2048 bytes for SIP messages, and 1024 bytes for SDP bodies; could you, please, confirm that FS is compliant with that ? If not, what the maximum message size supported / sent by FS ? Thanks a lot, * * * Bien cordialement, * * Best Regards, **Kevin MATHY* * * --

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Bridge of inbound/outbound calls via outbound socket or ht-tapi
by Denis Gasparin 05 Mar '13

05 Mar '13

Hi list. We would like to implement a workflow like this: • a user call an inbound dialplan handled with outbound socket (plivo) or ht-tapi. • via ht-tapi (or plivo) the user can navigate between classic ivr menus etc etc • when the user enters a particular menu we would like to initiate an outbound call to another number (external to freeswitch): the outbound call should be handled by ht-tapi (or plivo) too • after some business logic, the outbound call should be bridged to the inbound call. Is there any best practice or tip? We've got the bridge working with a lua script in the outbound call but never with both calls handled via outbound socket or ht-tapi. Thank you in advance. Denis

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tel URIs and FreeSwitch
by Kevin Mathy 05 Mar '13

05 Mar '13

Hi List, Could you, please, confirm that FreeSwitch (Version :* FreeSWITCH Version 1.3.13b+git~20130213T200819Z~cd7d1efc2a (git cd7d1ef 2013-02-13 20:08:19Z)*) supports "tel:" URIs in all SIP messages ? For example, if I receive on my FS an INVITE from a provider like this : *INVITE <tel:+33312345678@example.domain>* Could FS be able to parse and process this URI as it does with "sip:" URIs ? I've found that Sofia-SIP is able, but what about mod_sofia in FS ? Thanks a lot, * Bien cordialement, Best Regards, **Kevin MATHY* * * --

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