Re: [Freeswitch-users] SIP Invite IP fragmentation issue
Ok, my bad. Ethereal for some reason was showing only the first fragment (ethereal bug?). But, now it seems I have hit another problem - it seems that the SIP invites (which are fragmented) are being dropped by the firewall in between us and the SIP provider. Is it possible to shrink the size of the SIP invite so that it fits in a single packet? Any optional stuff in the SIP invite that is sent, that can be thrown away? -Saurabh From: [EMAIL PROTECTED]: [EMAIL PROTECTED]: Tue, 18 Nov 2008 07:34:11 +Subject: [Freeswitch-users] SIP Invite IP fragmentation issue I am having an *odd* issue, which i am not sure freeswitch is to be blamed for. Sometimes, the SIP invites are bigger than 1500 bytes causing IP fragmentation, but when I look at the TCP dump (on the same machine as freeswitch), I see that only the first packet of the fragment is captured. Is freeswitch trying to do its own IP fragmentation or is it relying on underlying linux (kernel 2.6.18)? I created a small program to send UDP packets of 2000 bytes, and also tried with ping -s 2000, and both were successful, so am leaning towards blaming Freeswitch. Any suggestions? -Saurabh Stay up to date on your PC, the Web, and your mobile phone with Windows Live Click here _ Stay up to date on your PC, the Web, and your mobile phone with Windows Live http://clk.atdmt.com/MRT/go/119462413/direct/01/___ 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] SIP Invite IP fragmentation issue
2008/11/18 Saurabh Aggarwal [EMAIL PROTECTED]: Ok, my bad. Ethereal for some reason was showing only the first fragment (ethereal bug?). But, now it seems I have hit another problem - it seems that the SIP invites (which are fragmented) are being dropped by the firewall in between us and the SIP provider. Is it possible to shrink the size of the SIP invite so that it fits in a single packet? Any optional stuff in the SIP invite that is sent, that can be thrown away? Welcome to the reason for which IETF is moving to SIP TCP/SCTP ;) -- Iñaki Baz Castillo [EMAIL PROTECTED] ___ 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] SIP Invite IP fragmentation issue
enabling compact headers - what is that? -Saurabh Date: Tue, 18 Nov 2008 04:29:28 -0600From: [EMAIL PROTECTED]: [EMAIL PROTECTED]: Re: [Freeswitch-users] SIP Invite IP fragmentation issueIts not really possible other then enabling compact headers or by getting rid of codecs that you don’t actually want to use... Another thing you could do is get your broken ISP to fix their firewall... It is not correct to just drop fragmented packets just because they are fragmented.. This is something that will happen on a regular basis on the internet as not everything has an MTU of 1500 From: Saurabh Aggarwal [EMAIL PROTECTED]Reply-To: freeswitch-users@lists.freeswitch.orgDate: Tue, 18 Nov 2008 10:19:55 +To: freeswitch-users@lists.freeswitch.orgSubject: Re: [Freeswitch-users] SIP Invite IP fragmentation issueOk, my bad. Ethereal for some reason was showing only the first fragment (ethereal bug?). But, now it seems I have hit another problem - it seems that the SIP invites (which are fragmented) are being dropped by the firewall in between us and the SIP provider. Is it possible to shrink the size of the SIP invite so that it fits in a single packet? Any optional stuff in the SIP invite that is sent, that can be thrown away? -Saurabh From: [EMAIL PROTECTED]: [EMAIL PROTECTED]: Tue, 18 Nov 2008 07:34:11 +Subject: [Freeswitch-users] SIP Invite IP fragmentation issueI am having an *odd* issue, which i am not sure freeswitch is to be blamed for. Sometimes, the SIP invites are bigger than 1500 bytes causing IP fragmentation, but when I look at the TCP dump (on the same machine as freeswitch), I see that only the first packet of the fragment is captured. Is freeswitch trying to do its own IP fragmentation or is it relying on underlying linux (kernel 2.6.18)? I created a small program to send UDP packets of 2000 bytes, and also tried with ping -s 2000, and both were successful, so am leaning towards blaming Freeswitch. Any suggestions? -Saurabh Stay up to date on your PC, the Web, and your mobile phone with Windows Live Click here http://clk.atdmt.com/MRT/go/119462413/direct/01/ Stay up to date on your PC, the Web, and your mobile phone with Windows Live Click here http://clk.atdmt.com/MRT/go/119462413/direct/01/ ___Freeswitch-users mailing [EMAIL PROTECTED]://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org _ Get more done, have more fun, and stay more connected with Windows Mobile®. http://clk.atdmt.com/MRT/go/119642556/direct/01/___ 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] SIP Invite IP fragmentation issue
Thanks, how do I enable this in freeswitch? Can this be done through the SIP configuration file? -Saurabh Date: Tue, 18 Nov 2008 12:05:18 +0100From: [EMAIL PROTECTED]: [EMAIL PROTECTED]: Re: [Freeswitch-users] SIP Invite IP fragmentation issueThe rfc also describes why:SIP provides a mechanism to represent common header field names in an abbreviated form. This may be useful when messages would otherwise become too large to be carried on the transport available to it (exceeding the maximum transmission unit (MTU) when using UDP, for example). These compact forms are defined in Section 20. A compact form MAY be substituted for the longer form of a header field name at any time without changing the semantics of the message. A header field name MAY appear in both long and short forms within the same message. Implementations MUST accept both the long and short forms of each header name. On Tue, Nov 18, 2008 at 11:52 AM, Iñaki Baz Castillo [EMAIL PROTECTED] wrote: 2008/11/18 Saurabh Aggarwal [EMAIL PROTECTED]: enabling compact headers - what is that?SIP allows compact headers names for a few heades: From = f To = t Via = v ...--Iñaki Baz Castillo[EMAIL PROTECTED]___Freeswitch-users mailing [EMAIL PROTECTED]://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org _ Get more done, have more fun, and stay more connected with Windows Mobile®. http://clk.atdmt.com/MRT/go/119642556/direct/01/___ 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] Listen to a file, while recording?
hi, i would like to be able to listen to conversations, while they are ongoing. this should not happen over a phone. i would like to be able to have a link or something in my admin-area, where i can click, if i want to listen to a conversation. i thought about to start a record with socket inbound on a specific uuid and while the recording is done, i would like to play the file. but i have the feeling, that this is not possible, while the file is not finished. is there a way to stream the audio over the web? 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] DTMF
These aren't inserting 1003 as the caller_id_number are they?
/b
On Nov 18, 2008, at 5:19 AM, Baskar wrote:
action application=db data=insert/spymap/$
{caller_id_number}/${uuid}/
action application=db data=insert/last_dial/$
{caller_id_number}/${destination_number}/
action application=db data=insert/last_dial/global/$
{uuid}/
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[Freeswitch-users] Unstable att_xfer
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! ___ 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] Unstable att_xfer
What device are you using? /b On Nov 18, 2008, at 8:03 AM, x y wrote: Im using latest fs, and sip phones without built in transfer. ___ 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] Unstable att_xfer
Datavox Ip-300, and X-Lite softphone for testing. Hirdetés (x) Váltson most olcsóbb kötelezőre a biztosítás-hu-val. www.biztositas.hu - a kötelező biztosítások kiindulópontja! ___ 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] RFC 4028 - SIP Session Timers
Hi, I've read that FS supports/implements Session Timers to monitorice both legs of a call. How to enable it? I mean: alice --- FS bob - alice calls bob vía FS - FS calls bob. - bob answers (sends 200 OK). - bypass_media mode, no RTP through FS. - FS establishes a SIP dialog with alice and other one with bob. - From this moment FS starts sending periodically in-dialog INVITE/UPDATE to both legs in order to check if each SIP dialog is still alive in both endpoints. - In case alice crashes (looses dialog info), alice will reply 481 Call/Transaction doesn't exist when the in-dialog INVITE/UPDATE arrives from FS, so FS will understand that alice has ended the dialog (or has crashed) and sends a BYE to bob. Is it possible with FS? how to enable it? -- Iñaki Baz Castillo [EMAIL PROTECTED] ___ 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] Javascript: stream, speak, stream - cepstral cut offs 2nd stream
Hi,
Thanks, Anthony, for your reply.
mSession.sleep(100);
does indeed do the trick!
Cheers, Birgit
On 17/11/08 19:51, Anthony Minessale wrote:
you never want to msleep during a js running on a call
you should use session.sleep(500);
msleep blocks the whole thread and thus the audio.
On Mon, Nov 17, 2008 at 1:05 PM, Birgit Arkesteijn [EMAIL PROTECTED]wrote:
Hi,
In javascript I doing the following:
var consumer_name = Birgit Arkesteijn;
var endpoint_url = 'sofia/gateway/westhawk/0662';
var mSession = new Session({ignore_early_media=true,originate_timeout=8}
+ endpoint_url);
var ready = mSession.ready();
// log(merchant answered: + ready);
if (ready == true)
{
mSession.streamFile(westhawk/lead_waiting.wav, dummy);
mSession.speak(cepstral, Diane, break time='2s'/ +
consumer_name);
// msleep(500);
mSession.streamFile(westhawk/consumer_hungup.wav,
dummy);
exit();
}
function dummy(session, type, data, arg){}
No matter what I try, cepstral somehow cuts off the beginning two
seconds of the second streamFile().
Adding a msleep(500) in between 'speak' and 'streamFile' makes it only
worse.
I'm running
FreeSWITCH Version 1.0.trunk (597:10325M) Started.
and
Cepstral_Diane_x86-64-linux_5.1.0/
on Linux Suse 10.0 x86_64
Any ideas?
Thanks, Birgit
--
-- Birgit Arkesteijn, [EMAIL PROTECTED],
-- 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
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[Freeswitch-users] Javascript: record ringing of session
Hi,
In Javascript, I do the following:
var mSession = new
Session({ignore_early_media=true,originate_timeout=8} + endpoint_url);
mSession.execute(record_session, recordfile);
This works very well, however, the client would like the rings to be
recorded as well.
I've tried using various channel variables, but 'new Session' only
returns after the call is answered.
I tried calling
apiExecute(record, file_path);
before 'new Session', but that returns an error.
Is is possible to record the rings (in javascript)?
And if so, how would I do that?
Thanks, Birgit
--
-- Birgit Arkesteijn, [EMAIL PROTECTED],
-- 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
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Re: [Freeswitch-users] Listen to a file, while recording?
you can use the eavesdrop dialplan app from a new call to spy on an in progress session it takes the uuid of the channel you want to listen to as the arg. On Tue, Nov 18, 2008 at 6:28 AM, Dennis [EMAIL PROTECTED] wrote: hi, i would like to be able to listen to conversations, while they are ongoing. this should not happen over a phone. i would like to be able to have a link or something in my admin-area, where i can click, if i want to listen to a conversation. i thought about to start a record with socket inbound on a specific uuid and while the recording is done, i would like to play the file. but i have the feeling, that this is not possible, while the file is not finished. is there a way to stream the audio over the web? 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:[EMAIL PROTECTED] [EMAIL PROTECTED] GTALK/JABBER/PAYPAL:[EMAIL PROTECTED][EMAIL PROTECTED] IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[EMAIL PROTECTED] [EMAIL PROTECTED] iax:[EMAIL PROTECTED]/888 googletalk:[EMAIL PROTECTED][EMAIL PROTECTED] pstn:213-799-1400 ___ 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] Problems with DTMF and * on inbound leg
set the variable playback_terminators=none before you execute playback. On Tue, Nov 18, 2008 at 6:21 AM, Dennis [EMAIL PROTECTED] wrote: i am using socket outbound and if an inbound call comes in, i answer the call and play a soundfile for the caller. if the caller presses the dtmf key * the playback is stopped and i receive an channel_execute_complete playback for this file. this only happens with the * - the other keys do nothing (as i whish). i there somewhere a default setting for this, which i can turn off? i looked through all conf-files, but could not find enything. for my tests i use the default 1000 user as inbound. in the ivr.conf.xml i found something with menu-top and digits *. i commented this line out, but this didn't change anything. 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:[EMAIL PROTECTED] [EMAIL PROTECTED] GTALK/JABBER/PAYPAL:[EMAIL PROTECTED][EMAIL PROTECTED] IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[EMAIL PROTECTED] [EMAIL PROTECTED] iax:[EMAIL PROTECTED]/888 googletalk:[EMAIL PROTECTED][EMAIL PROTECTED] pstn:213-799-1400 ___ 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] Unstable att_xfer
Devices: IP-300 with 1007 registered IP-300 with 1009 registered X-Lite with 1011 registered Situation where att_xfer success: -Dial IP-300 with 6691007 from X-Lite (669 is just a prefix, assuring to execute my extension) -1007 ringing, get the phone -press *1 on 1007, then dial 1009 when the dialing wav is starting to play, while X-Lite plays hold music -1009 (the other IP-300) ringing, get the phone -1009 and 1007 bridged together, they can talk, then I hang up 1007 -1009 and 1011 bridged together, they can talk -few secs later, I hang up 1009 -1011 hangs up too Situation where att_xfer fails: -Dial IP-300 with 6691007 from X-Lite (669 is just a prefix, assuring to execute my extension) -1007 ringing, get the phone -press *1 on 1007, then dial 1009 when the dialing wav is starting to play, while X-Lite plays hold music -1009 (the other IP-300) ringing, get the phone -1009 and 1007 bridged together, they can talk, then I hang up 1007 different from here- -1009 silent, but not on hold, nor on hung up, 1011 stops playing waiting music, being silent too -few secs later, I hang up 1009 -1011 doesnt react, so I hang up 1011 too When the att_xfer went failure, I didnt make any noise. When it went ok, I spoke into all phones. I dont know if it counts. Logst added to pastebin. Cheers, Viktor Hirdetés (x) Váltson most olcsóbb kötelezőre a biztosítás-hu-val. www.biztositas.hu - a kötelező biztosítások kiindulópontja! ___ 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] SIP Invite IP fragmentation issue
the only reliable answer is use TCP. The RFC is daft in this matter. They say when it's bigger than mtu to automatically use TCP instead. And timeout for 10 seconds then fall back to UDP. Its mandatory in SIP to support both TCP and UDP up to 64k per packet. As you can see, since barely anything will do this right, your best bet is to only use TCP when you have this kind of traffic. On Tue, Nov 18, 2008 at 5:52 AM, Saurabh Aggarwal [EMAIL PROTECTED] wrote: Thanks, how do I enable this in freeswitch? Can this be done through the SIP configuration file? -Saurabh -- Date: Tue, 18 Nov 2008 12:05:18 +0100 From: [EMAIL PROTECTED] To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] SIP Invite IP fragmentation issue The rfc also describes why: SIP provides a mechanism to represent common header field names in an abbreviated form. This may be useful when messages would otherwise become too large to be carried on the transport available to it (exceeding the maximum transmission unit (MTU) when using UDP, for example). These compact forms are defined in Section 20. A compact form MAY be substituted for the longer form of a header field name at any time without changing the semantics of the message. A header field name MAY appear in both long and short forms within the same message. Implementations MUST accept both the long and short forms of each header name. On Tue, Nov 18, 2008 at 11:52 AM, Iñaki Baz Castillo [EMAIL PROTECTED]wrote: 2008/11/18 Saurabh Aggarwal [EMAIL PROTECTED]: enabling compact headers - what is that? SIP allows compact headers names for a few heades: From = f To = t Via = v ... -- Iñaki Baz Castillo [EMAIL PROTECTED] ___ 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 -- Get more done, have more fun, and stay more connected with Windows Mobile(R). See how. http://clk.atdmt.com/MRT/go/119642556/direct/01/ ___ 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:[EMAIL PROTECTED] [EMAIL PROTECTED] GTALK/JABBER/PAYPAL:[EMAIL PROTECTED][EMAIL PROTECTED] IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[EMAIL PROTECTED] [EMAIL PROTECTED] iax:[EMAIL PROTECTED]/888 googletalk:[EMAIL PROTECTED][EMAIL PROTECTED] pstn:213-799-1400 ___ 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] Unstable att_xfer
You missed one thing. the console log with debug. /b On Nov 18, 2008, at 9:57 AM, x y wrote: Devices: IP-300 with 1007 registered IP-300 with 1009 registered X-Lite with 1011 registered Situation where att_xfer success: -Dial IP-300 with 6691007 from X-Lite (669 is just a prefix, assuring to execute my extension) -1007 ringing, get the phone -press *1 on 1007, then dial 1009 when the dialing wav is starting to play, while X-Lite plays hold music -1009 (the other IP-300) ringing, get the phone -1009 and 1007 bridged together, they can talk, then I hang up 1007 -1009 and 1011 bridged together, they can talk -few secs later, I hang up 1009 -1011 hangs up too Situation where att_xfer fails: -Dial IP-300 with 6691007 from X-Lite (669 is just a prefix, assuring to execute my extension) -1007 ringing, get the phone -press *1 on 1007, then dial 1009 when the dialing wav is starting to play, while X-Lite plays hold music -1009 (the other IP-300) ringing, get the phone -1009 and 1007 bridged together, they can talk, then I hang up 1007 different from here- -1009 silent, but not on hold, nor on hung up, 1011 stops playing waiting music, being silent too -few secs later, I hang up 1009 -1011 doesnt react, so I hang up 1011 too When the att_xfer went failure, I didnt make any noise. When it went ok, I spoke into all phones. I dont know if it counts. Logst added to pastebin. Cheers, Viktor ___ 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] Unstable att_xfer
And btw, sorry for my english... thinked and tought, in one sentenceif only my english teacher would saw this :) Hirdetés (x) Váltson most olcsóbb kötelezőre a biztosítás-hu-val. www.biztositas.hu - a kötelező biztosítások kiindulópontja! ___ 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] Unstable att_xfer
I tought u thinked of these: http://pastebin.freeswitch.org/6179 http://pastebin.freeswitch.org/6178 If not, please correct me . Viktor Hirdetés (x) Váltson most olcsóbb kötelezőre a biztosítás-hu-val. www.biztositas.hu - a kötelező biztosítások kiindulópontja! ___ 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] Javascript: record ringing of session
Hi Michael,
Thanks for your reply.
I tried a variety of options:
var mSession = new Session(endpoint_url);
var mSession = new Session({ignore_early_media=false} + endpoint_url);
both only return after the session is answered (originated?).
The customer wants
the number of seconds we wait for each user to pick up
configurable, hence I use originate_timeout. I noticed (by try and
error) that this is the timeout that does the trick. The other timeout
variables behaved differently.
I'm currently trying to see how I can do recording of the rings. If that
clashes with this neat originate_timeout variable, I'll have to see
what my options are.
I did read the http://wiki.freeswitch.org/wiki/Channel_Variables page,
but to be honest, even with descriptions I don't really understand the
implications of using these variables, sorry.
Thanks, Birgit
On 18/11/08 17:03, Michael S Collins wrote:
Birgit,
I'm pretty sure that ringing is early media and you've got
ignore_early_media set to true. If you can work without ignoring early
media then you'll get rings. However, there's probably a reason you're
ignoring early media so be sure to test thoroughly without ignoring
early media just to make sure something else doesn't break. Actually I
just noticed the call timeout you have. IIRC that does require you to
ignore early media. Is there a way you can work without that call
timeout? Let me know. Also, I might have an alternate solution but I
can't give you more info until I get to my desk in an hour or so.
-MC
Sent from my iPhone
On Nov 18, 2008, at 7:40 AM, Birgit Arkesteijn [EMAIL PROTECTED]
wrote:
Hi,
In Javascript, I do the following:
var mSession = new
Session({ignore_early_media=true,originate_timeout=8} +
endpoint_url);
mSession.execute(record_session, recordfile);
This works very well, however, the client would like the rings to be
recorded as well.
I've tried using various channel variables, but 'new Session' only
returns after the call is answered.
I tried calling
apiExecute(record, file_path);
before 'new Session', but that returns an error.
Is is possible to record the rings (in javascript)?
And if so, how would I do that?
Thanks, Birgit
--
-- Birgit Arkesteijn, [EMAIL PROTECTED],
-- 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
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Re: [Freeswitch-users] Javascript: record ringing of session
Birgit,
I'm almost to my office. I will give you more info soon. I have not
used js in this capacity so we will have to do some experimenting.
-MC
Sent from my iPhone
On Nov 18, 2008, at 9:51 AM, Birgit Arkesteijn [EMAIL PROTECTED]
wrote:
Hi Michael,
Thanks for your reply.
I tried a variety of options:
var mSession = new Session(endpoint_url);
var mSession = new Session({ignore_early_media=false} +
endpoint_url);
both only return after the session is answered (originated?).
The customer wants
the number of seconds we wait for each user to pick up
configurable, hence I use originate_timeout. I noticed (by try and
error) that this is the timeout that does the trick. The other timeout
variables behaved differently.
I'm currently trying to see how I can do recording of the rings. If
that
clashes with this neat originate_timeout variable, I'll have to see
what my options are.
I did read the http://wiki.freeswitch.org/wiki/Channel_Variables page,
but to be honest, even with descriptions I don't really understand the
implications of using these variables, sorry.
Thanks, Birgit
On 18/11/08 17:03, Michael S Collins wrote:
Birgit,
I'm pretty sure that ringing is early media and you've got
ignore_early_media set to true. If you can work without ignoring
early
media then you'll get rings. However, there's probably a reason
you're
ignoring early media so be sure to test thoroughly without ignoring
early media just to make sure something else doesn't break.
Actually I
just noticed the call timeout you have. IIRC that does require you to
ignore early media. Is there a way you can work without that call
timeout? Let me know. Also, I might have an alternate solution but I
can't give you more info until I get to my desk in an hour or so.
-MC
Sent from my iPhone
On Nov 18, 2008, at 7:40 AM, Birgit Arkesteijn
[EMAIL PROTECTED]
wrote:
Hi,
In Javascript, I do the following:
var mSession = new
Session({ignore_early_media=true,originate_timeout=8} +
endpoint_url);
mSession.execute(record_session, recordfile);
This works very well, however, the client would like the rings to be
recorded as well.
I've tried using various channel variables, but 'new Session' only
returns after the call is answered.
I tried calling
apiExecute(record, file_path);
before 'new Session', but that returns an error.
Is is possible to record the rings (in javascript)?
And if so, how would I do that?
Thanks, Birgit
--
-- Birgit Arkesteijn, [EMAIL PROTECTED],
-- 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
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[Freeswitch-users] accountcode and user id not present in CDRs
I am using acls (cidr) to accept incoming calls from a gateway that I do not want to register in my FS box. I have this gateway configured in a xml file : freeswitch/conf/directory/default/gateway1.xml include user id=GATEWAY1 mailbox= cidr=xxx.xxx.xxx.xxx/32 params param name=password value=1234/ /params variables variable name=accountcode value=CUSTOMER1/ variable name=user_context value=my_context/ variable name=effective_caller_id_name value=gateway1_callid/ variable name=effective_caller_id_number value=238383838383/ /variables /user /include I have the corresponding cidr in my ACL in acl.conf.xml. I am able to make a call from that gateway to my FS but in my CDRs (both xml or cdr_csv) the accountcode or user id is not present. Any help on how to define an endpoint (originating) and use some attribute (like account_code or user id) for billing purposes? ___ 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] accountcode and user id not present in CDRs
Add ${accuntcode} to the CDR template in cdr.conf.xml... the template
can include any variables from the session.
/b
On Nov 18, 2008, at 12:50 PM, [EMAIL PROTECTED] wrote:
Any help on how to define an endpoint (originating) and use some
attribute (like account_code or user id)
for billing purposes?
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Re: [Freeswitch-users] accountcode and user id not present in CDRs
The ${accountcode} variable IS set in the cdr_csv.xml conf file yet the
field is empty after the call.
Shouldn't it also show in the xml cdr? I thought the XML CDRs included
all of the session variables.
Brian West wrote:
Add ${accuntcode} to the CDR template in cdr.conf.xml... the template
can include any variables from the session.
/b
On Nov 18, 2008, at 12:50 PM, [EMAIL PROTECTED] wrote:
Any help on how to define an endpoint (originating) and use some
attribute (like account_code or user id)
for billing purposes?
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Re: [Freeswitch-users] accountcode and user id not present in CDRs
I'm going to guess that this is an inbound call to the user. Which
means the variables aren't set inbound to the user.
/b
On Nov 18, 2008, at 1:40 PM, [EMAIL PROTECTED] wrote:
The ${accountcode} variable IS set in the cdr_csv.xml conf file yet
the field is empty after the call.
Shouldn't it also show in the xml cdr? I thought the XML CDRs
included all of the session variables.
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Re: [Freeswitch-users] Relative timeout in Session.collectInput?
Yup, I tried different settings for dft_min and dft_confirm before but did not understand what they meant. Back then I guessed that dft meant discrete Fourier transform not default and these variables were cut off frequencies of a filter but that did not quite make sense. For instance, in the yes/no recognition part of the demo, one variable was set lower than the other but not in other places. Thanks for clearing up their meanings. I've played with values for those variables more based on the output and it does help. However, I'm and was getting recognition of strange patterns that all start with in superscript like: ? h p?o?? h\| ? What's going on here and what can I do about it? Thanks. Mark. -Original Message- From: Anthony Minessale [EMAIL PROTECTED] To: freeswitch-users@lists.freeswitch.org Sent: Mon, 17 Nov 2008 2:39 pm Subject: Re: [Freeswitch-users] Relative timeout in Session.collectInput? maybe its the detection of talk stop. are you pasing between your toppings. try saying them all very fast with no pause. It may be the pause between utterances that is catching you. the config has params. threshold (default 400): higher the number louder you have to talk to be considered talking silence-hits (default 35): number of hits below threshold before detecting stop talking On Mon, Nov 17, 2008 at 4:23 PM, [EMAIL PROTECTED] wrote: I'm suspecting that one of my pauses (or something else), as I'm saying my topping choices, is being interpreted as the end of my choices. I've tried adjusting the timeout argument value of collectInput but that doesn't seem to do anything. So, I'm left wondering what's causing my choices to be truncated and thus collectInput handles only a part of the toppings ordered. It doesn't seem like the speech recognizer is only recognizing one topping at a time as an event that's caught by collectInput since I have gotten recognition of more than one topping but usually not that many more. Nickolay Shmyrev at CMU Sphinx Help Forum thought there maybe some timeout involved because passing silence to the recognizer wasn't a good idea and suggested a relative timeout. So what's being passed to the recognizer or is there something else going on? Mark. -Original Message-From: Anthony Minessale [EMAIL PROTECTED] To: freeswitch-users@lists.freeswitch.org Sent: Sun, 16 Nov 2008 8:40 am Subject: Re: [Freeswitch-users] Relative timeout in Session.collectInput? The speech detector never stops working.? The collect input just pauses to wait for some input. It can catch the event any time a file is playing or that collect input is called. On Sat, Nov 15, 2008 at 1:19 PM, [EMAIL PROTECTED] wrote: Session.collectInput looks like it has a fixed timeout setting which blocks the channel until the timeout expires. I've been informed that appears to be the case from a brief examination of how speech is collected. SWITCH_DECLARE(switch_status_t) switch_ivr_collect_digits_callback? seems to use a fixed timeout to cancel recording. ?? Is it possible to set-up a relative timeout which is based on the time from when the last valid sound was recognized as in choosing toppings from a list of 20 toppings in the pizza demo? This situation involves an indeterminate number of valid sounds (toppings) and thus is unlike the dtmf example give in? http://wiki.freeswitch.org/wiki/Session_collectInput where a counter in the callback is also used to determine when to unblock the channel. Mark. ? Instant access to the latest most popular FREE games while you browse with the Games Toolbar - Download Now! ___ 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:[EMAIL PROTECTED] GTALK/JABBER/PAYPAL:[EMAIL PROTECTED] IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[EMAIL PROTECTED] iax:[EMAIL PROTECTED]/888 googletalk:[EMAIL PROTECTED] pstn:213-799-1400 ___ 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 Instant access to the latest most popular FREE games while you browse with the Games Toolbar - Download Now! ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org
Re: [Freeswitch-users] accountcode and user id not present in CDRs
you have to manually set the var on the channel in your dialplan.
On Tue, Nov 18, 2008 at 1:40 PM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
The ${accountcode} variable IS set in the cdr_csv.xml conf file yet the
field is empty after the call.
Shouldn't it also show in the xml cdr? I thought the XML CDRs included all
of the session variables.
Brian West wrote:
Add ${accuntcode} to the CDR template in cdr.conf.xml... the template
can include any variables from the session.
/b
On Nov 18, 2008, at 12:50 PM, [EMAIL PROTECTED] wrote:
Any help on how to define an endpoint (originating) and use some
attribute (like account_code or user id)
for billing purposes?
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FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
MSN:[EMAIL PROTECTED] [EMAIL PROTECTED]
GTALK/JABBER/PAYPAL:[EMAIL PROTECTED][EMAIL PROTECTED]
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:[EMAIL PROTECTED] [EMAIL PROTECTED]
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Re: [Freeswitch-users] RFC 4028 - SIP Session Timers
El Martes, 18 de Noviembre de 2008, Iñaki Baz Castillo escribió: Hi, I've read that FS supports/implements Session Timers to monitorice both legs of a call. How to enable it? I mean: alice --- FS bob - alice calls bob vía FS - FS calls bob. - bob answers (sends 200 OK). - bypass_media mode, no RTP through FS. - FS establishes a SIP dialog with alice and other one with bob. - From this moment FS starts sending periodically in-dialog INVITE/UPDATE to both legs in order to check if each SIP dialog is still alive in both endpoints. - In case alice crashes (looses dialog info), alice will reply 481 Call/Transaction doesn't exist when the in-dialog INVITE/UPDATE arrives from FS, so FS will understand that alice has ended the dialog (or has crashed) and sends a BYE to bob. Is it possible with FS? how to enable it? I've found those options in Sofia profiles: param name=enable-timer value=false/ param name=minimum-session-expires value=120/ They seem to be related to SIP Session Timers (nothing related to RTP), am I right? Thanks. -- Iñaki Baz Castillo ___ 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] RFC 4028 - SIP Session Timers
yes but RTP timers are in there too. /b On Nov 18, 2008, at 5:26 PM, Iñaki Baz Castillo wrote: They seem to be related to SIP Session Timers (nothing related to RTP), am I right? ___ 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] RFC 4028 - SIP Session Timers
El Miércoles, 19 de Noviembre de 2008, Brian West escribió: yes but RTP timers are in there too. Well, but I expect that RTP timers parameters are the following: param name=use-rtp-timer value=true/ param name=rtp-timer-name value=soft/ param name=rtp-timeout-sec value=300/ param name=rtp-hold-timeout-sec value=1800/ While SIP Session Timers parameters are those: param name=enable-timer value=false/ param name=minimum-session-expires value=120/ Am I right? Thanks a lot. -- Iñaki Baz Castillo ___ 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] Anthony is all heart ...
Hi All, We all like to be thanked and when someone does something for me, I like to show my gratitude towards them and I think what Anthony did for me today deserves a public show of gratitude. It goes like this: I have a single port FXO card in my home machine running FS and I needed to modify the tone_detect app so that it would not incorrectly detect busy tones during phone calls. If I call in and hang up on the remote end, FS would detect the busy tone and hang up correctly. The problem was that tone_detect would act as soon as it detected the requested tone (in my case, busy tone) so I might be in the middle of a phone call it would sometimes incorrectly detect a single busy tone and drop the call. I tried to ignore this at first but it started happening more often so I spoke to Anthony about the possibility of modifying tone_detect to have a hit count during a time window, so that it would have to detect X number of busy tones before hanging up the call. As you can all see, this is probably a minor issue on Anthony's priority list since there are many more important issues to fix or things to do on FS than fix tone_detect. Even so, he spent a good 6 hours on this issue and kept hacking at it until he got it fixed. Anthony, once again, fixing tone_detect really helped me but what I appreciate more than anything is the willingness to help and invest so much of your time on this .. so thank you once again! - Gonzalo (aka znoG) PS: I will soon update the wiki page http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect with details on how to use the hit count ___ 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] i need help regarding freeswitch
Dear all, Hi I ve just started reading on freeswitch. From where should i start? Basically, i am am a graduate electronic engineer. Can i be a successful developer of freeswitch? Its my first experience of any telephonic system. Faisal rehman, Pakistan ___ 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] i need help regarding freeswitch
On Wed, Nov 19, 2008 at 11:07 AM, Faisal Maqsoodi [EMAIL PROTECTED]wrote: Faisal: Welcome to the wonderful word of open source telephony. I ve just started reading on freeswitch. From where should i start? http://wiki.freeswitch.org/wiki/Main_Page http://wiki.freeswitch.org/wiki/Documentation are two good places to start ... Basically, i am am a graduate electronic engineer. Can i be a successful developer of freeswitch? Absolutely. Its my first experience of any telephonic system. Be prepared to read a lot, and then ask questions. When you get an answer, do document it for others on the wiki. -- wasim h. baig | principal consultant | convergence pk | +92 300 8508070 | as you scope creep, so shall we reap ... ___ 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] i need help regarding freeswitch
Thanks very much sir u r so nice n cooperative. --- On Tue, 11/18/08, Wasim Baig [EMAIL PROTECTED] wrote: From: Wasim Baig [EMAIL PROTECTED] Subject: Re: [Freeswitch-users] i need help regarding freeswitch To: freeswitch-users@lists.freeswitch.org Date: Tuesday, November 18, 2008, 10:36 PM On Wed, Nov 19, 2008 at 11:07 AM, Faisal Maqsoodi [EMAIL PROTECTED] wrote: Faisal: Welcome to the wonderful word of open source telephony. I ve just started reading on freeswitch. From where should i start? http://wiki.freeswitch.org/wiki/Main_Page http://wiki.freeswitch.org/wiki/Documentation are two good places to start ... Basically, i am am a graduate electronic engineer. Can i be a successful developer of freeswitch? Absolutely. Its my first experience of any telephonic system. Be prepared to read a lot, and then ask questions. When you get an answer, do document it for others on the wiki. -- wasim h. baig | principal consultant | convergence pk | +92 300 8508070 | as you scope creep, so shall we reap ... ___ 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
