- FreeSWITCH-users - FreeSWITCH Mailing Lists

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

3 4

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

5 7

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)

3 9

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

2 1

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

2 1

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

3 3

Bridging from Event Socket API
by Klaus Teller 03 Dec '08

03 Dec '08

Hi Folks, so far i could understand how to bridge calls with Javascript. I'm trying to do the same with Java via the Socket Interface. My first trials weren't successful. maybe you can help me understand what is goin on. What i want to do is to bridge an existing leg (Unique-ID is known) to a party that wasn't yet dialed (Unique-ID unknown). With javascript it is something like: session.bridge("sofia/internal/1002"); How do i do this using the event socket interface? what application/command would i use with which arguments? One way i tried to do this is to orginate a call to 'sofia/internal/1002' and bridge the two existing legs using uuid_bridge. Unfortunately, it wasn't successful. The only message i had on the FS console is: 2008-12-02 16:57:34 [DEBUG] switch_core_session.c:693 switch_core_session_queue_private_event() Send signal sofia/internal/1001(a)192.168.1.121 [BREAK] Any idea what i'm missing? Thanks, Klaus. -- Psssst! Schon vom neuen GMX MultiMessenger gehört? Der kann`s mit allen: http://www.gmx.net/de/go/multimessenger

3 5

Unstable att_xfer
by x y 03 Dec '08

03 Dec '08

Hy! I have tested several times so far the att_xfer function of the freeswitch, and I've found it unstable. I'm using a similar code to the example at the freeswitch wiki, and sometimes a call isnt transfered. The att_xfer works fine in the cases as I figured out from the log, when the switch_ivr_bridge comes back with a "ending bridge by request from read function" message. When it comes back with a "request from write function", the two channels dont get bridged together: they cant hear each other form the other side, and they dont even realize that the other hangs up. I have pasted some logs about this a week ago into paste bin, but if you wish, I could paste new logs with the newest fs. Could it has something with silence, because I think there are more failures when I dont talk into the phones? Im using latest fs, and sip phones without built in transfer. Cheers, Viktor Hirdetés (x) Váltson most olcsóbb kötelezőre a biztosítás-hu-val. www.biztositas.hu - a kötelező biztosítások kiindulópontja!

6 22

TLS receiving calls
by matrim 02 Dec '08

02 Dec '08

Hi, I'm having problems using TLS to receive calls. I'm using a Nokia N95 to test TLS against freeswitch. I can register my client against freeswitch and make outbound calls to the test numbers (e.g. 9999). I can also make calls to other users registered over UDP. However if I try to make a call to a user registered over TLS the leg of the call to that user always goes via UDP. e.g. 1000 registered via TLS 1001 registered via TLS 1002 registered via UDP 1003 registered via UDP 1000 -> 1002 works ok 1003 -> 1002 works ok 1001 -> 1000 Doesn't work. The leg of the call between freeswitch and 1000 tries to setup via UDP 1002 -> 1000 Doesn't work. The leg of the call between freeswitch and 1000 tries to setup via UDP === >From looking at some of the documentation it seems to me that the issue may be with the "tls-bind-params" being "transport=tls". The phone I'm using doesn't add the "transport=tls" parameter, and only uses "sips:" to specify that the connection is via TLS. I tried setting "tls-bind-params" to a blank string but it didn't change anything. Is there any way to receive calls over TLS if you don't specify "transport=tls" in your contact string during registration? According to RFC3261 the use of the "transport=tls" parameter isn't recommended anymore and is now deprecated. -- View this message in context: http://www.nabble.com/TLS-receiving-calls-tp20771637p20771637.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

7 11

Re: [Freeswitch-users] Support for Junghanns duoBRI
by Sergey Kirillov 02 Dec '08

02 Dec '08

Cool. Thanks for the answer. > All HFC-based cards supported by bristuffed Zaptel should work. > > Stefan

1 0