August 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

T.38 pass-through
by Tamas 10 Aug '08

10 Aug '08

Hello, I'm trying to make T.38 SIP working: PSTN-->Cantata IMG-->FS--->CW Unfortunately this does not work. I've set both the late negotiation and proxy_media. When I leave out FS from the chain, the fax receipt works well, also when I set bypass_media in FS, it works too, so something might be wrong with the proxy mode. Cantata and FS box are on the same switch so this is not a network issue. No NAT involved in the whole scenario. Could anybody give an advice? Thanks in advance, Tamas ps: FreeSwitch Version 1.0.trunk (9233M) Linux mt-sw1 2.6.18-12-fza-amd64 #1 SMP Sun May 18 09:57:08 UTC 2008 x86_64 GNU/Linux Debian Lenny

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user access
by Roberto Hernandez 09 Aug '08

09 Aug '08

Anyone have suggestions on how to the enable the following scenario? Allow internal but control external per user. More importantly how to disable or enable PSTN access for a specific user in a domain (profile)? user1(a)domain1.com <--> FS <--> Allowed PSTN user1(a)domain1.com <--> FS <--> user2(a)domain1.com user2(a)domain1.com <--> FS <--> Not Allowed PSTN user2(a)domain1.com <--> FS <--> user1(a)domain1.com Thanks, Roberto

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if i were at cluecon ...
by Wasim Baig 09 Aug '08

09 Aug '08

if i were at cluecon, i'd wish for ... .... mod_isup (and let it use mod_wanpipe for mtp2) ... and maybe someone will listen, just maybe ... -- wasim h. baig | principal consultant | convergence pk | +92 300 8508070 p.s. feel free to tack on your own wish ...

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seeking script to test FS loads (can be PAID work)
by Brian B 09 Aug '08

09 Aug '08

We're wanting to test our very particular FS conference configuration, 1st on a small VPS slice and then on a dedicated server. To do that, we have written our requirements for a testing script, to generate outbound SIP calls. Is anyone interested in getting involved? This could be paid work, or for the thrill of pushing the pedal to the metal. Ping me and I'll send you details. -Brian 415 308425 8 freeswitch(a)brian-burt.com

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dialplan of multi profile
by Lee JJ 09 Aug '08

09 Aug '08

Dear All Guru : 1. I am curious what's the relation between alias "name", "type" ? Does it affect the dialplan routing call ? How to know more inside of this ? [INFO] mod_dialplan_xml.c:222 dialplan_hunt() Processing 1526->1528@default ^^^^^^^^^ From the FAQ: Q: How do I assign endpoints to contexts with different sets of extensions Here are the different possible approaches: - Use 1 profile per context you want to route to (each one needs a distinct ip:port) [root@localhost dialplan]# netstat -aup Active Internet connections (servers and established) Proto Recv-Q Send-Q Local Address Foreign Address State PID/Program name udp 0 0 210.24y.y.y:sip *:* 3748/freeswitch udp 0 0 210.24x.x.x:sip *:* 3748/freeswitch udp 0 0 210.24y.y.y:5070 *:* 3748/freeswitch udp 0 0 210.24x.x.x:5070 *:* 3748/freeswitch udp 0 0 210.24x.x.x:5080 *:* 3748/freeswitch Sofia status still the same Name Type Data State ================================================================================================= internal profile sip:mod_sofia@210.24x.x.x:5060 RUNNING (0) external profile sip:mod_sofia@210.24x.x.x:5080 RUNNING (0) nat profile sip:mod_sofia@210.24x.x.x:5070 RUNNING (0) default alias internal ALIASED 210.24x.x.x alias internal ALIASED inter2 profile sip:mod_sofia@210.24y.y.y:5060 RUNNING (0) 210.24y.y.y alias inter2 ALIASED outbound alias external ALIASED ================================================================================================= 2. <anti-action application="bridge" data="user/${dialed_ext}@$${domain}"/<user/$%7Bdialed_ext%7D@$$%7Bdomain%7D%22/>> --- this syntax working <anti-action application="bridge" data=" sofia/profilename/${dialed_ext}@$${domain}"/<sofia/profilename/$%7Bdialed_ext%7D@$$%7Bdomain%7D%22/>> ---- while this not ! What is the syntax difference ? FYI: setting files sip profiles use the default config , and modify the correspoding parameter profile name , context , domain [root@localhost sip_profiles]# ls -F ./ ../ external/ external.xml inter2.xml internal/ internal.xml nat/ nat2.xml nat.xml  <profile name="inter2" domain="210.243.126.72">  <aliases> <alias name="210.243.126.72"/> <alias name="inter2"/> </aliases> [root@localhost dialplan]# ls -F ./ default.xml default.xml.old extensions/ inter2.xml ../ default.xml.simpl-test features.xml public.xml the most simple dialplan setting [root@localhost dialplan]# more default.xml <?xml version="1.0" encoding="utf-8"?>  <include> <context name="default">  <extension name="Local_Extension"> <condition field="destination_number" expression="^(1[0-9][0-9][0-9])$"> <action application="set" data="dialed_ext=$1"/> <action application="export" data="dialed_ext=$1"/> </condition> <condition field="destination_number" expression="^${caller_id_number}$"> <anti-action application="set" data="call_timeout=30"/> <anti-action application="set" data="hangup_after_bridge=true"/> <anti-action application="bridge" data=" user/${dialed_ext}@$${domain}"/ <user/$%7Bdialed_ext%7D@$$%7Bdomain%7D%22/>>  <anti-action application="answer"/> </condition> </extension> </context> </include> ------------------------------ *From:* Jengjr Lee ( 李正之 ) *Sent:* Thursday, August 07, 2008 6:23 PM *To:* 'freeswitch-users(a)lists.freeswitch.org' *Subject:* dialplan of two profile users Dear all : I would like to create 2 profiles which represent 2 company. There will be the same extension no in these 2 profiles. For example , one is "default" , another is "inter2". I am not familiar with the dialplan file , command, logic , control ; don't know how to control the call from these 2 profile extensions. Or is there any more info explain it deeply ? thanks Name Type Data State ================================================================================================= internal profile sip:mod_sofia@210.x.x.x:5060 RUNNING (0) external profile sip:mod_sofia@210.x.x.x:5080 RUNNING (0) nat profile sip:mod_sofia@210.x.x.x:5070 RUNNING (0) default alias internal ALIASED 210.x.x.x alias internal ALIASED inter2 profile sip:mod_sofia@210.y.y.y:5060 RUNNING (0) 210.y.y.y alias inter2 ALIASED outbound alias external ALIASED ================================================================================================= 4 profiles 4 aliases freeswitch@localhost> sofia_contact 1526(a)210.x.x.x API CALL [sofia_contact(1526(a)210.x.x.x)] output: sofia/internal/1526@219.8.z.z:15421

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Re: [Freeswitch-users] Where to find freeswitch.jar
by Adeel Ansari 08 Aug '08

08 Aug '08

Hi folks, I am looking to download freeswitch.jar, but couldn't find it till now. May be its hidden somewhere, or my mind is quite exhausted right now. Please help. Thanks. -- Best, Adeel Ansari http://www.linkedin.com/in/adeelansari

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web voice mail interface / web-vm.tpl
by John Wehle 08 Aug '08

08 Aug '08

The short version (to answer my own question) is: a) Enable mod_xml_rpc. b) Enable mod_shout. c) Change file-extension in voicemail.conf.xml to mp3. d) The URL is: http://my_ip:8080/api/voicemail/web -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john(a)feith.com | | John Wehle | Fax: 1-215-540-5495 | | -------------------------------------------------------------------------

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Getting user-agent string at time of call
by Ashutosh 07 Aug '08

07 Aug '08

Hi, I m doing user dialplan processing through mod_xml_curl . I understand that FS passes some pre-defined params to the mod_xml_curl handler script like dnis,dnid,callerid, but i wonder if i can also get the user-agent string of the user who is trying to make a call. This way, i want to be able to dynamically create the codec string to be used for the call according to the UA being used. (eg: AMR codec for nokia handsets etc. BTW, is AMR support avaialble yet in FS ?) Thanks very much! -ashutosh

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Documentation suggestion
by Robert Dyck 07 Aug '08

07 Aug '08

I have recently become aware of Freeswitch and I see much potential. Obviously there are huge gaps in the documentation. There are a few example setups and they are great but they suggest we are seeing only the tip of the iceberg. I believe what is needed is a list of all applications and channel variables, their purpose and the module which must be loaded to implement them if not in the core. Then the new user can begin to customize the examples. If this information already exists, kindly point me in the right direction. Thank you

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Anonymous checkout
by mayamatakeshi 07 Aug '08

07 Aug '08

Hello, I'm trying to update my sources, but now http://svn.freeswitch.org is asking for usr/pass. It never did this before. Is this on purpose? Should we register for them? Regards, Takeshi

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