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Question about execute_on_answer
by Helmut Kuper 13 Sep '10

13 Sep '10

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I have a xml dialplan which arms a session with execute_on_answer "lua on_answer.lua" on_anwer.lua is this: direction = session:getVariable("Call-Direction") uuid = session:get_uuid() session:execute("log", "NOTICE INSERT "..direction..", "..uuid.."\n") session:execute("info") When a call is answered the script is executed, but direction is empty. Top of info is this: 2010-09-13 19:09:23.583688 [INFO] mod_dptools.c:1003 CHANNEL_DATA: Channel-State: [CS_EXECUTE] Channel-Call-State: [ACTIVE] Channel-State-Number: [4] Unique-ID: [40e982cd-052c-4718-9567-7d64a937ba12] Call-Direction: [inbound] Presence-Call-Direction: [inbound] Channel-Presence-ID: [2850(a)85.16.246.6] Channel-Call-UUID: [40e982cd-052c-4718-9567-7d64a937ba12] Answer-State: [answered] Channel-Read-Codec-Name: [G722] Channel-Read-Codec-Rate: [16000] Channel-Write-Codec-Name: [G722] Channel-Write-Codec-Rate: [16000] Caller-Direction: [inbound] Caller-Username: [2850] Caller-Dialplan: [XML] Caller-Caller-ID-Name: [] Caller-Caller-ID-Number: [2850] Caller-ANI: [2850] Caller-Destination-Number: [4919] Caller-Unique-ID: [40e982cd-052c-4718-9567-7d64a937ba12] Caller-Source: [mod_sofia] Caller-Context: [et_internal_error] Caller-RDNIS: [4919] Caller-Profile-Index: [2] Caller-Profile-Created-Time: [1284397763581765] Caller-Channel-Created-Time: [1284397711332733] Caller-Channel-Answered-Time: [1284397713461801] Caller-Channel-Progress-Time: [1284397711963763] Caller-Channel-Progress-Media-Time: [1284397713461801] Caller-Channel-Hangup-Time: [0] Caller-Channel-Transfer-Time: [0] variable_direction: [inbound] When I try to get "direction:" (last in info) the I got a value. So are not all variables listed in info are readable in execute_on_answer script? regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFMjl0d4tZeNddg3dwRArjBAJ99nO1xHcIciCt2OGhE8Qi4iFHglwCeOtxG vIpIFn2l1A6bsnXa9jWmzBE= =V/E6 -----END PGP SIGNATURE-----

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FS sigfaults when client connects
by Ben Sanford 13 Sep '10

13 Sep '10

I've got a Freeswitch server running on a Seagate Dockstar segfaulting when any client connects to it. Core dump is way too large to post as an attachment (54MB), you can grab it from mediafire: http://www.mediafire.com/?0197kd8zhaqbq2b. I've got the latest build that "make current" gives me yet this still happens. Turning on console debugging is worthless as the server crashes as soon as a client attempts to connect. Thanks for the help in advance. - Ben

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Re: [Freeswitch-users] get lcr channel variable from esl
by Nigel Kent 12 Sep '10

12 Sep '10

My code is set to receiving events as type plain. Changing the event type to xml makes the existing code not work . Is there a way I can use lcr with is scenario? --- On Fri, 10/9/10, Rupa Schomaker <rupa(a)rupa.com> wrote: From: Rupa Schomaker <rupa(a)rupa.com> Subject: Re: [Freeswitch-users] get lcr channel variable from esl To: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> Date: Friday, 10 September, 2010, 13:13 It is streamed to you over ESL. On Fri, Sep 10, 2010 at 6:54 AM, Nigel Kent <ktngl(a)yahoo.co.uk> wrote: thanks I meant how does Freeswitch save "as xml" data and what is the way to access it --- On Fri, 10/9/10, Rupa Schomaker <rupa(a)rupa.com> wrote: From: Rupa Schomaker <rupa(a)rupa.com> Subject: Re: [Freeswitch-users] get lcr channel variable from esl To: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> Date: Friday, 10 September, 2010, 11:19 On Fri, Sep 10, 2010 at 5:27 AM, Nigel Kent <ktngl(a)yahoo.co.uk> wrote: Hi I am not clear when you say api version because I am calling lcr through the esl @con.execute method, do you mean when used from event socket. API version is the version that is used via event socket and is what you get when called from the fs_cli or console. If that is the case then I think I would need to look into the as xml option because I am doing everthing from event socket. Right. What is does 'default' set and which module is needed to handle xml because I have not used xml for any other part of my code lcr 15555551212 default as xml the "default" is the lcr profile you are using. If you don't specify one then default is the profile used. However, in order to specify "as xml" a profile name must be specified. In this case I suggested default. As for module to process the xml, that depends on what language you are using on the client side, no? --- On Thu, 9/9/10, Rupa Schomaker <rupa(a)rupa.com> wrote: From: Rupa Schomaker <rupa(a)rupa.com> Subject: Re: [Freeswitch-users] get lcr channel variable from esl To: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> Date: Thursday, 9 September, 2010, 23:21 The api version of mod_lcr doesn't give you lcr_auto_route. You can do lcr 15555551212 default as xml and get the lcr info as xml which can then be parsed on the client side. To build the equivalent of lcr_auto_route you can concatenate all the dialstrings separated by "|". I'd suggest transfering to the lcr/ endpoint instead. lcr/15555551212 The only way to get the channel var set is to use the application version of lcr. On Thu, Sep 9, 2010 at 2:49 PM, Nigel Kent <ktngl(a)yahoo.co.uk> wrote: I have used mod_lcr application in my esl script, digits are passed to mod_lcr and a dial string is found successfully as seen on the debug log. My question is how do I acces the variable ${lcr_auto_route} from esl. Previously in my script I have used this method to get variables which was set earlier in the script @con.api("uuid_getvar", "#{@uuid} myvarible").getBody However this is not working for lcr_auto_route. If I do(a)con.api("uuid_getvar", "#{@uuid} lcr_auto_route").getBody I get a _undef_ error -- -Rupa -----Inline Attachment Follows----- _______________________________________________ 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 _______________________________________________ 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 -- -Rupa -----Inline Attachment Follows----- _______________________________________________ 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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what happened to iax
by covici＠ccs.covici.com 12 Sep '10

12 Sep '10

I have a need to connect to an asterisk box running iax -- what happened to mod_iax I can't find it anywhere? Thanks. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici(a)ccs.covici.com

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call rejected?
by covici＠ccs.covici.com 12 Sep '10

12 Sep '10

Hi. I was trying to use a gateway to galaxyvoice, and I have registered correctly. However I get a 403 response from them. It does work in asterisk with the following additional parameters from-user=<user name>, auth-user=<user-name> and auth=md5. I tried putting the from-user and auth-username explicitly in the gateway parameters, but no joy. Any ideas would be appreciated. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici(a)ccs.covici.com

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Re: [Freeswitch-users] gTalk new phone calls feature
by Malay Thakershi 11 Sep '10

11 Sep '10

I saw two sections written on google and fs. Few questions: 1. Do I have to build to latest build? 2. I am familiar with sofia endpoints. How do I use google endpoint or extn defined in conf files to actually receive and originate calls? More interested in drawing parallels between the two. 3. What is limit of inbound and outbound calls simultaneously? Thank you all. On Sep 8, 2010 1:58 PM, "mazilo" <Nabble(a)slickdeals.endjunk.com> wrote: João Mesquita-2 wrote: > > Possible is, but I don't think that Tony or any core developer is goin... If anyone is interested, perhaps revisit my previous http://freeswitch-users.2379917.n2.nabble.com/Asterisk-equivalent-Bridge-fu… post which is about making the Google Voice DialOut Python Scripts to work with FS. ----- don't and stop are the ONLY two 4-letter words considered offensive to men, but not when used... View this message in context: http://freeswitch-users.2379917.n2.nabble.com/gTalk-new-phone-calls-feature… Sent from the freeswitch-users mailing list archive at Nabble.com. ________________________________... FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mail...

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flashing problems
by covici＠ccs.covici.com 11 Sep '10

11 Sep '10

Hi. I am using mostly the default freeswitch configs, but I cannot seem to make flash work properly. I was dialing out from a freetdm extension over a freetdm fxo channel and got bridged to a number. Then I had another phone call into that same fxo line. I may be doing something wrong -- newbie that I am -- but I could not answer the second call -- I heard the call waiting noise, flashed the hook, which did get recognized, but how do I answer that call? If I were dialing out through sip, how would I do this in that case? Thanks in advance for any suggestions. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici(a)ccs.covici.com

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analogue flash/recall
by Brian May 11 Sep '10

11 Sep '10

On 10 September 2010 08:06, Moises Silva <moises.silva(a)gmail.com> wrote: > on-hook/off-hook done quickly. (changed the subject) The that is correct. Often it is called "Recall" or "PBX Recall". It is required if you want to transfer calls etc from analogue phones without the PBX listening for DTMF tones inband. On Asterisk, the idea is if you push this button, you can the dial tone, and can make a new call while the old call is on hold. If you push it again you get a conference call, and then if you hang up you have effectively made an attended transfer of the call. > Brian, what is it that does not work with flash? When I push the flash button, I get loss of audio for a moment, and then the current call resumes. Admittedly I am not entirely sure what to expect with Freeswitch, but would be nice if I could use it to transfer calls. Same as with Asterisk. Although I never could get this working with Asterisk either, I ended up having to push the off-hook button manually, which is risky (push it for too long and you disconnect the call) and not always possible with modern phones. Changing the time intervals never seemed to help. -- Brian May <brian(a)microcomaustralia.com.au>

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Re: [Freeswitch-users] External caller-number not shown on forward to external number
by Peter Waldheim 10 Sep '10

10 Sep '10

Thanks for the tipps, Michael. It seems there is a (provider-)feature called "CLIP / no screening" which had to be activated. But now that it is active, it still does not work. I'm rather sure that the variables are set correctly, cause it works perfectly well for numbers of our range. Also (one of) the test-number(s) I use is just one bit outside of our range, so the format should be fine. Still for such numbers the callee always sees our base number. I also found a simple cli test for this: originate {origination_caller_id_number=<test-caller-id-number>}openzap/1/a/<target-number> &park() Is it possible that there is some spoof-protection somewhere in the FreeSWITCH/openzap/libpri stack, or am I doing somethign wrong somehow? Any further help would be very welcome. Thanks and regards Peter Von: Michael Collins [mailto:msc@freeswitch.org] Gesendet: Freitag, 27. August 2010 18:55 An: FreeSWITCH Users Help Betreff: Re: [Freeswitch-users] External caller-number not shown on forward to external number Put in some log apps or the info app prior to your transfer and bridge apps. Make sure that these variables actually contain what you hope they do. Also, check with the telco to make sure that they allow you to pass customized caller ID like this. Some support it and some don't, but sometimes you have to ask explicitly to have the feature allowed. Finally, make sure that you are passing the phone number in a format that the telco wants, i.e. do they require the leading 1. -MC On Thu, Aug 26, 2010 at 3:32 AM, Peter Waldheim <peter.waldheim(a)framesoft.com<mailto:peter.waldheim@framesoft.com>> wrote: Hi all, In a diaplan I'm forwarding a call to an external (PRI) number when it fails to reach my SIP-Client: <action application="bridge" data="user/${dialed_extension}@${domain_name}"/>  <action application="set" data="outbound_caller_id_number=${caller_id_number}"/> <action application="set" data="outbound_caller_id_name=${caller_id_name}"/> <action application="transfer" data="#external-number# XML default"/>  Then in that dialplan: <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/> <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/> <action application="bridge" data="openzap/1/a/$1"/>  So far this works just fine but one thing: the external party only sees the original caller-id if it is an internal one. If the caller was external, I only see the default caller-number of my installation. Ideally I'd like to always see the original caller id - especially if it is an external number. Can anybody please give me a hint whether I'm missing something or this might be some call-spoof protection in freeswitch or openzap or libpri or my PRI provider? Thanks a lot Peter ________________________________ Framesoft AG Software Applications Sumpfstrasse 15 6301 Zug Switzerland Handelsregister des Kantons Zug: CH-170.3.022.876-2 Pr?sident & Vorsitzender der Gesch?ftsleitung: Toralf Dittmann Framesoft AG Software Applications Reuterweg 49 D-60323 Frankfurt am Main Germany HRB 34142 Vorsitzender des Aufsichtsrates: Toralf Dittmann Vorstand: Jens Saarholz Framesoft Ltd. Business Address: 60 Lombard Street London EC3V9EA, UK Registered Office: Hackwood Secretaries Limited 1 silk Street London EC2Y 8HQ, UK Company Number: 4055017 Directors: Toralf Dittmann, Volker Weber Secretary: Volker Weber Confidentiality Notice: The information contained in this e-mail is intended for the named recipient(s) only. It may contain privileged and confidential information, and if you are not the addressee or the person responsible for delivering this to the addressee, you may not copy, distribute or take action in reliance on it. If you have received this e-mail in error, please notify us immediately by returning the original message to the sender by e-mail and delete this message. Thank you for your cooperation. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ________________________________ Framesoft AG Software Applications Sumpfstrasse 15 6301 Zug Switzerland Handelsregister des Kantons Zug: CH-170.3.022.876-2 Pr?sident & Vorsitzender der Gesch?ftsleitung: Toralf Dittmann Framesoft AG Software Applications Reuterweg 49 D-60323 Frankfurt am Main Germany HRB 34142 Vorsitzender des Aufsichtsrates: Toralf Dittmann Vorstand: Jens Saarholz Framesoft Ltd. Business Address: 60 Lombard Street London EC3V9EA, UK Registered Office: Hackwood Secretaries Limited 1 silk Street London EC2Y 8HQ, UK Company Number: 4055017 Directors: Toralf Dittmann, Volker Weber Secretary: Volker Weber Confidentiality Notice: The information contained in this e-mail is intended for the named recipient(s) only. It may contain privileged and confidential information, and if you are not the addressee or the person responsible for delivering this to the addressee, you may not copy, distribute or take action in reliance on it. If you have received this e-mail in error, please notify us immediately by returning the original message to the sender by e-mail and delete this message. Thank you for your cooperation.

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Siptapi and Freeswitch
by Peter P GMX 10 Sep '10

10 Sep '10

Anybody tried siptapi with freeswitch? http://sourceforge.net/projects/siptapi/ This may enable Click2Dial e.g. from Outlook to Freeswitch. So anybody has experience with that solution? Best regards Peter

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