July 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

contrib directory location
by Lars Zeb 15 Jul '09

15 Jul '09

What is the address of the contrib directory? I would like to download it and its contents for study. Thanks, Lars

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copy and past "oops" in mod_event_socket.c
by Andy Spitzer 15 Jul '09

15 Jul '09

Woof! Too simple to open a JIRA with a patch (and it actually works as written): 1804: } else if (!strncasecmp(cmd, "nolinger", 6)) { That should be an 8 as nolinger is 8 characters long. --Woof!

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leg_timeout
by Phillip Jones 15 Jul '09

15 Jul '09

Hi there, Here is my call flow: 1) leg A is bridged to leg B 2) when leg B is answered I play a confirm script - "please 1 to accept this call" I only want leg B to ring 20 seconds. BUT when the caller party answers, he should have as long as he needs to press 1. "leg_timeout" seems to be in play until the bridge is completed. I need it to reset when leg b is answered. I tried resetting the leg_timeout in the confirm script after leg b is answered. I also tried using leg_progress_timeout. Neither seemed to work. Any help or suggestions would be welcome. Phillip Jones

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How to set the IVR of VM menu??
by Brad Tuan 15 Jul '09

15 Jul '09

How to set the date format , and the IVR flow ........??

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Originate in Dial plan
by Dome Charoenyost 15 Jul '09

15 Jul '09

Dear sir, I want to user dialplan callback to customer. is posible to to this is dialplan XML ? Now i use javascript. my call flow. 1. customer call 2. FS rining and wait until customer hangup 3. callback to customer number Best Regards. Dome C.

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Re: [Freeswitch-users] Freeswitch-users Digest, Vol 37, Issue 90
by Kareem Hamdy 15 Jul '09

15 Jul '09

Thanks Michael, but I'm setting up a T1, not a PRI. I should be able to use all 24 channels. -----Original Message----- From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of freeswitch-users-request(a)lists.freeswitch.org Sent: Wednesday, July 15, 2009 1:06 AM To: freeswitch-users(a)lists.freeswitch.org Subject: Freeswitch-users Digest, Vol 37, Issue 90 Send Freeswitch-users mailing list submissions to freeswitch-users(a)lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request(a)lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner(a)lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of Freeswitch-users digest..." Today's Topics: 1. Re: OpenZAP and FreeSWITCH w/ Sangoma (Michael Collins) 2. GXW4104 & FreeSwitch (DigiLord) 3. Re: GXW4104 & FreeSwitch (Brian West) 4. SIP Trace Option at Runtime (Muhammad Shahzad) 5. Re: SIP Trace Option at Runtime (Jason White) 6. Re: Get voicemail messages (Eli Hayun) 7. How to set the IVR of VM menu?? (Brad Tuan) ---------------------------------------------------------------------- Message: 1 Date: Tue, 14 Jul 2009 17:24:18 -0700 From: Michael Collins <msc(a)freeswitch.org> Subject: Re: [Freeswitch-users] OpenZAP and FreeSWITCH w/ Sangoma To: freeswitch-users(a)lists.freeswitch.org Message-ID: <87f2f3b90907141724q2735fac1jdacea3994db62782(a)mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" See inline comments On Tue, Jul 14, 2009 at 5:04 PM, Kareem Hamdy <Kareem.Hamdy(a)trustvesta.com>wrote: > Hello: > > I would like to connect a Sangoma T1 card to FreeSWITCH. Most of the docs > I see pertain to a PRI. When I leave out the d-chan notation, I get errors > regarding not able to get the d-chan up and running in the CLI. > > Here's my info: > > [span wanpipe T1] > trunk_type => t1 > b-channel => 1:1-24 b-channel => 1:1-23 d-channel => 1:24 > > > [span wanpipe T2] > trunk_type => t1 > b-channel => 2:1-24 > set up like span 1 example > > ---- > > #================================================ > # WANPIPE1 Configuration File > #================================================ > # > # Date: Wed Dec 6 20:29:03 UTC 2006 > # > # Note: This file was generated automatically > # by /usr/local/sbin/setup-sangoma program. > # > # If you want to edit this file, it is > # recommended that you use wancfg program > # to do so. > #================================================ > # Sangoma Technologies Inc. > #================================================ > > [devices] > wanpipe1 = WAN_AFT_TE1, Comment > > [interfaces] > w1g1 = wanpipe1, , TDM_VOICE_API, Comment > > [wanpipe1] > CARD_TYPE = AFT > S514CPU = A > CommPort = PRI > AUTO_PCISLOT = NO > PCISLOT = 1 > PCIBUS = 6 > FE_MEDIA = T1 > FE_LCODE = B8ZS > FE_FRAME = ESF > FE_LINE = 1 > TE_CLOCK = NORMAL > TE_REF_CLOCK = 0 > > TE_HIGHIMPEDANCE = NO > LBO = 0DB > FE_TXTRISTATE = NO > MTU = 1500 > UDPPORT = 9000 > TTL = 255 > IGNORE_FRONT_END = NO > TDMV_SPAN = 1 > TDMV_DCHAN = 0 TDMV_DCAHN = 24 > > TDMV_HW_DTMF = YES > TDMV_HW_FAX_DETECT = NO > > [w1g1] > ACTIVE_CH = ALL > TDMV_HWEC = YES > MTU = 80 > > > --- > > > > In freeSWITCH's openzap.conf.xml, I've tried pri_span as well as analog. > > I cannot find a straight up T1 wiki anywhere. Would someone please provide > an example? > > > Thanks, > Kareem > > _______________________________________________ > 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 >

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Preventing disconnect on event_sockte close
by Weaver, Eric 15 Jul '09

15 Jul '09

Using mod_event_socket in outbound mode, is there any to prevent a call from being disconnected when the outbound socket is closed ? I would like to handle the initial inbound call using outbound but after the disposition of the call is determined, close the socket and have that call managed using an inbound socket instead. Eric Weaver

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fs_cli - display variable values?
by Lars Zeb 15 Jul '09

15 Jul '09

Is it possible to display the value of a variable in fs_cli? I tried "echo ${domain_name}", but it just echoed what I typed (${domain_name}), rather than its value. I do not know how to get help on an individual command from the help facility in fs_cli. I tried fs_cli itself and also the docs but could find nothing. Thanks, Lars

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Re: [Freeswitch-users] Get voicemail messages
by Eli Hayun 15 Jul '09

15 Jul '09

did you bind your lua script to directory lookups in addition to the dialplan? On Tue, Jul 14, 2009 at 7:02 AM, Eli Hayun <elihayun(a)gmail.com> wrote: > > Hi > > I am not using fixed xml files for the extension registration. I have > > LUA script to return an XML string to FS. > > Everything goes fine until I am trying to get the voice messages. > > When am entering my id, FS (or voicemail module) try to get the xml for > > that id, but it cant find it. My lua script did NOT recieved any xml > > request at that point. > > What should I do to solve the problem. > > > > Thanks > > Eli Hayun > > > Yes I did bind it: my lua.conf.xml is like this <configuration name="lua.conf" description="LUA Configuration"> <settings> <param name="xml-handler-script" value="GenXml.lua"/> <param name="xml-handler-bindings" value="directory"/> </settings> </configuration> When an extension tried to register, I have no problem. But when I want to use VoiceMail to retrieve my messeges, I got a problem. Here is the partial log: 2009-07-15 07:44:49.373089 [INFO] mod_dialplan_xml.c:252 Processing Phone2->*98 in context default 2009-07-15 07:44:49.386466 [NOTICE] mod_dptools.c:649 Channel [sofia/internal/80671(a)132.64.3.86] has been answered 2009-07-15 07:44:51.933664 [WARNING] mod_voicemail.c:2072 Can't find user [80671(a)132.64.3.86] 2009-07-15 07:44:52.533435 [NOTICE] switch_core_state_machine.c:179 Hangup sofia/internal/80671(a)132.64.3.86 [CS_EXECUTE] [NORMAL_CLEARING] 2009-07-15 07:44:52.545698 [NOTICE] switch_core_session.c:1085 Session 3 (sofia/internal/80671(a)132.64.3.86) Ended 2009-07-15 07:44:52.545698 [NOTICE] switch_core_session.c:1087 Close Channel sofia/internal/80671(a)132.64.3.86 [CS_DESTROY]

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GXW4104 & FreeSwitch
by DigiLord 15 Jul '09

15 Jul '09

Hello all, I am getting my feet wet with FreeSwitch by migrating my Asterisk box over. I have run into a few things that I am not sure how to accomplish. I have a Grandstream GXW4104 with one analog line connected. I have it connected and I am able to receive calls on my Polycom 501 (ext 2101) that is registered to the FreeSwitch server. The one problem is that CallerID is not the CallerID from the caller, it's the CallerID from the Grandstream device (ext 2100). On the same device there is HORRIBLE echo. I have set echo cancellation on the device to enabled and disabled to no avail. Under Asterisk there was no echo. I setup the device as a provider. Was that the right way to accomplish connecting this device to FS? Is there a way to enable sending an e-mail containing my voicemail messages like Asterisk does? Thanks in advance for any help you can give! Dan

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