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dialing codes
by Jonas Gauffin 11 Nov '08

11 Nov '08

Hello Do anyone know where I can find a xml/csv file with international/national dialing codes? Thanks, Jonas

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Question about FIFO event
by Woody Dickson 11 Nov '08

11 Nov '08

Hi, I am trying to write an event listener that can record the time when a FIFO consumer rejoins the queue after the caller hangs up. Tracing through all the event traffic, I notice that there is FIFO-Action= consumer_start, consumer_stop, and consumer_pop, but there isn't one that indicates that the consumer rejoins the queue after the caller hangs up. Is there anyway I can detect this particular event? Thanks for your help. Woody

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487's ...
by Shelby Ramsey 11 Nov '08

11 Nov '08

Hello FS'ers, I'm having an issue where I get a call and send it via a profile ... but if the "b" leg cancels the request (a 487) then fs does not respond resulting in lots of 487's coming back at me ... Of course I'm not sure why I'm getting 487's from the b leg side ... SR

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Socket outbound: How to bridge two calls?
by Dennis 11 Nov '08

11 Nov '08

hi, i am using socket outbound and want to bridge two calls with each other. i do the following: a call comes in and it gets answered (inbound call). after i send the answer to the inbound, i do an originate to &park (outbound). after i answered the outbound, i want both sides to be able to talk to each other. but when i try uuid_bridge <inbound-uuid> <outbound-uuid> it does not work. i do not even get a channel_bridge or something. when i do 2 originates and send them to &park(), i can bridge them with each other with uuid_bridge. what am i doing wrong? the other question is, how to unbridge two calls, if they are bridged? i can't find anything like an unbridge command. thanks dennis93

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Problem with conditions depending on values generated by dialplan
by Helmut Kuper 11 Nov '08

11 Nov '08

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I want a dialplan setting a channel variable with a certain value. After that I want to do some action depending of that variable/value. But it looks like that FS parses the regular expressions of dialplan conditions only at the beginning of dialplan_hunt, so that conditions based on channel variables are only possible with the variable values at the time of starting dialplan_hunt. Is that true or is there a way to do what I like to do ? regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iEYEARECAAYFAkkZf78ACgkQ4tZeNddg3dzW7gCffmFMZrR9h1eDPZrXXAxrkJ5/ z/cAnjxmYAvHqdxAGJ8iIlrV+h5fv+1Q =F9QJ -----END PGP SIGNATURE-----

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freepy and outbound event socket
by Novak Joe 11 Nov '08

11 Nov '08

Hi, I'd like to know if there are any python examples floating around which use outbound event socket for mod_event_socket. I've recently had some problems with mod_python and sqlalchemy which have been causing freeswitch to dump core on a regular basis. I haven't been able to solve this, and have instead decided to switch to mod_event_socket and freepy. I've been looking through the scripts/socket/ dir in the freeswitch source, and the following wiki pages: http://wiki.freeswitch.org/wiki/Event_Socket http://wiki.freeswitch.org/wiki/Event_socket_outbound there is a pretty nice, simple perl example (sock.pl) using outbound socket, which I've played around a bit with, but the freepy examples don't seem to have anything similar; rather they all seem to use inbound socket. If anyone is using python/twisted for outbound socket applications (e.g. to answer calls) I would appreciate any pointers, or trivial examples similar to sock.pl In the meantime I'll try to come up with something of my own. Cheers

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Can anybody tried with Avya to free switch trunk.....?
by sambasivarao Vemula 11 Nov '08

11 Nov '08

Hi, I want establish a trunk between avaya communication manager to freeswitch (h323/Sip trunk ). Is any body trying to establish these trunk ..? Please forward the details with configuration settings from avaya side and freeswitch side. Regards Samba DISCLAIMER ========== This e-mail may contain privileged and confidential information which is the property of Persistent Systems Ltd. It is intended only for the use of the individual or entity to which it is addressed. If you are not the intended recipient, you are not authorized to read, retain, copy, print, distribute or use this message. If you have received this communication in error, please notify the sender and delete all copies of this message. Persistent Systems Ltd. does not accept any liability for virus infected mails.

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RPID ...
by Shelby Ramsey 11 Nov '08

11 Nov '08

Hello, Any way to disable sending / using RPID in the dialplan and not at the SIP profile level? Thanks! SR

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SIP packets sending rport instead of 5060
by Josh Forman 11 Nov '08

11 Nov '08

I've been trying to send out SIP calls from a softphone to an outside line. the call makes it to the freeswitch box and is sent to our cisco router, but the source port is coming up as "rport" instead of 5060. In what may be a related problem when the router sends messages the freeswitch box, freeswitch does not return an ACK message. The gateway and sip profile settings are pasted below. <gateway name="main-outbound"> <param name="username" value="1000"/> <param name="realm" value="xxx.xxx.xxx.xxx"/> <param name="password" value="1234"/> <param name="register" value="false"/> <param name="caller-id-in-from" value="true"/> <param name="contact-params" value="tport=tcp"/> </gateway> <settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${outbound_codec_prefs}"/> <param name="hold-music" value="$${hold_music}"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="manage-presence" value="false"/> <param name="aggressive-nat-detection" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="false"/> <param name="rtp-timeout-sec" value="1800"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="ext-rtp-ip" value="$${external_rtp_ip}"/> <param name="ext-sip-ip" value="$${external_sip_ip}"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> </settings>

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Linksys/SPA3000 port problem.
by henkoegema 10 Nov '08

10 Nov '08

The Linksys/SPA3000 has two ports (1 FXS and 1 FXO port) The FXS interface (ext. 2014 in my case) is on port 5090 (profile name: doublenat and sip-port 5090) and 2014 is registered ok with FS. What to do with the FXO interface? This is what I've tried: I've created a new profile ( doublenat2) with sip-port 5091 ? I've put the FXO interface (ext. 2015) on port 5091, and 2015 is registerd ok with FS. Is this the right way froward? :working: I can call 2014 with: <action application="bridge" data="sofia/doublenat/2014%my.dyndns.domain"/> When I try to call 2015 with: <action application="bridge" data="sofia/doublenat2/2015%my.dyndns.domain"/> the softphone shows: Call rejected: 480 temporarily Unavailble CLI: 2008-11-11 00:17:29 [ERR] switch_ivr_originate.c:1064 switch_ivr_originate() Cannot create outgoing channel of type [sofia] cause: [USER_NOT_REGISTERED] 2008-11-11 00:17:29 [INFO] mod_dptools.c:1848 audio_bridge_function() Originate Failed. Cause: USER_NOT_REGISTERED The PSTN line in not actually connected o the FXO line. (I'm testing for somebody else. I don't have a PSTN line myself) Could that be the reason that it's not working properly? -- View this message in context: http://www.nabble.com/Linksys-SPA3000-port-problem.-tp20430421p20430421.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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