February 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

conference and xmpp
by Frank Carmickle 01 Feb '10

01 Feb '10

Hello I am trying to interact with conferences over xmpp. I want to be able to see status changes and mute/unmute etc... I can only list the members if I ask for a list. I am having trouble understanding the wiki entree on this topic. Profile names are confusing because all the examples show default as a name so it is hard to see what each profile name corresponds to. My config looks like this currently. <profile name="ourchatperms">  <user name="shawn(a)djernes.net" commands="all"/>  <user name="frankiec(a)djernes.net" commands="|deaf|dial|energy|kick|list|lock|mute|norecord|play|record|relate|say|saymember|stop|transfer|undeaf|unlock|unmute|volume_in|volume_out|"/> </profile> </chat-permissions> <profile name="cdquality"> <param name="caller-controls" value="custom-ewr"/> <param name="chat-permissions" value="ourchatperms"/> <param name="domain" value="$${domain}"/> Also we can only talk to the conference by conf+confnum-fqdn@fqdn. Is there a way to make it conf+confnum@fqdn? Any help would be greatly appreciated. Thank you. --FC

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Wrong RTP port submitted?
by Peter P GMX 01 Feb '10

01 Feb '10

I have defined the rtp port range for 12000-12100 in switch.conf.xml. However Freeswitch is offering a port 48320 in the invite message. The result is, that the incoming RTP stream is blocked by the firewall (I can see a reject for UDP 48320). Any hint how to solve this? Best regards Peter See config and invite message:  <param name="rtp-start-port" value="12000"/>--> <param name="rtp-end-port" value="12100"/>--> Invite: INVITE sip:027xxxxxxxx@sip.itsp.de SIP/2.0. Via: SIP/2.0/UDP 217.24.xx.xxx:5080;rport;branch=z9hG4bKjD923NvctXaFm. Max-Forwards: 69. From: "0608xxxxxxx" <sip:4918xxxxxxxxxxxx@sip.itsp.de;transport=udp>;tag=0Kp4tvU44UmXp. To: <sip:027xxxxxxxx@sip.itsp.de>. Call-ID: 30c86b94-85ca-122d-f88e-080027e51f59. CSeq: 126174137 INVITE. Contact: <sip:gw+tgnet3@217.24.xx.xxx:5080;transport=udp;gw=tgnet3>. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16032. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY. Supported: timer, precondition, path, replaces. Allow-Events: talk, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 320. P-Key-Flags: keys="3". X-FS-Support: update_display. Remote-Party-ID: "0608xxxxxxx" <sip:0608xxxxxxx@sip.itsp.de>;party=calling;screen=yes;privacy=off. . v=0. o=FreeSWITCH 1264536651 1264536652 IN IP4 217.24.xx.xxx. s=FreeSWITCH. c=IN IP4 217.24.xx.xxx. t=0 0. m=audio 48320 RTP/AVP 8 0 98 3 101 13. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:98 SPEEX/8000. a=rtpmap:3 GSM/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=rtpmap:13 CN/8000. a=ptime:20.

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latency growing with conference
by Frank Carmickle 01 Feb '10

01 Feb '10

Hello I am noticing that when short phrases are spoken with cepstral in a conference it pauses all the audio before speaking it. It seems as though the pauses are of different lengths and over time this adds up to a great amount of latency for the conference members to each other. Rejoining the conference fixes this. I already have <param name="member-flags" value="waste"/> I also notice that one member, who has lots of dropped packets, becomes very latent also. I have <action application="set" data="rtp_autoflush=true"/> Any thing else I can do about the two similar yet different issues? I am on 16431 at the moment. Thanks --FC

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Equivalent to Asterisk's "directrtpsetup=yes"?
by Fred-145 01 Feb '10

01 Feb '10

Hello Thanks to "directrtpsetup=yes", it appears that Asterisk can act as an SIP proxy and a PBX at the same time, ie. RTP packets will flow directly between the two SIP end-points with Asterisk still being able to provide PBX services like call transfer, MoH, etc. The point is to lower latency for packets to reach their final destination and lower CPU load on the Freeswitch server. Does Freeswitch offer the same feature, or must RTP packets always go through the Freeswitch servers? Thank you. -- View this message in context: http://old.nabble.com/Equivalent-to-Asterisk%27s-%22directrtpsetup%3Dyes%22… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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nixevent behavior
by lakshmanan ganapathy 01 Feb '10

01 Feb '10

Dear all I've done the following sample script to experiment the nixevent. I found some difference in behavior because of nixevent. Let me explain my question down the script. require ESL; use IO::Socket::INET; my $ip = "192.168.1.222"; my $sock = new IO::Socket::INET ( LocalHost => $ip, LocalPort => '8447', Proto => 'tcp', Listen => 1, Reuse => 1 ); die "Could not create socket: $!\n" unless $sock; for(;;) { my $new_sock = $sock->accept(); next if (not defined ($new_sock)); my $pid = fork(); if ($pid) { close($new_sock); next; } print "CHILD PID: $$\n"; my $host = $new_sock->sockhost(); my $fd = fileno($new_sock); my $con = new ESL::ESLconnection($fd); my $info = $con->getInfo(); my $uuid = $info->getHeader("unique-id"); printf "Connected call %s, from %s\n", $uuid, $info->getHeader("caller-caller-id-number"); my $r=$con->execute("answer"); $con->events("plain","all"); ########################## $con->send("nixevent DTMF"); my $val=$con->api("create_uuid"); $val = $val->getBody(); # LINE 1 chomp($val); print "UUID is $val\n"; my $e = $con->recvEvent(); $val = $e->getBody(); # LINE 2 chomp($val); print "UUID is $val\n"; close($new_sock); } # If the line ($con->send("nixevent DTMF");) is commented, then the result of create_uuid is obtained in LINE 1. # else, the result isn't obtained in the LINE 1 and it has nothing. The result is obtained only when I do a recvEvent, # followed by a getBody (LINE 2) Just want to know why the behavior differs when nixevent is present???

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determining the source of receyved call in public context
by Meftah Tayeb 01 Feb '10

01 Feb '10

hi, how do i determine the gateway or the Ip of a receyved call from ITSP's? i am calling my did from my mobile, but i see is processing the the mobile number no the ITSP User or did thanks

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How to populate Voice recording metadata with custom CRM data?
by Roly Maz 01 Feb '10

01 Feb '10

Hi All, I am using Freeswitch 1.0.4 (Latest Windows Installer version) on Windows 7 Ultimate for testing Assuming an outbound call, what would be the most sensible approach to pass custom CRM data into the voice-recording? I would like the voice recording of the call to include the customers social security number in the file title, or even metadata. In other words, is there a way to pass custom info, at time of call, to the dialplan for use in the creating the voice recording. Any pointers would be much appreciated Rgds Roly

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Re: [Freeswitch-users] Freeswitch core dump after upgrade to latest version
by Yehavi Bourvine 01 Feb '10

01 Feb '10

It works ok now (fixed on r16534). Thanks! __Yehavi: 2010/1/25 Yehavi Bourvine <yehavi.bourvine(a)gmail.com> > OK. The sources are under /home/freeswitch. Note that it has Hebrew modules > which we are now developing, but the problem is seen with the vanilla > version as well. > > The execs and all other stuff is at /freeswitch. > > regards, __Yehavi: > > 2010/1/25 Michael Jerris <mike(a)jerris.com> > > I have not had a chance to actually try it yet. I will let you know. >> >> Mike >> >> On Jan 25, 2010, at 2:43 PM, Yehavi Bourvine wrote: >> >> > Hello Mike, >> > >> > I see you are logged-in into our machine. Before I go to sleep: Do you >> need any help? >> > >> > Thanks, __yehavi: >> > >> >>

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Question about javascript
by Tomoya Matsubara 01 Feb '10

01 Feb '10

Hello, When the following scripts were tested, it seems to do the memory leak. Please teach when there is a problem in this script. -- test script -- session.execute("ring_ready"); session.answer(); session.setVariable("ringback", "%(1000, 2000, 440, 460)"); var bleg = new Session(); var sound_wav = "sounds/test01.wav"; var sound_leg = "both"; var op = "api_on_answer=uuid_broadcast "+session.uuid+" "+sound_wav+" "+sound_leg; var ret = bleg.originate(session, "{"+op+"}" + "sofia/gateway/profile0_gateway1/1000"); if(!session.ready()){ return; } if(!ret){ // bleg not answered. var sound_wav = "sounds/test02.wav"; session.streamFile(sound_wav); if(session.ready()){ session.hangup(); } return; } if(bleg.ready()){ bridge(session, bleg); }

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Replace Internal IP with External IP in From Header
by Code Ghar 01 Feb '10

01 Feb '10

I followed the example in Freeswitch behind NAT ( http://wiki.freeswitch.org/wiki/NAT_Traversal#Freeswitch_behind_NAT) In the Contact header of invite sent to an external gateway, I see sip:extension@ExternalIP:port but in the From header I see sip:extension@InternalIP. How can I change the From header of SIP message so that it displays the external IP instead of internal IP? The reason for doing this is that the external gateway authenticates and authorizes call based on the IP in From header. They expect an external IP and not an internal IP.

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