March 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

CID Question: How pass the CID?
by Javier Aristizábal 08 Mar '10

08 Mar '10

Hi folks!! I have an incoming call being bridged to an outgoing call ann i'm trying to pass the phone number of the incoming caller as the caller id on the leg B. But it doesn't work. Here is my dialplan: <extension name="incoming"> <condition field="destination_number" expression="^(3055908934)$"> <action application="info"/> <action application="set" data="effective_callee_id_number=${caller_id_number}"/> <action application="bridge" data="{sip_cid_type=rpid}sofia/gateway/xcast/17862065658"/> </condition> </extension> Here is the Log and SIP trace: http://pastebin.freeswitch.org/12368 This is the INVITE on the bLeg: INVITE sip:17862065658@wholesale.xcastlabs.com<sip%3A17862065658(a)wholesale.xcastlabs.com>SIP/2.0 Via: SIP/2.0/UDP 66.231.242.173:5080;rport;branch=z9hG4bKS6jFBtp9crtFp Max-Forwards: 68 From: "Javier" <sip:xcast@wholesale.xcastlabs.com<sip%3Axcast(a)wholesale.xcastlabs.com> ;transport=udp>;tag=K6U13Hy2rpgBe To: <sip:17862065658@wholesale.xcastlabs.com<sip%3A17862065658(a)wholesale.xcastlabs.com> > Call-ID: 0c42bf79-a59c-122d-b98c-0014384f7ec2 CSeq: 127923447 INVITE Contact: <sip:gw+xcast@66.231.242.173:5080;transport=udp;gw=xcast> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-16921 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: 250 X-FS-Support: update_display Remote-Party-ID: "Javier" <sip:3052606062@wholesale.xcastlabs.com<sip%3A3052606062(a)wholesale.xcastlabs.com> >;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1268060050 1268060051 IN IP4 66.231.242.173 s=FreeSWITCH c=IN IP4 66.231.242.173 t=0 0 m=audio 23516 RTP/AVP 18 101 13 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 And i'm on FreeSWITCH Version 1.0.trunk (16921) Many thanks /Javier

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hunt group - play music when trying external numbers
by William Traenkle 08 Mar '10

08 Mar '10

I am sending this again as I did not see it go through the first time ... FS Community, I am new to freeSWITCH and this mailing list and I appreciate your support in advance. My Goal: To create a hunt group x7000 that sequentially dials an internal extension, i.e. x1000, and then a cell phone number, i.e. 6194548924, while playing music the entire time when each number is being tried. The Issue: This was working great a few months back, but now music plays when only when x1000 is tried but NOT when the cell phone is tried. Ideas? Please advise. My hunt group code is below: session.answer(); var domain_name = session.getVariable("domain_name"); var extension = '7000'; var result; var timeoutpin = 7500; var objdate = new Date(); var adjusthours = 0; //Adjust Server time that is set to GMT 7 hours var adjustoperator = "-"; //+ or - if (adjustoperator == "-") { var objdate2 = new Date(objdate.getFullYear(),objdate.getMonth(),objdate.getDate(),(objdate.getHours() - adjusthours),objdate.getMinutes(),objdate.getSeconds()); } if (adjustoperator == "+") { var objdate2 = new Date(objdate.getFullYear(),objdate.getMonth(),objdate.getDate(),(objdate.getHours() + adjusthours),objdate.getMinutes(),objdate.getSeconds()); } var Hours = objdate2.getHours(); var Mins = objdate2.getMinutes(); var Seconds = objdate2.getSeconds(); var Month = objdate2.getMonth() + 1; var Date = objdate2.getDate(); var Year = objdate2.getYear() var Day = objdate2.getDay()+1; var exit = false; function get_sofia_contact(extension,domain_name, profile){ if (profile == "auto") { profile = "internal"; session.execute("set", "sofia_contact_"+extension+"=${sofia_contact("+profile+"/"+extension+"@"+domain_name+")}"); sofia_contact = session.getVariable("sofia_contact_"+extension); if (sofia_contact == "error/user_not_registered") { profile = "external"; session.execute("set", "sofia_contact_"+extension+"=${sofia_contact("+profile+"/"+extension+"@"+domain_name+")}"); sofia_contact = session.getVariable("sofia_contact_"+extension); } } else { session.execute("set", "sofia_contact_"+extension+"=${sofia_contact("+profile+"/"+extension+"@"+domain_name+")}"); sofia_contact = session.getVariable("sofia_contact_"+extension); } console_log( "info", "sofia_contact "+profile+": "+sofia_contact+".\n" ); return sofia_contact; } function mycb( session, type, obj, arg ) { try { if ( type == "dtmf" ) { console_log( "info", "digit: "+obj.digit+"\n" ); if ( obj.digit == "#" ) { //console_log( "info", "detected pound sign.\n" ); exit = true; return( false ); } dtmf.digits += obj.digit; if ( dtmf.digits.length >= digitmaxlength ) { exit = true; return( false ); } } } catch (e) { console_log( "err", e+"\n" ); } return( true ); } //end function mycb dialed_extension = session.getVariable("dialed_extension"); domain_name = session.getVariable("domain_name"); domain = session.getVariable("domain"); us_ring = session.getVariable("us-ring"); caller_id_name = session.getVariable("caller_id_name"); caller_id_number = session.getVariable("caller_id_number"); effective_caller_id_name = session.getVariable("effective_caller_id_name"); effective_caller_id_number = session.getVariable("effective_caller_id_number"); outbound_caller_id_name = session.getVariable("outbound_caller_id_name"); outbound_caller_id_number = session.getVariable("outbound_caller_id_number"); session.execute("set", "ringback=${hold_music}"); //set to ringtone session.execute("set", "transfer_ringback=${hold_music}"); //set to ringtone session.execute("set", "call_timeout=30"); session.execute("export", "call_timeout=30"); session.execute("set", "continue_on_fail=true"); session.execute("set", "hangup_after_bridge=true"); //console_log( "info", "dialed extension:"+dialed_extension+".\n" ); //console_log( "info", "domain: "+domain+".\n" ); //console_log( "info", "us_ring: "+us_ring+".\n" ); //console_log( "info", "domain: "+domain+".\n" ); //console_log( "info", "domain_name: "+domain_name+".\n" ); //console_log( "info", "action call now don't wait for dtmf\n" ); if ( session.ready() ) { session.answer(); sofia_contact_1000 = get_sofia_contact("1000",domain_name, "auto"); session.execute("bridge", sofia_contact_1000); session.execute("bridge", "sofia/gateway/flowroute.com/16194548924"); //timeout if (session.getVariable("originate_disposition")!="SUCCESS" && session.getVariable("originate_disposition")!="ORIGINATOR_CANCEL"){ session.execute("info", ""); session.execute("transfer", "*991000"); } //clear variables dialed_extension = ""; new_extension = ""; domain_name = ""; domain = ""; } //end if session.ready Thanks. -Will

1 0

g711 to g722 transcoding causing distortion on Polycom IP6000 handsets
by Dan Lane 08 Mar '10

08 Mar '10

This is almost certainly more of a Polycom issue than a FreeSWITCH one but I'll mention it here on the offchance that someone else has experienced the following: Loud G.711a calls that are being transcoded to G.722 are massively distorted on Polycom IP6000 handsets and on all other handsets the call just sounds loud and isn't distorted. This is most common in conference calls where callers are using a mixture of codecs. If I disable G.722 the Polycom sounds fine even though the G.711a call is still too loud and if I lower the volume of the conference the distortion still happens. So, has anyone else experienced this and if so, did you find a solution? Regards, Dan Lane

2 4

XML curl and multiple extension in dialplan
by paul gore 08 Mar '10

08 Mar '10

Greetings, We use XML curl to fetch our dial plan extensions from a database. As per FS wiki we can only return one XML extension upon FS request. Now we need to return a set of extensions with same number condition and context but with different second condition so that FS can pick up the right one. Is it possible at all? And if yes how response XML with multiple extensions should look like? Thanks a lot.

5 14

Send a call directly to voicemail Transfer Help
by Aloysius Lloyd 08 Mar '10

08 Mar '10

Hi All, I would like to setup a dial plan, Dial *99[extension number] send a call directly to voicemail. I setup the following Dial Plan . But when I transfer the call *99[extension number] the call failed. <extension name="voicemail_transfer"> <condition field="destination_number" expression="^(\*9920[0-9])$"> <action application="answer"/> <action application="sleep" data="1000"/> <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/> </condition> </extension> But when I transfer the call *99[extension number] the call failed.My extensions range 201 - 209 Please let me know where the mistake is. Thanks Lloyd

6 6

server to server jingle
by Madovsky 08 Mar '10

08 Mar '10

Hi all, is anyone knows a good example to configure freeswitch and an xmpp server (openfire if possible) I read this article http://wiki.freeswitch.org/wiki/Dingaling#Getting_it_working but no example and this one http://www.alijawad.org/cms/index.php?option=com_content&task=view&id=21&It… but seems to doesn't work as expected. Thanks ! Franck

2 2

Conference Recording
by Josh Rivers 08 Mar '10

08 Mar '10

On last week's conference call, someone was recording, did that recording get put online somewhere? Josh

3 2

Virtualized FreeSWITCH
by Rob Forman 08 Mar '10

08 Mar '10

Hi all, Is anyone running FreeSWITCH virtualized (xen or vmware) in production? I saw ec2 and xen mentioned a few places. If so, are there tips or tricks for handle timer issues which affect virtual machines in general, such as recommended divider or clocksource settings? I'm seeing funny timing issues while testing with audio recording and playback that speed up or slow down. I have tried both paravirt and hvm. The default xen kernel has a timer of 250, so I tried to hvm instead using the stock 5.3 kernel with the 1khz timer but didn't see much improvement. All of this was tested on 64-bit CentOS 5.3. - Physical host: Linux localhost.localdomain 2.6.18-164.11.1.el5xen #1 SMP Wed Jan 20 08:06:04 EST 2010 x86_64 x86_64 x86_64 GNU/Linux CONFIG_HZ=250 - Paravirt host: Linux localhost.localdomain 2.6.18-128.el5xen #1 SMP Wed Jan 21 11:12:42 EST 2009 x86_64 x86_64 x86_64 GNU/Linux CONFIG_HZ=250 - Hvm host: Linux localhost.localdomain 2.6.18-128.el5 #1 SMP Wed Jan 21 10:41:14 EST 2009 x86_64 x86_64 x86_64 GNU/Linux CONFIG_HZ=1000 I'm sure physical hosts are the de-facto best practice, but are virtual machines not worth it at all? Cheers, Rob

12 17

FScomm
by Madovsky 08 Mar '10

08 Mar '10

Hi, is http://wiki.freeswitch.org/wiki/FSComm available yet ? because QT framework link has broken and some instruction don't work with last svn trunk (for example to compile FScomm in FS svn root) Thanks Franck

4 11

Need help on dingaling module
by ITS Solution 08 Mar '10

08 Mar '10

Hi all, I recently installed FS version 16983 with dingaling module enabled. I am using the client configuration. My system is Linux Ubuntu 8.04 hardy running on Xen VM. When I set up a call between a phone and a google talk, the audio from google talk is all noise. The audio from phone to google talk sounds good though. Before I did the upgrade, I was running on version 16892. In that release, I can hear audio on both paths. The audio on the path from google talk to FS results in about 3-5 seconds delay. Does anybody experience the similar issue or can help me to point out where could be the problem? Thanks. Chaofeng

1 0