July 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

From domain
by Alex Gusak 24 Jul '08

24 Jul '08

We have sheme: +-----+ +----+ +-----+ | GW1 |-->| FS |-->| GW2 | +-----+ +----+ +-----+ <gateway name="GW1" ip="X.X.X.X"> <param name="caller-id-in-from" value="true"/> <param name="realm" value="domain.net"/> <param name="from-domain" value="domain.net"/> <param name="username" value="1"/> <param name="password" value="1"/> <param name="register" value="false"/> </gateway> <gateway name="GW2" ip="X.X.X.X"> <param name="caller-id-in-from" value="true"/> <param name="realm" value="domain.net"/> <param name="from-domain" value="domain.net"/> <param name="username" value="1"/> <param name="password" value="1"/> <param name="register" value="false"/> </gateway> When call from GW1 to GW2 FreeSWITCH recv INVITE from GW1: From: "Alex"<sip:alex_gusak@domain.net>;transport=UDP;tag=7552bb34 To: <sip:XXXXXX@domain.net>;transport=UDP and send INVITE to GW2: From: "Alex" <sip:alex_gusak@X.X.X.X>;tag=BK71NZ3yZvptK To: <sip:XXXXXX@domain.net> where X.X.X.X IP address FreeSWITCH. How to make that pass in the header From: the domain name domain.net? -- Alex Gusak

3 10

Custom sip headers
by Simon Capper 24 Jul '08

24 Jul '08

Is there a way to add custom headers to the sip messages sent from mod_sofia? Simon

2 1

lua, sound playback and interruption on DTMF entry
by Alex Kinch 24 Jul '08

24 Jul '08

Hi all, I've got a lua script that plays a sound file then listens for DTMF. What I'd like to do is interrupt the sound file playback and process the DTMF if the user starts entering digits whilst the sound file is playing. At the moment they have to wait til the sound file has finished before entering digits. Any suggestions? Here's the script snippet.. ession:streamFile("/usr/local/freeswitch/sounds/dk/gsm/vm-enter-num-to- call.gsm"); session:streamFile("/usr/local/freeswitch/sounds/dk/gsm/vm-then- pound.gsm"); digits = session:getDigits(20, "#", 3000); Thanks, Alex

3 4

problems with dingaling
by Alex Gusak 24 Jul '08

24 Jul '08

We have some problems with dingaling. When call from SIP to GTALK, the connection is established, but clients do not hear each other. During connection comes message: <iq type="error" to="alex.gusak(a)gmail.com/gtalk37B4CA2D" id="304" from="Gennadiy.Lunkov(a)gmail.com/Talk.v105176AFA53"> <session type="transport-info" id="4768150905" initiator="alex.gusak(a)gmail.com/gtalk37B4CA2D" xmlns="http://www.google.com/session"> <transport xmlns="http://www.google.com/transport/p2p"> <info_element name="caller_id_number" value="alex_gusak" xmlns="http://www.freeswitch.org/jie"></info_element> <info_element name="caller_id_name" value="Alex" xmlns="http://www.freeswitch.org/jie"></info_element> <candidate name="rtp" address="X.X.X.X" port="17762" username="XXXXXXX" password="XXXXXXXX" preference="1.0" protocol="udp" type="local" network="0" generation="0"></candidate> </transport> </session> <error code="501" type="cancel"> <feature-not-implemented xmlns="urn:ietf:params:xml:ns:xmpp-stanzas"></feature-not-implemented> </error> </iq> Jingle profile looks like this: <profile type="client"> <param name="name" value="gmail.com"/> <param name="login" value="alex.gusak(a)gmail.com/gtalk"/> <param name="password" value="XXXXXXXXXX"/> <param name="server" value="talk.google.com"/> <param name="context" value="from_google"/> <param name="dialplan" value="XML"/> <param name="message" value="FreeSwitch"/> <param name="rtp-ip" value="X.X.X.X"/> <param name="auto-login" value="true"/> <param name="auto-reply" value="Press"/> <param name="use-rtp-timer" value="true"/> <param name="sasl" value="plain"/> <param name="tls" value="true"/> <param name="vad" value="both"/> <param name="exten" value="100"/> </profile> Dialplan: <extension name="out"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^(.*)$"> <action application="bridge" data="dingaling/gmail.com/gennadiy.lunkov@gmail.com"/> </condition> </extension> FreeSWITCH has public IP. Where could be the problem? -- Alex Gusak

1 0

execute_on_answer problem
by Erol Akarsu 24 Jul '08

24 Jul '08

Hi , I have started using freeswitch and impressed with it, how easy to set up and run quick examples. It is awesome! What I do here is I bridge the call between IVR and a remote phone. Then remot eparty pick up the phone and, IVR plays a prompt. I want the remote user listen all prompt content. If we executed these 2 action inside my dialmap extension , playback action don't wait until remote end pick up the phone. Therefore, the user misses the beginning part of prompt. <action application="answer"/> <action application="playback" data="/home/eakarsu/kurumcell.g729"/> Now I started using LUA scripts. When I execute " originate sofia/gateway/voipwise/001804XXXYYYY(a)sip.voipwise.com 1238" from FS command line, I am getting error and calls dropped. I have attached log file where 'nua_r_cancel: unknown event 32: 200 Ok" debug statement is inserted. My dialmap extension and kurumcell.lua script is here. call is dropped. I apprecciate if you can guide me on this. <extension name="kurumcell_test"> <condition field="destination_number" expression="^1238$"> <action application="set" data="execute_on_answer=kurumcell.lua"/> </condition> </extension> kurumcell.lua: -- answer the call session:answer(); -- play a file session:streamFile("/home/eakarsu/kurumcell.g729"); -- hangup session:hangup(); 2008-07-23 20:52:25 [DEBUG] switch_core_state_machine.c:140 switch_core_standard_on_execute() sofia/external/0018042489925(a)sip.voipwise.com Execute set(execute_on_answer=kurumcell.lua) 2008-07-23 20:52:25 [DEBUG] mod_dptools.c:715 set_function() SET [execute_on_answer]=[kurumcell.lua] 2008-07-23 20:52:25 [NOTICE] switch_core_state_machine.c:157 switch_core_standard_on_execute() Hangup sofia/external/0018042489925(a)sip.voipwise.com [CS_EXECUTE] [NORMAL_CLEARING] 2008-07-23 20:52:25 [DEBUG] switch_channel.c:1360 switch_channel_perform_hangup() Kill sofia/external/0018042489925(a)sip.voipwise.com [KILL] 2008-07-23 20:52:25 [DEBUG] switch_core_session.c:676 switch_core_session_signal_state_change() Kill sofia/external/0018042489925(a)sip.voipwise.com [BREAK] 2008-07-23 20:52:25 [DEBUG] switch_core_state_machine.c:430 switch_core_session_run() (sofia/external/0018042489925(a)sip.voipwise.com) State EXECUTE going to sleep 2008-07-23 20:52:25 [DEBUG] switch_core_state_machine.c:365 switch_core_session_run() sofia/external/0018042489925(a)sip.voipwise.com Running State Change CS_HANGUP 2008-07-23 20:52:25 [DEBUG] switch_core_state_machine.c:393 switch_core_session_run() (sofia/external/0018042489925(a)sip.voipwise.com) State HANGUP 2008-07-23 20:52:25 [DEBUG] mod_sofia.c:264 sofia_on_hangup() Channel sofia/external/0018042489925(a)sip.voipwise.com hanging up, cause: NORMAL_CLEARING 2008-07-23 20:52:25 [DEBUG] mod_sofia.c:300 sofia_on_hangup() Sending CANCEL to sofia/external/0018042489925(a)sip.voipwise.com 2008-07-23 20:52:25 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() Standard HANGUP sofia/external/0018042489925(a)sip.voipwise.com, cause: NORMAL_CLEARING 2008-07-23 20:52:25 [DEBUG] switch_core_state_machine.c:393 switch_core_session_run() (sofia/external/0018042489925(a)sip.voipwise.com) State HANGUP going to sleep 2008-07-23 20:52:25 [DEBUG] switch_core_session.c:735 switch_core_session_thread() Session 6 (sofia/external/0018042489925(a)sip.voipwise.com) Locked, Waiting on external entities 2008-07-23 20:52:25 [NOTICE] switch_core_session.c:753 switch_core_session_thread() Session 6 (sofia/external/0018042489925(a)sip.voipwise.com) Ended 2008-07-23 20:52:25 [NOTICE] switch_core_session.c:755 switch_core_session_thread() Close Channel sofia/external/0018042489925(a)sip.voipwise.com [CS_HANGUP] 2008-07-23 20:52:25 [DEBUG] sofia.c:319 sofia_event_callback() nua_r_cancel: unknown event 32: 200 Ok 2008-07-23 20:52:25 [DEBUG] sofia.c:190 sofia_event_callback() event [nua_r_invite] status [487][Request terminated] session: n/a 2008-07-23 20:52:25 [DEBUG] sofia.c:190 sofia_event_callback() event [nua_i_state] status [487][Request terminated] session: n/a

2 1

ring groups
by Star Wind 24 Jul '08

24 Jul '08

Who can tell me, how I can create a static ring group? I'm using third-party SBC, which changes "Contact" field on every registration, every 15 minutes. That's why I can't use "add-group" and "del-group" dialplan features, because new "Contact" field generates randomly. All I need - simply ring group, with ring scheme like this: <phone 1 rings...> - timeout - <phone 2 rings...> - timeout - ... and so on.

4 6

Timeout parameter in originate
by Ashutosh 24 Jul '08

24 Jul '08

Hi, We see originate function as follows: result = new_session.originate(session, dest[[[[[[[[[[, dialplan], context], cid_name], cid_num], network_addr], ani], aniii], rdnis], username], to]); Where to= timeout When does the timeout counter start, after call to the originate Api, or after the remote end has picked up the call ? Thanks, ashutosh

1 0

Operator/Receptionist console
by Brian Snipes 24 Jul '08

24 Jul '08

Ok.. I need some suggestions. Currently I have a couple of snom 370 phones with 2 sidecars on each phone for the receptionist and her backup location. Those phones monitor the status of ~55 extensions. The Snoms seem to be having a devil of a time doing this. Several times a day they get REALLY slow. This happens consistently also right after the phone is rebooted and the subscriptions start showing up. By slow, I mean I can hit 3 or 4 keys before anything happens on the display and when it starts filling in the display it is slow. If I remove all subscription monitoring from the phones they speed along like I would expect them to do. Pulling on all of your vast wisdom - what options do I have? What phones will work properly with that many subscriptions in a sip environment? On another note - I am thinking about creating an application similar to FOP for the desktops and need suggestions on the best way to do this. This won't be anything fancy. I only need it to alert whether a phone is in use or not. Here are the ideas so far: 1. Query existing call db (maybe the limit table) every minute. Dump results of extensions in use to an html file. Clients view html file and it is set to refresh every 15 seconds. < I don't really like this too much > 2. Have something similar to another product I've used. What it did was send events to a modified irc daemon and a windows client listened to the channel and updated the extension buttons with their current status. Thanks! Brian

4 8

safe_freeswitch (like safe_asterisk): restarting FS automatically?
by Birgit Arkesteijn 24 Jul '08

24 Jul '08

Hi all, We've got an older version of FreeSWITCH (Trunk 7948) running on a Linux x86_64 machine. At the moment it's crashing few times a day, making our services very unreliable. At the moment we don't have the time to rebuild this version, so I'm looking for an equivalent of the "safe_asterisk" script. This script runs Asterisk in a loop, restarting it when it goes down. I couldn't find any equivalent script, but maybe I using the wrong keywords in my search. Does anyone know if such a script is available, and (if so) could you point out where I can find it? Thanks, Birgit -- -- Birgit Arkesteijn, birgit(a)westhawk.co.uk, -- Westhawk Ltd, Albion Wharf, 19 Albion Street, Manchester M1 5LN, UK -- Company no: 1769350 -- Registered Office: -- 15 London Road, Stockton Heath, Warrington WA4 6SJ. UK. -- tel.: +44 (0)161 237 0660 -- <URL: http://www.westhawk.co.uk>

6 9

Documentation questions
by Michael Collins 23 Jul '08

23 Jul '08

Guys, I was just looking over the source directory an I noticed that the common files for an open source project may need some serious attention: AUTHORS INSTALL NEWS README I'm sure that the COPYING file is just fine. NEWS and README are completely empty. Should we maybe just put references in them that point to interesting spots on the freeswitch.org site? Does anyone have suggestions for content in these files? Should we delete them altogether? INSTALL basically just says to do the "./config && make && make install" shuffle. Is that adequate or should we possibly add a link to the GSG on the wiki? AUTHORS looks pretty good except that maybe BKW should be moved up under MikeJ and referenced as the tester/QA/etc. person for FS. Also, for those who have contributed in the past but who are not current contributors should we have a separate section that says something like, "We'd like to thank the past contributors for their efforts:" and then list guys like Bret McDanel and Josh Culp? Last question - could those in the know look to see if there are any glaringly obvious omissions? As far as OpenZAP goes I will work on these same files and come up with the requisite additions. (I think I'm unofficially the official OZ documentation guy.) Please give us your $0.02 so we can act accordingly. Thanks, MC

1 0