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Question about P-Asserted-Identity
by David Ponzone 27 May '11

27 May '11

All, I am using a GIT version of beginning of May. I receive a P-Asserted-Identity from a provider, but I can't find the corresponding variable in the XML CDR. Is there a known bug that was fixed ? While I send this email, I am upgrading, so I'll let you know if that was it. Thanks David Ponzone Direction Technique email: david.ponzone(a)ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.

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Privacy: id / Inbound
by Liam Farr 27 May '11

27 May '11

Hi, When I receive an inbound call with Privacy: id set in the SIP invite freeswitch isn’t stripping off the caller I’d when passing the call onto an internal extension? The inbound invite looks like this; recv 1091 bytes from udp/[23.51.15.101]:5060 at 04:14:28.246255: ------------------------------------------------------------------------ INVITE sip:92568888@202.55.98.6 SIP/2.0 Max-Forwards: 69 Supported: 100rel To: <sip:92568888@202.55.98.6> From: <sip:225553389@23.51.15.101>;tag=3515458594-499967 P-Asserted-Identity: <sip:225553389@23.51.15.101> Privacy: id Call-ID: 23213457-3515458594-499959(a)MSX-101.squiggly.net CSeq: 1 INVITE Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK Via: SIP/2.0/UDP 23.51.15.101:5060 ;branch=z9hG4bKf91a7732fbc823dc2c3252d3f7ec83fc Contact: <sip:225553389@23.51.15.101:5060;tgrp=2Degrees_sst03> Expires: 180 Call-Info: <sip:23.51.15.101>;method="NOTIFY;Event=telephone-event;Duration=1000" Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 369 v=0 o=MSX-101 8787 8322 IN IP4 23.51.15.101 s=sip call c=IN IP4 23.51.15.103 t=0 0 m=audio 29708 RTP/AVP 8 0 18 4 100 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:4 G723/8000 a=fmtp:4 bitrate=6.3;annexa=yes a=rtpmap:100 X-NSE/8000 a=fmtp:100 192-194 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 With the internal invite looking like this; send 1083 bytes to udp/[202.55.98.122]:5080 at 04:14:28.260497: ------------------------------------------------------------------------ INVITE sip:555@202.55.98.122:5080 SIP/2.0 Via: SIP/2.0/UDP 202.55.98.7;rport;branch=z9hG4bK20Bv5jcFpj00c Max-Forwards: 67 From: "225553389" <sip:225553389@202.55.98.7>;tag=8eK40BpUtXSUB To: <sip:555@202.55.98.122:5080> Call-ID: a7bce49f-02ba-122f-a19a-8903425aa839 CSeq: 12898770 INVITE Contact: <sip:mod_sofia@202.55.98.7:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.7-svn-exported Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Privacy: id Content-Type: application/sdp Content-Disposition: session Content-Length: 203 X-FS-Support: update_display P-Asserted-Identity: "225553389" <sip:225553389@202.55.98.7> v=0 o=FreeSWITCH 1306450788 1306450789 IN IP4 202.55.98.7 s=FreeSWITCH c=IN IP4 202.55.98.7 t=0 0 m=audio 18880 RTP/AVP 8 9 0 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 As you can see the caller id is passed onto the internal extension, where I really need to strip it off / replace it with restricted / anonymous. Is there a way to catch this as a variable in the inbound dial plan? Thanks Liam

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XCAP Server
by Jakson Kalsson 27 May '11

27 May '11

Dear all, does free switch support XCAP? Thanks -- havesoftware, Inc. http://www.havesoftware.com Jakson Kalsson Senior Programmer jakkalsoon(a)havesoftware.com

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trying to get mod_xml_cdr working
by deniro 27 May '11

27 May '11

Hi All I am trying to get mod_xml_cdr working My webserver is like www.xxx.com at port 80 the test web site is working fine with perl http://www.xxx.com/PERL/test3.pl I have mod_xml_cdr and mod_xml_curl modules loaded. several questions; 1-- do I really need to load mod_xml_curl for mod_xml_cdr? trying to figure out if my problem is with cdr or with curl 2-- when dialing into conference, getting following from fs_cli [ERR] mod_xml_cdr.c:365 Got error [502] posting to web server [ http://www.xxx.com/PERL/cdr.pl] 3-- I am not quiete sure if this line is correct in xml_cdr.conf.xml <param name="url" value="http://www.xxx.com/PERL/cdr.pl"/> 4-- if I need mod_xml_curl (question 1) what should this line be like in xml_curl.conf.xml? <param name="gateway-url" value="http://www.xxx.com:80 <http://www.xxx.com/>" bindings="dialplan"/> again my webserver www.xxx.com at port 80 Thx deiro

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PRI Signaling Converters
by Spencer Thomason 27 May '11

27 May '11

Hello all, Does anyone have any experience with any PRI -> SIP signaling converters? Basically what I'm trying to do is migrate a Dialogic PBX that has a PRI card to a SIP trunk and eventually to a Freeswitch based PBX. And since this will be a transitional piece of equipment cost does become a factor. Thanks, Spencer

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CNAM Providers
by Spencer Thomason 26 May '11

26 May '11

Hello all, We are considering adding US CNAM lookups on our inbound DIDs. Does anyone have any experience with any of the providers listed on the wiki or any others for that matter. (i.e. accuracy, uptime, query response time, etc..)? Thanks, Spencer

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Duplicate Pastebin Post
by Spencer Thomason 26 May '11

26 May '11

Hello, I accidentally made a duplicate pastebin post.. :-/ Could someone with admin access please remove http://pastebin.freeswitch.org/16379 Thanks, Spencer

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using $${var} and ${var} in javascript
by Yungwei Chen 26 May '11

26 May '11

Hi, I am wondering if there's a way to use $${base_dir}, ${strftime(%Y-%m-%d-%H-%M-%S), and ${destination_number} in a javascript program. For example, new_session.execute("record_session", "$${base_dir}/recordings/archive/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"); Thanks.

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Freeswitch DBH - Possible to create connection pool to other DBs?
by Alan Frisch 26 May '11

26 May '11

Been playing with the DBH functionality in LUA. The connection pooling works great if you need to access the core DSN... but if you need to access an external DB, it seems that FS will open a single connection for the maximum number of concurrent calls accessing the LUA script. In * one can limit the connection pool to any DB in res_odbc.conf... but is there a way of doing this in FS? After a learning curve and lots of time, tthis is the one hangup in FS that is preventing me from trashing * altogether. Help me put * in the dustbin for good!

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ESL event filter work scheme
by Anton VG 26 May '11

26 May '11

Trying to find out if the given is intended behavior or not: I am noticed that I do not receive the BACKGROUND_JOB event if filter for the given event is added after filter on Unique-ID filter. e.g. if I do filter Unique-ID XXXXX filter Unique-ID YYYY filter Event-Name BACKGROUND_JOB I do not receive the BACKGROUND_JOB event at all, only events consisting filtered UUIDs but if I do: filter Event-Name BACKGROUND_JOB filter Unique-ID XXXXX filter Unique-ID YYYY I receive it, and other events with my UUID too. Seems a little odd, since described that it's a filter in Is the given intended behavior or a bug?

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