December 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

voicemail disk quota poll
by Tamas Cseke 05 Dec '08

05 Dec '08

Hello, I would like to use a disk quota in users' voicemail (http://jira.freeswitch.org/browse/MODAPP-173) We haven't decided yet what would be the better prompts to play to the caller when the mailbox is full. Please advice some messages! Thanks in advance, Tamas

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re gistration and calls through different user agent
by Carole O. 04 Dec '08

04 Dec '08

Hello, I have recently installed freeswitch on Opensuse 10.3. I have a question about the sofia agents which are already present in the default installation: I have 2 snom phones (each with a user1001 and user1002), I configured one such that it registered on the default port 5060 (so on the user agent internal) and the other one on the port 5080 (on the external user agent). The aim was to see what happen. Since both were able to register I guess that the directory /directory/default/ is used by both sofia agent else user1002 should not be able to register, isn't? I am confused here because I would have said that the directory /directory/default/ was just for the users that register on internal... I had a second problem: if user1001 tries to call user1002 it did not work (but the opposite is ok). Has this something to do with the dialplan? Thanks, Carole -- View this message in context: http://www.nabble.com/registration-and-calls-through-different-user-agent-t… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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mod_spidermonkey() Session is not active!
by Faisal Maqsoodi 04 Dec '08

04 Dec '08

Hi I m trying to use a Javascript application which uses mod_pocketsphinx and mod_spidermonkey. Both r loaded but this error msg is displayed: 2008-12-04 16:31:09 [ERR] inline:1 mod_spidermonkey() Session is not active! How can i remove this error? Faisal

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freeswitch enum
by Pieter Eduard 04 Dec '08

04 Dec '08

Hi, I'm trying to query freeswitch to use my bind base enum server but i'm having trouble to query the enum from Freeswitch CLI. this what i did, i add 7.6.5.4.3.2.1.e164.arpa at my dns and after restart the dns, i have this : [root@fsbox]# host -t naptr 8.7.6.5.4.3.2.1.e164.arpa localhost Using domain server: Name: localhost Address: 127.0.0.1#53 Aliases: 8.7.6.5.4.3.2.1.e164.arpa has NAPTR record 10 10 "U" "E2U+tel" "!^.*$!tel:\\12345678!" . 8.7.6.5.4.3.2.1.e164.arpa has NAPTR record 5 10 "U" "E2U+sip" "!^.*$!sip:1000@test.com!" . and if use the ip, the result is the same : [root@fsbox]# host -t naptr 8.7.6.5.4.3.2.1.e164.arpa 1.2.3.4 Using domain server: Name: 1.2.3.4 Address: 1.2.3.4#53 Aliases: 8.7.6.5.4.3.2.1.e164.arpa has NAPTR record 5 10 "U" "E2U+sip" "!^.*$!sip:1000@test.com!" . 8.7.6.5.4.3.2.1.e164.arpa has NAPTR record 10 10 "U" "E2U+tel" "!^.*$!tel:\\12345678!" . But if i try to query from the freeswitch CLI that's installed in the same box, i get this : freeswitch@fsbox> enum 12345678 localhost API CALL [enum(12345678 localhost)] output: No Match! freeswitch@fsbox> enum 12345678 1.2.3.4 API CALL [enum(12345678 1.2.3.4)] output: No Match! even if i change the enum.conf.xml dns root with my enum ip and reload the freeswitch i still get the same error. Can anyone help me on this? i just want the fs to query on my local enum or to query on different enum server. Regards, -Pieter-

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screen_bit is always true
by Piotr Sobolewski 04 Dec '08

04 Dec '08

Hi I was trying to create extension in which I would check whether privacy=full in Remote-Party-ID header is set. So I made this. <condition field="${screen_bit}" expression="^true$" break="never"> <action application="set" data="origination_caller_id_number=Anonymous"/> </condition> But screen_bit is always true, regardless RPID privacy value. mod_dialplan_xml.c:117 parse_exten() Regex: [tooser] ${screen_bit}(true) =~ /^true$/ As it wasn't strange enough, in cdr_csv screen_bit take "true" or "false" (as it should). Is it a bug or me doing something wrong? -- regards Piotr Sobolewski

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Fax and Freeswitch: What is the status, what works?
by Dennis 04 Dec '08

04 Dec '08

hi, because we do not get tired of testing and playing a lot with the beloved fs, we now arrived at the fax feature :-) i am not sure if the docs are up to date or if there was a lot of development in the meantime. therefore i would like to ask, what is possible and what will come in the near future. we are using fs, socket outbound and php and would like to make something like fax to mail as an additional service. is t38 supported? can i pass incoming faxes over the same socket as calls? can i convert faxes into pdf? is fax over sip reliable (as far as i have heard, under asterisk fax is nothing one should use)? and so on, and so on.... i would be very happy to hear some user experiences with fs and fax. if it seems, that we can use fax with over socket outbound, we will do hardcore testing ;-) thanks, dennis

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How to get info from the b-leg
by regs＠kinetix.gr 03 Dec '08

03 Dec '08

Hi, I am making a simple bridge between two call legs : Client --(a-leg)--> FS --(b-leg)-->Provider How can I get information like network-address of the Provider, media-address, port used, media port used etc. from the second leg (b-leg)? Is all the information provided by the a-leg available for the b-leg as well? If, yes how can I access it? (and log it to my CDR file eventually)

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Redirect calls between FS
by Barray McKee 03 Dec '08

03 Dec '08

Hello, I am implementing 2 load balancing FS behind a pair of sip proxies. Since traffic is routed to one of the two FS on a round-robin basis, I need a mechanism to reroute call from one FS to another FS under a specific special circumstance. I am think to use the bridge command with bypass_media = true, so that the call can just pass through one FS and reach the second FS. Is this the approach that I should use? I am wondering if there is a better approach out there that I just can't think of. Thanks in advance, Barray

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Placing call to remote extension
by Gab Tai 03 Dec '08

03 Dec '08

Hi everyone, I am Gab and just joined the group. Also, I am new to FS but want to learn and delve into the dept as fast as possible. I have one last mile question and was hoping I could pick from someone's wealth of knowledge and understanding of the platform. I have setup FS with 5 extensions as follows: 1.. 1 extension [UA(a)] locally registered on the same network NET(a) as the realm of the FS(a) 2.. 2 extensions [UA(b1) & UA(b2) ] remotely registered to FS(a) from subnet B 3.. 2 extensions [UA(c1) & UA(c2) ] remotely registered to FS(a) from subnet C 4.. I am not using any provider Current situation a.. All remote extensions can call UA(a) and transfer media (voice) b.. UA(a) cannot call remote extensions. Error message "Sofia cannot open channel,; user not registered". But, please note that the user is actually registered. c.. Remote extensions UA(b1) cannot call UA(b2) and cannot call UA(c1) nor UA(c2) d.. Remote extensions UA(c1) cannot call UA(c2) and cannot call UA(b1) nor UA(b2) Need a.. How do I place call to remote extensions from local extensions? b.. How do I bridge media between 2 remote extensions, registered to FS(a) from same network or different network? I hope this is not too much for a starter but would greatly appreciate any thoughts and/or guidance. Sincere regards to all. Gab

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CDR generated on maximum sessions reach
by Lachezar Valchev 03 Dec '08

03 Dec '08

Hello everybody, I am new to the list and I hope I can find some help here, regarding an issue I am experiencing with the CDRs written by Freeswitch. The thing is, I am using the "max-sessions" and the "sessions-per-second" parameters in switch.conf.xml to limit the maximum number of simultaneous calls, I want to go through my Freeswitch server. These options are working well, but I was expecting to have CDRs generated for the calls, that are dropped when the limit is reached. Unfortunately there is no such one. My question is: Is there an option, which allows the generation of CDR, when a call is dropped, because the maximum sessions limit is reached? If there is no such option, is there any way to achieve it? Probably by using the the mod_limit module? Can you, please tell me how to do it? Any kind of advice is welcomed. Thank you in advance. Regards, Lachezar

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