- FreeSWITCH-users - FreeSWITCH Mailing Lists

Freeswitch freezes on increasing call traffic
by shehzad p 06 Feb '09

06 Feb '09

Hi all, Yesterday my Freeswitch server faced a problem when call traffic increased to more than 100. When I start Freeswitch, it works fine and then after some time (approximately 15 to 20 minutes) it stops functioning (means no call is being processed, no CLI command is working and it just freezes) until I restart the freeswitch. I am using Freeswitch 1.0.1. Debug (gdb) trace as on wiki page http://wiki.freeswitch.org/wiki/Debugging_Freeswitch#fs_debug.sh is attached http://www.nabble.com/file/p21701744/fs_debgu.txt fs_debgu.txt -- View this message in context: http://www.nabble.com/Freeswitch-freezes-on-increasing-call-traffic-tp21701… Sent from the Freeswitch-users mailing list archive at Nabble.com.

6 27

Freeswitch hangup cause code issue
by Ankit Gandhi 06 Feb '09

06 Feb '09

Hello, I want my caller to detect hangup cause as 34 so that he can try next provider according to lcr. Here is my setup. Caller -> switch (fs) -> Terminator. Now when terminator sends "503 Service Unavailable", I want to override this cause, so that the caller gets hangup case 34. (According to the terminator he is sending hangup cause 34 from his side, but in freeswitch we are getting hangup cause 41 for that call and the same hangup cause on caller side). When I tried asterisk as caller, I get hangup cause 34 in that case. But when I tried freeswitch as caller, then we are getting hangup cause 41, the same as we are getting in switch (fs). >From the switch, I tried one of this condition through javascript to override the hangup cause before sending to caller: -> session.execute("respond","503"); -> session.execute("hangup","NORMAL_CIRCUIT_CONGESTION"); -> session.execute("hangup","34"); -> session.hangup(34); In all the above cases, asterisk properly detects the hangup cause 34, but freeswitch does not detect that. It detects the same hangup cause 41 for the call. Other callers also get the same hangup cause 41 for such calls. How can I override this cause, so that the caller gets hangup cause 34 in such cases? Here is the sip trace, on the caller side returned through switch (fs). ss = switch cc = caller -------------------------------------------------------------------- U ss.ss.ss.ss:5060 -> cc.cc.cc.cc:5080 SIP/2.0 503 Service Unavailable. Via: SIP/2.0/UDP cc.cc.cc.cc:5080;rport=5080;branch=z9hG4bK8HvDU4Z88KU5m;received=122.169.29.122. From: "654321" <sip:654321@cc.cc.cc.cc>;tag=8p7aF18aN0mSj. To: <sip:123456@ss.ss.ss.ss>;tag=Nj8ypjDyvKUXe. Call-ID: fd1ab774-6ec5-122c-9fac-001cc086141d. CSeq: 110835048 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-hacked. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. Supported: timer, precondition, path, replaces. Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Reason: Q.850;cause=34;text="NORMAL_CIRCUIT_CONGESTION". Content-Length: 0. ----------------------------------------------------------------------- -- View this message in context: http://www.nabble.com/Freeswitch-hangup-cause-code-issue-tp21868781p2186878… Sent from the Freeswitch-users mailing list archive at Nabble.com.

2 1

streamFile and read doesn't get DTMF while used with originate
by shehzad p 06 Feb '09

06 Feb '09

Hi all My set up is : When i originate the call from CLI, using originate command, It comes in a dialplan and from there it goes to javascript for handling simple IVR. In IVR I tested both streamFile and dialplan read application to get the DTMF from user, but it was not working, it just play the file and does nothing (streamFile doesn't call the onInput function at all).. This setup was working before, when I call a javascript from CLI, and in JS I create a session to originate the call. Is there any settings missing for my first setup, so that I can get DTMF properly... Thank.. msp -- View this message in context: http://www.nabble.com/streamFile-and-read-doesn%27t-get-DTMF-while-used-wit… Sent from the Freeswitch-users mailing list archive at Nabble.com.

1 0

shoutcast skips
by e schmidbauer 05 Feb '09

05 Feb '09

hey everyone. just wondering if anyone has tested recording conferences at 48000h celt to a shoutcast stream or wav file. we are able to have cd quality conferences with 3 members each using the celt codec with little or no noise disturbances or skipping. but when we try to record the conference either to a wav file or to a shoutcast stream, the quality significantly decreases due to skipping or popping noises. im not sure but maybe we are having this problem because our server doesnt have the CPU power to handle reencoding on the fly like that. we are using a 2.8ghz amd64 dual core, 4gig ddr 800 as our freeswitch server. im thinking if there is a way to record the conference as a celt audio file (instead of reencoding to mp3) that may reduce the CPU power needed and therefore solve the problem or we just need a more powerful server. could anyone recommended what kind of server we would need to handle such instances as i described above? thank you.

3 10

Dialplan variables
by Nik Middleton 05 Feb '09

05 Feb '09

Hi Guys, Simple question, tried asking on IRC but no joy, they're too busy slating other systems. I'm trying to dial out via a remote sip gateway via the dial plan <extension name="dial_my_mobile"> <condition field="destination_number" expression="^0773400000$"> <action application="bridge" data="sofia/${use_profile}/0773400000(a)21X.XXX.XXX/XXX"/> </condition> </extension> This works fine, but I'd like to wild card the extension so it matches on anything starting with a 0 the a number > 0 How do I pass the number dialed using a variable? In asterisk I would put ${EXTEN} Finally I also have the sip gateway registered Mag gateway sip:xxx@hostname.net REGED Is it possible to use the name of this gateway instead of the IP address as in 21X.XXX.XXX/XXX ? Regards

2 4

spidermonkey problems
by Jonas Gauffin 05 Feb '09

05 Feb '09

Hello I got problems with hanging spidermonkey sessions and need some advice on how to debug them. I've made a javascript queue application that uses mod_spidermonkey_socket. It works fine for a while, but after some calls I noticed that calls didnt get transferred to agents. The reason was that earlier calls had not been terminated properly. freeswitch@test1> hupall 2009-01-15 12:15:04 [CRIT] switch_core_session.c:147 switch_core_session_hupall() Giving up with 8 sessions remaining API CALL [hupall()] output: +OK hangup all channels with cause MANAGER_REQUEST freeswitch@test1> show calls API CALL [show(calls)] output: 0 total. As you can see, 8 sessions are alive, but none of them are listed as calls. What kind of logs should I turn on to see what is happening with those sessions? Thanks, Jonas

5 7

FreeSWITCH calls database
by Ali Al-Rubaie 05 Feb '09

05 Feb '09

Hi, Is there a way to find some calls' statistics in FreeSWITCH, like no. of calls, durations, calls' records..etc. Thanks,

2 1

Re: [Freeswitch-users] SIP Authentication
by Ali Al-Rubaie 05 Feb '09

05 Feb '09

We're using HelpCaster softphone. The issue here is that in Digest Authentication, if the server sends the parameter "qop" in the challenge then the client should respond with the "cnonce" parameter. The parameter "qop" is optional in Digest Auth. So the question here is that, can we configure FreeSWITCH so that it will not send "qop" in the challenge? Thanks! --- On Wed, 2/4/09, freeswitch-users-request(a)lists.freeswitch.org <freeswitch-users-request(a)lists.freeswitch.org> wrote: From: freeswitch-users-request(a)lists.freeswitch.org <freeswitch-users-request(a)lists.freeswitch.org> Subject: Freeswitch-users Digest, Vol 32, Issue 39 To: freeswitch-users(a)lists.freeswitch.org Date: Wednesday, February 4, 2009, 2:05 PM Send Freeswitch-users mailing list submissions to freeswitch-users(a)lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request(a)lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner(a)lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of Freeswitch-users digest..." Today's Topics: 1. Re: SIP Authentication (Brian West) 2. Re: origainate through sofia gateway (Michael Collins) 3. Recording background music and voice is out of sync (Daniel Liang) 4. Re: Q931 decoding Update (Gopalakrishnan A.N) 5. mod_limit (Chav Paskov) 6. Re: mod_limit (Michael Collins) 7. Re: mod_limit (Chav Paskov) 8. Re: mod_limit (Michael Collins) ---------------------------------------------------------------------- Message: 1 Date: Wed, 4 Feb 2009 10:52:45 -0600 From: Brian West <brian(a)freeswitch.org> Subject: Re: [Freeswitch-users] SIP Authentication To: freeswitch-users(a)lists.freeswitch.org Message-ID: <7DAC91F6-2FD3-464D-AA81-321EBCADC8C0(a)freeswitch.org> Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes What client is this? I also notice we receive port 3458 and reply to port 1059... /b On Feb 4, 2009, at 10:17 AM, Ali Al-Rubaie wrote: > What I have noted is that the client does not send the values for > "cnonce" and "nc" in the response. I'm not sure if this is the > reason, however how this problem can be solved? > > Thanks, > > Ali ------------------------------ Message: 2 Date: Wed, 4 Feb 2009 09:41:07 -0800 From: Michael Collins <msc(a)freeswitch.org> Subject: Re: [Freeswitch-users] origainate through sofia gateway To: freeswitch-users(a)lists.freeswitch.org Message-ID: <87f2f3b90902040941r61d669aaie949aa7cc8578a9a(a)mail.gmail.com> Content-Type: text/plain; charset=ISO-8859-1 I'll make sure the substance of this is in the wiki and I'll look for references to the deprecated way and remove those. -MC On Wed, Feb 4, 2009 at 6:09 AM, Anthony Minessale <anthony.minessale(a)gmail.com> wrote: > Where did you learn how to use js this way? > session.originate is being misused here and is depricated and may be > removed. > > the first arg to session.originate is either undefined or a *different* > session (the a leg) > > session1 = new Session(); > session1.originate(undefined, > "{ignore_early_media=true}user/1008(a)192.168.1.122"); > > session1.setVariable("effective_caller_id_number","fixed0248b"); > > //once you have session1 when you originate session2 you pass session1 as > the arg > // the effective_caller_id is taken from session1 > > session2 = new Session(); > session2.originate(session1, "sofia/gateway/halonet/0225490317"); > > Anyway this whole code is depricated in favor of this: > > session1 = new Session("{ignore_early_media=true}user/1008(a)192.168.1.122"); > if (session1.ready()) { > session1.setVariable("effective_caller_id_number","fixed0248b"); > session2 = new Session("sofia/gateway/halonet/0225490317", session1); > } > > and could be further refactored down to this: > > session1 = new Session("{ignore_early_media=true}user/1008(a)192.168.1.122"); > if (session1.ready()) { > session1.setVariable("effective_caller_id_number","fixed0248b"); > session1.execute("bridge", "sofia/gateway/halonet/0225490317"); > } > > or down to this one line of code that will setup the call detached from the > script and exit. > > var result = apiExecute("originate", > "{effective_caller_id_number=fixed0248b,origination_caller_id_number=1000,ignore_early_media=true}user/1008(a)192.168.1.122 > bridge:sofia/gateway/halonet/0225490317 inline"); > > if you dont care about the result and want to exit even before the call is > completed. > > var result = apiExecute("bgapi", "originate > {effective_caller_id_number=fixed0248b,origination_caller_id_number=1000,ignore_early_media=true}user/1008(a)192.168.1.122 > bridge:sofia/gateway/halonet/0225490317 inline"); > > > > On Wed, Feb 4, 2009 at 2:51 AM, Jacek Sokulski <jsokulski(a)dotsystems.pl> > wrote: >> >> We have tried setting both effective_caller_id_number and >> origination_caller_id_number: >> >> >> session1.originate(session1,"{origination_caller_id_number=fixed0248b}sofia/gateway/halonet/0225490317",15); >> but the problem still exists. The solution we have found for the case >> when we originate two calls, local and external, is as follow: >> >> session1 = new Session(); >> session1.originate(session1,"user/1003(a)192.168.1.122",15);//local >> if(session1.ready()) { >> session1.execute("execute_extension","00930691688627 XML >> default");//external >> } >> >> so the external call goes through the dialplan. >> It does not work if both calls are external. One possible solution could >> be >> to pass the originating call through dialplan (loopback?) but we have not >> managed >> to figure out how to do it. >> >> Thanks >> Jacek >> >> Dnia 03-02-2009, wto o godzinie 14:31 -0300, Nicolas Brenner pisze: >> > Oops! Well, fortunately I don't use that voip provider anymore (nor the >> > script). >> > >> > Thanks Brian. >> > >> > Nicolas >> > >> > On Tue, Feb 3, 2009 at 2:25 PM, Brian West <brian(a)freeswitch.org> wrote: >> > > YOU should NEVER use this method or call setCallerData at all you >> > > should use the correct methods to override the callerid. >> > > >> > > If its a B-Leg born from an A-Leg you use these on the on the A-Leg: >> > > >> > > >> > > http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_name >> > > >> > > http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_number >> > > >> > > If you're originating you use this: >> > > >> > > >> > > http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_name >> > > >> > > http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_num… >> > > >> > > /b >> > >> > _______________________________________________ >> > Freeswitch-users mailing list >> > Freeswitch-users(a)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(a)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/ > > AIM: anthm > MSN:anthony_minessale@hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888@conference.freeswitch.org > iax:guest@conference.freeswitch.org/888 > googletalk:conf+888@conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ------------------------------ Message: 3 Date: Wed, 4 Feb 2009 09:43:10 -0800 From: "Daniel Liang" <Daniell(a)airg.com> Subject: [Freeswitch-users] Recording background music and voice is out of sync To: <freeswitch-users(a)lists.freeswitch.org> Message-ID: <0B02E756F603CC409EB553879B090CC80A23EBB5(a)HPEXCHVS01.exchange.airg> Content-Type: text/plain; charset="us-ascii" What I did was the following: First, I sent the playback command: call-command: execute execute-app-name: playback execute-app-arg: <filename> Then I send uuid_record (Sorry, it was not Record command): api uuid_record <uuid> start <filename> 120 I also tried replacing the playback command with: api uuid_displace <uuid> start <filename> 0 mux But the end results are the same. The recorded user's voice is about 0.5 second behind the expected result. Thanks, Daniel -----Original Message----- From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: February 3, 2009 6:36 PM To: freeswitch-users(a)lists.freeswitch.org Subject: Re: [Freeswitch-users] Recording background music and voice is outof sync Can you show us an example of how you're doing this? Playback and Record aren't async so you'll need to show us how you're doing this. Also don't hijack threads you hit replay on the one "Re: [Freeswitch- users] FreeSwitch setup as a "Dumb" SBC" as the subject, deleted the subject and started a new body. That hijacks the thread and that can cause your problem to go ignored in some cases if people aren't interested in the thread topic depending on how their reader threads the emails. Please click new message and type freeswitch- users(a)lists.freeswitch.org in and then input your subject and body to start a new thread. Thanks, Brian West FreeSWITCH.org On Feb 3, 2009, at 8:21 PM, Daniel Liang wrote: > Hi, > > I was trying to record a background music with a user's voice at the > same time. I did a playback and started recording. But the recorded > user's voice and the background music is about 0.5 second out of sync. > I also tried to use uuid_displace instead of playback, but I got the > same result. _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org <http://www.freeswitch.org/>

2 1

origainate through sofia gateway
by Jacek Sokulski 05 Feb '09

05 Feb '09

Hello I am trying to initiate a call from javascript, it works fine for local numbers: > session1.originate(session1, "{ignore_early_media=true}user/1008(a)192.168.1.122"); but when I am trying to connect through sofia gateway, the connection is not being established: > session2.originate(session2, "sofia/gateway/halonet/0225490317"); although I can call to this number from softphone. I have also tried setting effective_caller_id_number: > session1.originate(session1, "{effective_caller_id_number=fixed0248b}sofia/gateway/halonet/0225490317"); with the same result. A configuration in the dialplan that works is: > <extension name="halonet.pl"> > <condition field="destination_number" expression="^0095(\d{10})$"> > <action application="set" data="effective_caller_id_number=fixed0248b"/> > <action application="set" data="bypass_media=true"/> > <action application="set" data="hangup_after_bridge=true"/> > <action application="bridge" data="sofia/gateway/halonet/$1"/> > </condition> > </extension> Would appreciate any help. Jacek

5 10

XML CDR ERROR ...
by Shelby Ramsey 05 Feb '09

05 Feb '09

Hello, I'm having it on both Fedora and Ubuntu boxes: 2009-02-05 01:26:27 [ERR] mod_xml_cdr.c:290 mod_xml_cdr_load() Open of xml_cdr.conf failed 2009-02-05 01:26:27 [CRIT] switch_loadable_module.c:839 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_xml_cdr.so **Module load routine returned an error** Details: -- Ubuntu 6.04 LTS -- Fedora 8 Tried a couple of things: -- messing with libcurl -- ./configure --without-libcurl Thanks! SDR

3 3