Re: [Freeswitch-users] Unable to register UA
Your SIP UA needs to take the info in the 401 and use it to digest authenticate. If you trace a SIP UA that supports authentication you will see that they also get the 401/407 and only then are able to authenticate. This is just a fact of how digest auth works in SIP ... see section 22.4 The Digest Authentication Scheme: http://www.ietf.org/rfc/rfc3261.txt On November 11, 2009 04:58:28 pm Fede wrote: Hi! I'm trying to register a SIP UA to my FreeSWITCH server and for some reason I always get a 401 Unauthorized response. I've tried with other UA (X-Lite and Ekiga) and they do work. The UA is: http://www.doddlephone.com My user configuration is: include user id=doddle params param name=password value=doddle/ param name=vm-password value=1234/ /params variables variable name=toll_allow value=domestic,international,local/ variable name=accountcode value=doddle/ variable name=user_context value=default/ variable name=effective_caller_id_name value=doddle/ variable name=effective_caller_id_number value=doddle/ /variables /user /include Can someone help me and tell me what I'm doing wrong? Here's the FreeSWITCH trace if it's useful: freeswi...@fc1160102.aspadmin.net tport_wakeup_pri(0xae7056b0): events IN tport_recv_event(0xae7056b0) tport_recv_iovec(0xae7056b0) msg 0xae703f88 from (udp/216.75.60.102:5060) has 444 bytes, veclen = 1 recv 444 bytes from udp/[190.179.3.18]:4375 at 21:51:10.194004: REGISTER sip:216.75.60.102 SIP/2.0 From: sip:dod...@216.75.60.102 sip%3adod...@216.75.60.102;tag=633f3915 To: sip:dod...@216.75.60.102 sip%3adod...@216.75.60.102 Call-Id: 186e708700bcf9a944855105fc3dce0e Cseq: 101 REGISTER Contact: sip:dod...@192.168.0.1:4375;LINEID=fc9c0e49caad Expires: 3600 Date: Wed, 11 Nov 2009 21:51:51 GMT Max-Forwards: 70 User-Agent: Doddle WebPhone Supported: replaces Via: SIP/2.0/UDP 192.168.0.1:4375;branch=z9hG4bK-0f9263f10caa;rport Content-Length: 0 tport_deliver(0xae7056b0): msg 0xae703f88 (444 bytes) from udp/ 190.179.3.18:5060/sip next=(nil) nta: received REGISTER sip:216.75.60.102 SIP/2.0 (CSeq 101) nta: Via check: received=190.179.3.18 nta: canonizing sip:216.75.60.102 with contact nta: REGISTER (101) going to a default leg nua: nua_stack_process_request: entering nua: nh_create: entering nua: nh_create_handle: entering nua: nua_stack_set_params: entering soa_clone(static::0xae7098a0, 0xae704890, 0xae718fc8) called soa_set_params(static::0xae719398, ...) called nua: nua_application_event: entering nua: nua_respond: entering nua(0xae718fc8): sent signal r_respond nua: nua_handle_destroy: entering nua(0xae718fc8): sent signal r_destroy nua: nua_stack_set_params: entering nua: nua_handle_magic: entering nua: nua_handle_destroy: entering soa_set_params(static::0xae719398, ...) called tport_tsend(0xae7056b0) tpn = UDP/190.179.3.18:4375 tport_resolve addrinfo = 190.179.3.18:4375 tport_by_addrinfo(0xae7056b0): not found by name UDP/190.179.3.18:4375 tport_vsend returned 631 send 631 bytes to udp/[190.179.3.18]:4375 at 21:51:10.198733: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.0.1:4375 ;branch=z9hG4bK-0f9263f10caa;rport=4375;received=190.179.3.18 From: sip:dod...@216.75.60.102 sip%3adod...@216.75.60.102;tag=633f3915 To: sip:dod...@216.75.60.102 sip%3adod...@216.75.60.102 ;tag=U8QpcZyvrQ3Fg Call-Id: 186e708700bcf9a944855105fc3dce0e Cseq: 101 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.0.5pre5-15326M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces WWW-Authenticate: Digest realm=216.75.60.102, nonce=132239a5-e37e-4698-af61-5df9b3b67de8, algorithm=MD5, qop=auth Content-Length: 0 nta: sent 401 Unauthorized for REGISTER (101) nta: timer set to 32000 ms nta_leg_destroy((nil)) soa_destroy(static::0xae719398) called tport_wakeup_pri(0xae7056b0): events IN tport_recv_event(0xae7056b0) tport_recv_iovec(0xae7056b0) msg 0xae716af8 from (udp/216.75.60.102:5060) has 706 bytes, veclen = 1 recv 706 bytes from udp/[190.179.3.18]:4375 at 21:51:10.453646: REGISTER sip:216.75.60.102 SIP/2.0 From: sip:dod...@216.75.60.102 sip%3adod...@216.75.60.102;tag=633f3915 To: sip:dod...@216.75.60.102 sip%3adod...@216.75.60.102 Call-Id: 186e708700bcf9a944855105fc3dce0e Cseq: 102 REGISTER Expires: 3600 Date: Wed, 11 Nov 2009 21:51:51 GMT Max-Forwards: 70 User-Agent:
Re: [Freeswitch-users] Mod_pjsip
My favorite part of this 'civilized' discussion on IRC was when DelphiWord and diegoviola sat around tryin to take the piss outta stkn on this issue for seemingly no reason. Thanks for making the channel a cool place, guys ;) On October 31, 2009 07:32:03 pm Meftah Tayeb wrote: Anthony Minessale a écrit : Meftah, Feel free. thanks P.S. STKN was the guy who made the first mod_pjsip for FS that we abandoned years ago. So you should believe him. Both him and I agreed it was not working out. So if you don't believe me, find out for yourself. anthony, why i don't believe you? never say that. i believe you and all Freeswitch Staf and thank you and to all Freeswitch Staf. On Sat, Oct 31, 2009 at 6:06 PM, Meftah Tayeb tayeb.mef...@gmail.com mailto:tayeb.mef...@gmail.com wrote: hi Anthony i agry i say that because STKN hate all my suggestions. about pjsip, i will contribute aditional module in the contrib. thanks Anthony Anthony Minessale a écrit : Meftah, He is 100% correct. Please do not insult my volunteer developers. Without help from him you would not have any FreeSWITCH right now so please drop this subject we are not using pjsip. On Sat, Oct 31, 2009 at 5:44 PM, Meftah Tayeb tayeb.mef...@gmail.com mailto:tayeb.mef...@gmail.com wrote: hi, Pjsip support ICE, STUN and TURN! to STKN: if you don't pjsip, please stop talking or exit the discution we want to kype Freeswitch Clean and universal Stefan Knoblich a écrit : Michael S Collins wrote: I can guarantee that the FS devs are well aware of pj-sip. If it was/ is a viable alternative then it would be considered. The fact that it isn't being used is a pretty good indication that it isn't suitable for FS at this time. -MV Sent from my iPhone We already mentioned some of the reasons why it did get dropped 3 years ago (first two points from memory, last two from old IRC logs): [License incompatible (GPL), but i think tony tried to negotiate on alternate license terms] Not possible to have multiple SIP profiles (due to global variables being used in the lib). A race-condition under high load, that couldn't be resolved back then (with the help of the pjsip developers). And the sofia module just working and surviving the scalability tests, so all efforts were focussed on mod_sofia and pjsip got dropped. stkn ___ FreeSWITCH-users mailing list FreeSWITCH-users@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-user s http://www.freeswitch.org __ Information from ESET NOD32 Antivirus, version of virus signature database 4539 (20091024) __ The message was checked by ESET NOD32 Antivirus. http://www.eset.com __ Information from ESET NOD32 Antivirus, version of virus signature database 4539 (20091024) __ The message was checked by ESET NOD32 Antivirus. http://www.eset.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com mailto:msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com mailto:paypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net http://irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org mailto:sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 http://iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org mailto:googletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org mailto:FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
Re: [Freeswitch-users] Freeswitch in signaling path only
Do you have a debug level message from switch_ivr_originate.c in your log? Channel is already up, delaying proxy mode 'till both legs are answered. Set bypass_media b4 each bridge. It is unsetting on you and setting bypass_media_after_bridge because you already answered the channel running the lua script. On October 30, 2009 12:03:29 pm DJB wrote: I am wondering why I cannot do as condition#2. For Lua in dialplan, when I have the followings: --WORKING-- (Condition#1) . . session:execute(set,bypass_media=true) session:execute(set,hangup_after_bridge=true) session:execute(set,continue_on_fail=true) . . session:execute(bridge,sofia/external/ .. called_num .. @1.1.1.1|sofia/external/ .. called_num .. @1.1.1.2) . . --NOT WORKING-- (Condition#2) Note: FS tries to be in media path and send re-invite. . . session:execute(set,bypass_media=true) session:execute(set,hangup_after_bridge=true) session:execute(set,continue_on_fail=true) . . session:execute(bridge,sofia/external/ .. called_num .. @1.1.1.1) session:execute(bridge,sofia/external/ .. called_num .. @1.1.1.2) . . Thank you, Dorn B. ___ FreeSWITCH-users mailing list 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 ___ FreeSWITCH-users mailing list 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
Re: [Freeswitch-users] Determining Frame Size
Depends on your codec. Frame size = ptime. So it can vary from call to call, but most you will find it to be 20ms. So if we are using speex-wb at 20ms, each frame is 0.02 seconds long. 50 frames = 1 second of sound. On October 28, 2009 12:37:56 pm Matthew Fong wrote: I noticed that the wait_for_silence's silence_hits listen_hits use number of frames as base values (rather than seconds or milliseconds). I'd like to wait for 5 seconds of consistent silence, but I'm wondering how I can determine the sample rate (and thus frames per second) for an existing call. Is there a variable I can read? Thanks. --matt ___ FreeSWITCH-users mailing list 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
Re: [Freeswitch-users] sched_api doesn't get launched
poor bbhenry :)
Added in r15207 please test and update docu if necessary:
http://wiki.freeswitch.org/wiki/Variable_api_on_answer
On October 23, 2009 11:02:07 am Anthony Minessale wrote:
it's probably related to escaping the data.
I was sick of watching you suffer so i added api_on_answer variable to
trunk.
On Fri, Oct 23, 2009 at 3:54 AM, Henry Huang
red.rain.se...@gmail.comwrote:
Thanks to c6burns on IRC channel for the tip to use execute_on_answer in
combination with eval, and of course everyone here that pointing me to
the right direction.
I was able to execute sched_api with eval, but not with the combination
of execute_on_answer. The argument just don't get parsed as the eval
argument. Here is the code:
setVariable(execute_on_answer, eval sched_api(+%ld none uuid_displace
some uuid start
/usr/local/freeswitch/sounds/en/us/callie/misc/8000/final.wav 20 mux));
Here is the log:
2009-10-23 16:17:53.633189 [DEBUG] switch_scheduler.c:214 Added task 2
sched_api_function (none) to run at 1256285888
EXECUTE sofia/internal/1688...@192.168.1.66 *eval(+OK *Added: 2
)
2009-10-23 16:18:04.52701 [DEBUG] switch_channel.c:1927 sofia/internal/
1688...@192.168.1.66 *execute on answer: eval(sched_api(+37 none
uuid_displace 84d74f41-8668-4138-943f-f076e94046ad start
/usr/local/freeswitch/sounds/en/us/callie/misc/8000/final.wav 20 mux))*
EXECUTE sofia/internal/1688...@192.168.1.66 eval(sched_api(+37 none
uuid_displace 84d74f41-8668-4138-943f-f076e94046ad start
/usr/local/freeswitch/sounds/en/us/callie/misc/8000/final.wav 20 mux))
The first part of the log is when I just do execute(eval,
${sched_api(+%ld none uuid_displace some uuid start
/usr/local/freeswitch/sounds/en/us/callie/misc/8000/final.wav 20 mux)});
And I simple get a OK, and then later on the sched_api gets executed.
However, on the excerpt of the second log. which I use the combination
with execute_on_answer like specified in the yellow hight code above. It
doesn't simply return a OK, instead it shows the whole blob of code. and
does nothing later.
On Thu, Oct 22, 2009 at 4:09 AM, Anthony Minessale
anthony.miness...@gmail.com wrote:
Yes you need an API object as described in other email.
Which line of code from java caused that segfault
It looks like a simple NULL string issue that we may want to hunt down.
On Wed, Oct 21, 2009 at 4:44 AM, Henry Huang
red.rain.se...@gmail.comwrote:
I can't seem to find the right thing to use in mod_java to execute api
commands, only api_after_bridge
2009-10-21 17:42:46.593094 [NOTICE] mod_sofia.c:1509 Pre-Answer
sofia/internal/1688...@192.168.1.66!
#
# A fatal error has been detected by the Java Runtime Environment:
#
# SIGSEGV (0xb) at pc=0x004e4480, pid=1927, tid=16116624
#
# JRE version: 6.0_16-b01
# Java VM: Java HotSpot(TM) Client VM (14.2-b01 mixed mode linux-x86 )
# Problematic frame:
# C [libc.so.6+0x6f480] strcpy+0x10
#
# An error report file with more information is saved as:
# /usr/local/freeswitch/bin/hs_err_pid1927.log
2009-10-21 17:42:59.883729 [ERR] switch_core_session.c:1374 Invalid
Application sched_api
2009-10-21 17:42:59.883729 [NOTICE] switch_core_session.c:1375 Hangup
sofia/internal/1688...@192.168.1.66 [CS_EXECUTE]
[DESTINATION_OUT_OF_ORDER]
#
# If you would like to submit a bug report, please visit:
# http://java.sun.com/webapps/bugreport/crash.jsp
# The crash happened outside the Java Virtual Machine in native code.
# See problematic frame for where to report the bug.
On Sat, Oct 17, 2009 at 3:37 AM, Michael Collins
m...@freeswitch.orgwrote:
On Fri, Oct 16, 2009 at 11:53 AM, Henry Huang
red.rain.se...@gmail.com
wrote:
So how would you trigger it from a script dialplan? The only time it
seemed to work is when I did setVariable(api_after_bridge,
sched_api blah blah blah);
but then it gets executed after the channel's been teared down. I
thought api_after_bridge means right after the call gets connected.
I need something to execute an api command right before or right
after the call gets bridged.
api_after_bridge is a channel variable, so using setVariable works
just
fine. If you need to sched_api is an API only. Check these out:
http://wiki.freeswitch.org/wiki/Mod_commands#Misc._Commands
So you need an API object in order to use it. I don't know the syntax
for creating an api obj in Java but in Lua it goes like this:
api = freeswitch.API();
res = api:execute(sched_api,+300 none my_api my_api_args)
Remember, if the method you are using isn't found in the dial plan
tools then it isn't a dial plan application. Make sure it's on the
list: http://wiki.freeswitch.org/wiki/Mod_dptools
On the other hand, API commands are listed here:
http://wiki.freeswitch.org/wiki/Mod_commands
dptools require a session object, api commands require an api
object...
-MC
Re: [Freeswitch-users] Hostname
one real quick way would be put different GET var in each server's binding On October 23, 2009 03:46:11 pm Kristian Kielhofner wrote: Can't you use different contexts or something else to tell them apart? On Fri, Oct 23, 2009 at 3:34 PM, freeswitch noob freeswitch.n...@gmail.com wrote: I have mod xml_curl installed and I am getting the following passed to my script. [hostname] = myhost.local [section] = dialplan I also have multiple versions of FS running on the same box. Is there a way to have each FS instance on my box have a unique hostname ? Thanks in advance. ___ FreeSWITCH-users mailing list 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 ___ FreeSWITCH-users mailing list 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
Re: [Freeswitch-users] Hostname
Fake the hostname in the query string: http://test/dir.php?hostname=mmmpotatoes Add this to the top of your buddy's script (if its even PHP): $hostname = $_POST['hostname'] = $_REQUEST['hostname'] = $_GET['hostname']; On October 23, 2009 06:18:41 pm Metik wrote: Forget it--this will not work because FS uses a POST (vs GET). Most likely it would attempt to actually POST to a file named index.php?xhostname=myhost. However, if there was some way to add an arbitrary POST variable to the HTTP transaction, it would work. I am sure the FS developers have already thought of this and its a case of RTFM. -metik - Original Message - From: Metik To: Metik ; freeswitch-users@lists.freeswitch.org Sent: Friday, October 23, 2009 5:48 PM Subject: Re: [Freeswitch-users] Hostname Please note that this would essentially be taking Chris' suggestion a little further but the effort involved would be minimal. - Original Message - From: Metik To: freeswitch-users@lists.freeswitch.org Sent: Friday, October 23, 2009 5:37 PM Subject: Re: [Freeswitch-users] Hostname Why not simply overwrite the value of the variable used throughout the script... -- xml_curl.conf -- ... param name=gateway-url value=http://localhost/index.php?xhostname=myhost; bindings=dialplan/ ... -- index.php -- ? $_REQUEST['hostname'] = $xhostname; ... - Original Message - From: freeswitch noob To: freeswitch-users@lists.freeswitch.org Sent: Friday, October 23, 2009 4:42 PM Subject: Re: [Freeswitch-users] Hostname Yeah, I was just trying to make it easier on myself. I have scripts from a friend that parse xml_curl requests based on the hostname, I was hoping to not have to re-write them to read something else from the post that FS makes from xml_curl. But from what it sounds like I will have to. On Fri, Oct 23, 2009 at 2:58 PM, Chris Burns ch...@cloudtel.com wrote: one real quick way would be put different GET var in each server's binding On October 23, 2009 03:46:11 pm Kristian Kielhofner wrote: Can't you use different contexts or something else to tell them apart? On Fri, Oct 23, 2009 at 3:34 PM, freeswitch noob freeswitch.n...@gmail.com wrote: I have mod xml_curl installed and I am getting the following passed to my script. [hostname] = myhost.local [section] = dialplan I also have multiple versions of FS running on the same box. Is there a way to have each FS instance on my box have a unique hostname ? Thanks in advance. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswi tch-users http://www.freeswitch.org ___ FreeSWITCH-users mailing list 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 -- ___ FreeSWITCH-users mailing list 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 ___ FreeSWITCH-users mailing list 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
Re: [Freeswitch-users] Connect PHP SOAP Web Server with SQLite database of FS
If you really wanted: http://php.net/manual/en/book.sqlite.php But I would recommend you make use of ODBC to use a client/server RDBMS. Here's some good reading: http://www.sqlite.org/cvstrac/wiki?p=WhenToUseSqlite On October 20, 2009 10:53:01 am homqua wrote: Now I am building a PHP SOAP Web Service to access the database of FS. Anyone has idea about how to access sqlite database of FS through PHP ? I have read about socket event in FS, but I don't know whether it can response with the query of database or not. Thanks for your help. ___ FreeSWITCH-users mailing list 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
Re: [Freeswitch-users] check to see if freeswitch is alive
There's a check_sip plugin for nagios if you're into that sorta thing On October 19, 2009 12:39:53 pm Michael Collins wrote: 2009/10/19 Christian Löschenkohl christian.loeschenk...@xpirio.com hello we have the problem here that our freeswitch server freezes from time to time (no sip traffic is possible any more). Monitoring is definitely important, but I'm sure the FS devs would like to know more about what happens when FS freezes - is it FS or some other process that is messing things up? If there's an issue with FS then the developers would like to know more. -MC ___ FreeSWITCH-users mailing list 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
Re: [Freeswitch-users] Polycom MWI Forgetfulness
This happens with our polycoms as well ... NAT on phone and PBX. Still haven't had time to look into it so I disabled the sound for new message waiting ... for now it doesn't keep beeping every few minutes. On September 23, 2009 08:08:39 pm Brian West wrote: NO I have never seen it happen what firmware version are you running? /b On Sep 23, 2009, at 6:38 PM, Daniel Morrigan wrote: Brian, It was set for contact. Would that cause this behavior? Daniel ___ FreeSWITCH-users mailing list 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 ___ FreeSWITCH-users mailing list 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
Re: [Freeswitch-users] Music Background
Check out the variables ringback and transfer_ringback. The local extension in the default dialplan is a good example. For romance, I recommend 80s rock ballads. YMMV. On September 11, 2009 12:22:20 pm Dome Charoenyost wrote: Dear Sir, Is posible to play music for background when call connect ? Example when i call my wife some time i need romantic song :) BG Dome C. ___ FreeSWITCH-users mailing list 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 ___ FreeSWITCH-users mailing list 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
Re: [Freeswitch-users] Music Background
There are a few ways you could go about dropping into a conference and playing the song in from a separate channel. On September 11, 2009 01:47:43 pm Dome Charoenyost wrote: 2009/9/12 Chris Burns ch...@cloudtel.com: Check out the variables ringback and transfer_ringback. The local extension in the default dialplan is a good example. Music rinback is Ok now. but I'm looking for solution for stream sound to channel both leg when call is answer. For romance, I recommend 80s rock ballads. YMMV. I'll try :) On September 11, 2009 12:22:20 pm Dome Charoenyost wrote: Dear Sir, Is posible to play music for background when call connect ? Example when i call my wife some time i need romantic song :) BG Dome C. ___ FreeSWITCH-users mailing list 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 ___ FreeSWITCH-users mailing list 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 ___ FreeSWITCH-users mailing list 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 ___ FreeSWITCH-users mailing list 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
Re: [Freeswitch-users] Confused about conferences
I couldn't imagine managing a conference without a GUI. I need to see who is
making noise so I can boot/mute em ;)
If I were you I would dive into ESL and make a simple web app to frontend
the conferences. There will surely be something in contrib to get you
started.
On Thu, Aug 13, 2009 at 8:48 AM, Bradley Brashier bjbrash...@gmail.comwrote:
So it sounds like set can work. But you'd still have to parse it. And
even then it's not recommended.
I have another couple of possible methods for you:
1) modification of mod_conference.
2) event socket.
If you modify mod_conference, you can probably do what you want, but it
obviously requires using C and modifying existing code.
If you use the event socket, you've got a bigger learning curve, perhaps,
but you can use a variety of languages, your code is separate (and therefore
easier to maintain), and you then know how the event socket works in case
you need to do something else later.
Good luck with whatever you end up doing.
BB
On Thu, Aug 13, 2009 at 12:11 AM, Alan Chandler
a...@chandlerfamily.org.uk wrote:
Bradley Brashier wrote:
I wrote:
This is a significant new fact for me. What you seem to be doing is
calling the commands referenced in the conference api here
http://wiki.freeswitch.org/wiki/Mod_conference#API_Reference
by using application=conference and then the data string as the
second
part of the command. Am I correct in the assumption that you can do
this.
I agree that that's what it looks like. What I don't know is if it
works. I got this example from the page
http://wiki.freeswitch.org/wiki/Conferencing_and_Intercom. I never did
exactly what you're trying, and never tried using the API in this
fashion.
I just found this - which I think helps
http://wiki.freeswitch.org/wiki/Mod_commands#From_the_Dialplan
An API can be called from the dialplan but it is not recommended. Example:
extension name=Make API call from Dialplan
condition field=destination_number expression=^(999)$
!-- next line calls hupall, so be careful! --
action application=set
data=api_result=${hupall(normal_clearing)}/
/condition
/extension
Anyway - thanks for you help - I am going away to rethink that
particular interface again. Its getting so complicated that it might be
better to copy the Javascript approach in the examples.
--
Alan Chandler
http://www.chandlerfamily.org.uk
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Re: [Freeswitch-users] Polycom configuration problems?
Basically read the polycom manual ... it is the polycom producing the dialtone and deciding when to dial the number you are entering, using its own dialplan and interdigit timers. On Tue, Jun 23, 2009 at 10:38 AM, Rupa Schomaker r...@rupa.com wrote: How are you configuring your polycom? On Tue, Jun 23, 2009 at 8:26 AM, Lars Zeb larc...@yahoo.com wrote: I’m sorry Chris, but I don’t know where the look for the “global sip.cfg and mac/phone specific cfg” settings. I also looked for digitmap but could find nothing. Can you be more specific? Thanks, Lars *From:* freeswitch-users-boun...@lists.freeswitch.org [mailto: freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Chris Burns *Sent:* Monday, June 22, 2009 2:57 PM *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] Polycom configuration problems? Sounds like a config issue in the dialplan/ tag. Check global sip.cfg and mac/phone specific cfg. When you are dialing on-hook I don't think it will use your .digitmap or ..digitmap.timer settings. When you dial off-hook it sure will. On Mon, Jun 22, 2009 at 3:49 PM, Lars Zeb larc...@yahoo.com wrote: I am having difficulty with a Polycom 501 and Freeswitch. There are 3 lines on the phone. The first two are registered with a SwitchVox, the last with Freeswitch. When I select the 3rd line and begin to press numbers, pressing the 3rd digit automatically causes the phone to begin to dial. It does not matter which three numbers I press, the 3rd one is magic. However, if I do not select a line before dialing and key a 10-digit number into the phone, then select the 3rd line, it dials out fine. You can see from the debug console output that Processing begins before it hits any dialplan, so that cannot be the problem. I must have the line defined incorrectly for Freeswitch. Thanks for any suggestions, Lars. PolycomSoundPointIP-SPIP_501-UA/2.1.2.0078 Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 i386 GNU/Linux 2009-06-22 12:32:18.588457 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected by acl domains. Falling back to Digest auth. 2009-06-22 12:32:18.706419 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected by acl domains. Falling back to Digest auth. 2009-06-22 12:32:18.708461 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1...@192.168.10.29 [661a46fa-5f63-11de-85d0-1157463d23c2] 2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1...@192.168.10.29) Running State Change CS_NEW 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3210 Channel sofia/internal/ 1...@192.168.10.29 entering state [received][100] 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3217 Remote SDP: v=0 o=- 1245682011 1245682011 IN IP4 192.168.10.101 s=Polycom IP Phone c=IN IP4 192.168.10.101 t=0 0 m=audio 2254 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:403 (sofia/internal/1...@192.168.10.29) State NEW 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G7221:115:32000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G7221:107:16000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[G722:9:8000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare [PCMU:0:8000:0]/[PCMU:0:8000:20] 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:2026 Set Codec sofia/internal/1...@192.168.10.29 PCMU/8000 20 ms 160 samples 2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3028 Set 2833 dtmf payload to 101 2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3376 (sofia/internal/ 1...@192.168.10.29) State Change CS_NEW - CS_INIT 2009-06-22 12:32:18.711463 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/1...@192.168.10.29 [BREAK] 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1...@192.168.10.29) Running State Change CS_INIT 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480 (sofia/internal/1...@192.168.10.29) State INIT 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:83 sofia/internal/ 1...@192.168.10.29 SOFIA INIT 2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:111 (sofia/internal/ 1...@192.168.10.29) State Change CS_INIT - CS_ROUTING 2009-06-22 12:32:18.721428 [DEBUG] switch_core_session.c:933 Send signal sofia/internal/1...@192.168.10.29 [BREAK] 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480 (sofia/internal/1...@192.168.10.29) State INIT going to sleep 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397 (sofia/internal/1...@192.168.10.29) Running State Change CS_ROUTING 2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:483 (sofia/internal/1
Re: [Freeswitch-users] Polycom configuration problems?
Sounds like a config issue in the dialplan/ tag. Check global sip.cfg and
mac/phone specific cfg. When you are dialing on-hook I don't think it will
use your .digitmap or .digitmap.timer settings. When you dial off-hook it
sure will.
On Mon, Jun 22, 2009 at 3:49 PM, Lars Zeb larc...@yahoo.com wrote:
I am having difficulty with a Polycom 501 and Freeswitch. There are 3
lines on the phone. The first two are registered with a SwitchVox, the last
with Freeswitch.
When I select the 3rd line and begin to press numbers, pressing the 3rd
digit automatically causes the phone to begin to dial. It does not matter
which three numbers I press, the 3rd one is magic.
However, if I do not select a line before dialing and key a 10-digit number
into the phone, then select the 3rd line, it dials out fine.
You can see from the debug console output that Processing begins before it
hits any dialplan, so that cannot be the problem. I must have the line
defined incorrectly for Freeswitch.
Thanks for any suggestions, Lars.
PolycomSoundPointIP-SPIP_501-UA/2.1.2.0078
Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686
i386 GNU/Linux
2009-06-22 12:32:18.588457 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected
by acl domains. Falling back to Digest auth.
2009-06-22 12:32:18.706419 [DEBUG] sofia.c:4549 IP 192.168.10.101 Rejected
by acl domains. Falling back to Digest auth.
2009-06-22 12:32:18.708461 [NOTICE] switch_channel.c:602 New Channel
sofia/internal/1...@192.168.10.29 [661a46fa-5f63-11de-85d0-1157463d23c2]
2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:397
(sofia/internal/1...@192.168.10.29) Running State Change CS_NEW
2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3210 Channel sofia/internal/
1...@192.168.10.29 entering state [received][100]
2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3217 Remote SDP:
v=0
o=- 1245682011 1245682011 IN IP4 192.168.10.101
s=Polycom IP Phone
c=IN IP4 192.168.10.101
t=0 0
m=audio 2254 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
2009-06-22 12:32:18.711463 [DEBUG] switch_core_state_machine.c:403
(sofia/internal/1...@192.168.10.29) State NEW
2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare
[PCMU:0:8000:0]/[G7221:115:32000:20]
2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare
[PCMU:0:8000:0]/[G7221:107:16000:20]
2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare
[PCMU:0:8000:0]/[G722:9:8000:20]
2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3068 Audio Codec Compare
[PCMU:0:8000:0]/[PCMU:0:8000:20]
2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:2026 Set Codec
sofia/internal/1...@192.168.10.29 PCMU/8000 20 ms 160 samples
2009-06-22 12:32:18.711463 [DEBUG] sofia_glue.c:3028 Set 2833 dtmf payload
to 101
2009-06-22 12:32:18.711463 [DEBUG] sofia.c:3376 (sofia/internal/
1...@192.168.10.29) State Change CS_NEW - CS_INIT
2009-06-22 12:32:18.711463 [DEBUG] switch_core_session.c:933 Send signal
sofia/internal/1...@192.168.10.29 [BREAK]
2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397
(sofia/internal/1...@192.168.10.29) Running State Change CS_INIT
2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480
(sofia/internal/1...@192.168.10.29) State INIT
2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:83 sofia/internal/
1...@192.168.10.29 SOFIA INIT
2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:111 (sofia/internal/
1...@192.168.10.29) State Change CS_INIT - CS_ROUTING
2009-06-22 12:32:18.721428 [DEBUG] switch_core_session.c:933 Send signal
sofia/internal/1...@192.168.10.29 [BREAK]
2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:480
(sofia/internal/1...@192.168.10.29) State INIT going to sleep
2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:397
(sofia/internal/1...@192.168.10.29) Running State Change CS_ROUTING
2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:483
(sofia/internal/1...@192.168.10.29) State ROUTING
2009-06-22 12:32:18.721428 [DEBUG] mod_sofia.c:130 sofia/internal/
1...@192.168.10.29 SOFIA ROUTING
2009-06-22 12:32:18.721428 [DEBUG] switch_core_state_machine.c:78
sofia/internal/1...@192.168.10.29 Standard ROUTING
2009-06-22 12:32:18.721428 [INFO] mod_dialplan_xml.c:252 Processing
1001-323 in context default
Dialplan: sofia/internal/1...@192.168.10.29 parsing [default-unloop]
continue=false
Dialplan: sofia/internal/1...@192.168.10.29 Regex (PASS) [unloop]
${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/internal/1...@192.168.10.29 Regex (FAIL) [unloop]
${sip_looped_call}() =~ /^true$/ break=on-false
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Re: [Freeswitch-users] Buzzing when people speak in conference
Try turning off comfort noise completely in the conference profile? My 650s sound great in conference w/ PCMU and G722 On April 1, 2009 03:10:35 pm Giovanni Maruzzelli wrote: To make a long story short, a ground loop is when an electric circuit is made between different audio device that are connected to the same electric power grid with badly grounded connections. This is an electrical problem generating noise, nothing to do with software. To test if this is the origin of your problem, try to use the devices unplugged from the electrical grid and check if the noise still there Sincerely, Giovanni Maruzzelli = www.celliax.org via Pierlombardo 9, 20135 Milano Italy gmaruzz at celliax dot org Cell : +39-347-2665618 Fax : +39-02-87390039 2009/4/1 Stromin Normin stormin.nor...@hotmail.co.uk: Cheers for the replies. I'm not sure if I'm replying properly but here goes. I'm using Polycom 650 phones. I'm not really sure what a 60hz ground loop is so will need clarification, sorry I'm new to this. The phones are all on the same LAN and the conferencing is done on internal calls. Cheers From: stormin.nor...@hotmail.co.uk To: freeswitch-users@lists.freeswitch.org Date: Wed, 1 Apr 2009 22:09:03 +0100 Subject: [Freeswitch-users] Buzzing when people speak in conference Hi, I've been asked to do some testing on Freeswitch by work, we currently use Asterisk. I'm quite new to telephony so please go easy. I have FS setup on a windows box and at the moment I'm testing internal calls only, when I transfer calls or call extensions everything sounds great. The problem occurrs when I setup conferencing, people can log in ok and we can talk, the trouble is as people start to talk a buzzing sound is heard in the background, once the talking stops the buzzing stops. If the person goes on mute there is no buzzing. Hopefully this is enough info cheers for any help. Upgrade to Internet Explorer 8 Optimised for MSN. Download Now Surfing the web just got more rewarding. Download the New Internet Explorer 8 ___ Freeswitch-users mailing list 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 ___ Freeswitch-users mailing list 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 ___ Freeswitch-users mailing list 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
Re: [Freeswitch-users] make freeswitch-snapshot
apt-get install unixodbc-dev On March 5, 2009 11:02:45 pm mashudi wrote: Hi Folk, i got error while conduct ./make freeswitch-snapshot on debian 2.6 x86 here is the error : /usr/bin/ld: cannot find -lodbc collect2: ld returned 1 exit status make[2]: *** [libfreeswitch.la] Error 1 Making all in src Making all in mod making all mod_amr make[5]: *** No rule to make target `/usr/src/freeswitch-snapshot/libfreeswitch.la', needed by `mod_amr.so'. Stop. make[4]: *** [all] Error 1 make[3]: *** [mod_amr-all] Error 1 make[2]: *** [all-recursive] Error 1 Making all in build + FreeSWITCH Build Complete ---+ + FreeSWITCH has been successfully built. + + Install by running: + + + + make install + +--+ make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 Did I miss something ? thank you for your support. mashudi * Sekarang Gratis Nelpon SLJJ Flexi diperluas ke Yogya * ___ Freeswitch-users mailing list 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 ___ Freeswitch-users mailing list 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
