February 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Passing custom variables for managed code in xml_curl
by Ashley B 13 Feb '11

13 Feb '11

Hi How do I set my own global vars (X-PRE-PROCESS) using mod_xml_curl that I can use in managed code / script?). Eg mod_managed does not always have the values of the variables required @ runtime, they are always changing. I have to curl them from another server, i know i can set them up as channel variables when calling for dial plan with xml_curl but need to be able to fetch them @ the same time as configuration when calling reloadxml. Any assistance is appreciated. Thanks Ashley

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How to set up a DTMF callback in C or C#?
by Frankie Yiu 12 Feb '11

12 Feb '11

Hi there, If I want to set up a DTMF callBack to a function (in either C or C#) when a caller presses a key, what function should I use to set it up? Thanks, Frankie

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can't email voicemail messages
by Roger Young 12 Feb '11

12 Feb '11

Outbound mail is rejected because the "from" address is "1000@localcomputername". Voicemail messages are sent using the sendmail command. The mailer is actually postfix. I have set vm_mailfrom and email_from both to "workingmail(a)realdomain.com" both in conf/directory/default/1000.xml and conf/autoload_configs/voicemail.conf.xml. I have changed the voicemail.tpl so it has the literal, correct sender address. I have tried to change the "mailer-app-args" in conf/autoload_confings/switch.conf.xml to include a " -f workingmail(a)realdomain.com ". All of this is ignored, and "1000@computername" appears as the from address in the rejection notices. Can anybody point me to the correct way to set the "from" address so it is actually applied to outbound emails? The sendmail command works fine from the command line. This is my first post. Let me know if I am not following the rules. Thanks, Roger Young

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call getting hangup after git update
by ovvenkat 12 Feb '11

12 Feb '11

Hi, Today, I have updated freeSwitch to latest git. After that, When I am trying to do outbound call, Call is getting hangup. When I check the fs_cli logs, its showing that, *mod_dptools.c:2610 Originate Failed. Cause: NO_ROUTE_DESTINATION* Here is the logs for the same http://pastebin.freeswitch.org/14951 I have added STD code , before the number, Still no luck. I am getting error like *Originate Failed. Cause: NORMAL_UNSPECIFIED* Here is the log after adding STD code http://pastebin.freeswitch.org/14952 Regards, Venkat. -- If you have come to help me, you are wasting your time. If you have come to because your liberation is bound up in mine, we can work together. Regards Venkatesan OV.

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Re: [Freeswitch-users] How to set up a DTMF callback in C or C#?
by Kris 12 Feb '11

12 Feb '11

I'll try to put it on the list again. I don't use any of those functions, but there is a way to do it somehow with delegates(the C# pointers). I use the API like this. I don't know much..3 months ago I didn't know anything about C# or FreeSwitch. Just have to read and try things since there aren't many C# examples out there. FreeSWITCH.Native.Api fsApi = new FreeSWITCH.Native.Api(); string filename; filename = @"E:\LiveMatch\Sounds\en\us\female\conference\conf-help_menu.vox"; fsApi.Execute("uuid_displace", ev_all.GetHeader("Caller-Unique-ID") + " start " + filename + " 10 mux"); ----- Original Message ----- From: "Frankie Yiu" <yky1628(a)yahoo.com> To: <kris(a)livecall.com> Sent: Friday, February 11, 2011 7:03 PM Subject: Re: [Freeswitch-users] How to set up a DTMF callback in C or C#? Thanks for your reply earlier, it works. For some reason, I can not reply the thread so I send you this message privately. Since you might know about using C# w/ FreeSwitch, I might ask you this: 1) If I want to use the Media bugs, how can I do this in C#? I see a method that I want to use in swig.cs: "switch_core_media_bug_add", but what is the parameters for the call back function? >From the definition in switch_types.h: switch_bool_t (*switch_media_bug_callback_t) (switch_media_bug_t *, void *, switch_abc_type_t); but what is it in C# since it does not have pointers? 2) I see that some functions have type like this "SWIGTYPE_p_p_ ..." (for example: SWIGTYPE_p_p_switch) , do these type really work (expecially for those pointer to pointer type in C/C++)? I tried to call: "switch_core_session_read_frame(SWIGTYPE_p_switch_core_session session, SWIGTYPE_p_p_switch_frame frame, uint flags, int stream_id) " but seem like when I pass in a variable for frame, when the program gets to freeswitch_wrap.cxx side, it lost its frame's memory address/place holder. Am I doing something wrong? Thanks in advance. Frankie ____________________________________________________________________________________ Bored stiff? Loosen up... Download and play hundreds of games for free on Yahoo! Games. http://games.yahoo.com/games/front

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mod_distributor and dial strings
by David Villasmil 12 Feb '11

12 Feb '11

hello all I'm playing with mod_distributor... it's nice.. but i'm having this problem: suppose i have 2 gateways i want to use in round-trip fashion.. that works fine... but suppose i must send to gw1 with a prefix and to gw2 with a different prefix... is this possible? thanks David

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hold_music var gets unset/lost by cancelled/failed att_xfer
by Fraser Redmond 11 Feb '11

11 Feb '11

I'm trying to do a second att_xfer on a call so that if the first attended transfer fails (c-leg is busy, or presses do-not-answer, or is an extn that doesn't exist) then the call could be transferred to someone else. On the first att_xfer the person on hold hears the hold_music correctly. Once that transfer is cancelled or fails: -- On any subsequent att_xfer's the person on hold just hears silence. -- If they are put on hold they just hear silence. I tried setting hold_music again for each channel after the first att_xfer, but that didn't work, so it's probably not actually a problem with hold_music per se, but some other variable/setting that decides whether to use hold_music. I also tried doing a uuid_dump before and after each attempt, but didn't notice anything too different - unless it's a matter of unsetting one of the couple of changed/new vars like: variable_originate_disposition variable_current_application variable_playback_seconds I get the feeling other variables are probably also lost by the first failed transfer as the second att_xfer has some odd things happen if the third party does answer. Haven't been able to narrow it down as closely as the hold_music, but two things I've seen happen are: -- The party that initiated the transfer gets hung up automatically (after 30 sec) -- When the party that initiated the transfer hangs up it should connect the other two parties, but instead it hung up all three Cheers, Fraser

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Re: [Freeswitch-users] 503 Maximum Calls In Progress
by Gary Chen 11 Feb '11

11 Feb '11

After restart, everything back to normal. I can make call again. I will do more tests. I will let you know if it happen again. Gary From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Collins Sent: Friday, February 11, 2011 12:54 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] 503 Maximum Calls In Progress If you stop freeswitch and restart it does it still say that or are you able to make some calls for a while before it throws that error? -MC On Fri, Feb 11, 2011 at 11:26 AM, Gary Chen <gchen00(a)insightbb.com> wrote: Yes. I did some test calls between two Cisco sip phones and they all work fine and then it just stop working. At that point, FS is still up and I can see the system information by press F keys. It just won’t accept any calls. I looked at log file and it just has that one line: 2011-02-11 17:39:58.691008 [CRIT] switch_core_session.c:1629 The system cannot create any sessions at this time. The register message is still working at that time. Gary

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Using an Empty FROM field for SIP Messages
by Matthew Fong 11 Feb '11

11 Feb '11

I am trying to setup a customer who wanted to utilize the airespring national did presence, where in they rewrite the Caller ID of outbound calls. In order to do this tho, they say I need to send a blank caller id. FreeSWITCH will send 0000000000 if the caller id is not specified. Airespring wants something formatted like INVITE sip:jungar@alpha-org.com SIP/2.0 Via: SIP/2.0/UDP zetamachine.beta-org.com:5060;branch=as82je8ei4kr Max-Forwards: 70 To: 2556112121 <sip:jungar@alpha-org.com> From: <sip:jsmith@beta-org.com>;tag=13456 Call-ID: @zetamachine.beta-org.com with the From: field looking like any of the following From: <sip:jsmith@beta-org.com>;tag=13456 From: < @208.76.54.59>;tag=13456 From: @208.76.54.59;tag=13456 The closest I got was to make FreeSWITCH to the following, but airespring won't budge. Can anyone tell me how to make the From string as airespring wants for their national DID presence service. Thanks. From: "" <sip:0000000000@208.76.54.59>;tag=UyNBXjyUUay8m

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how to set up a DTMF callback function
by Frankie Yiu 11 Feb '11

11 Feb '11

If I want to get a DTMF callBack to a function when a caller presses a key, what function should I use to set it up? Thanks, Frankie ____________________________________________________________________________________ Never miss an email again! Yahoo! Toolbar alerts you the instant new Mail arrives. http://tools.search.yahoo.com/toolbar/features/mail/

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