Re: [Freeswitch-users] Wideband / HD phones
I'd rather go with the ip7000 since it has better audio gear in it. For a deskphone everything Polycom = ip450 is absolutely wideband enough for a deskphone. Personally I am currently a total fan of the VVX which is a video deskphone with the same audio as a IP6000 Am 05.11.2009 um 18:45 schrieb Frank Carmickle: On Fri, Nov 06, Steve Underwood wrote: If your idea of high def is G.722 there are more conventional phones for half that price. And there is portaudio in freeswitch itself. With a usb headset and celt at 48k how can you go wrong? --FC ___ 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 Michal Bielicki HaloKwadrat | ul. Polna 46/14, 00-644 Warszawa t. +48228753290 | f. +48228753291 michal.bieli...@halokwadrat.pl | w. www.halokwadrat.pl Knowledge Low Prices. Guaranteed! ___ 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] no REINVITE on Blind Transfer with bypass_media
-You could check the sofia debug for r15332 here: http://pastebin.freeswitch.org/11008 Phone/Devices: The caller is the DID provider's Switch. The callee (which also sends the REFER) is Asterisk 1.4.26. Testing with other devices(linksys SPA962,Grandstream GXV3000) gathers the same results. I did not ask you to send me a ladder diagram. I asked you to send me a console trace from FreeSWITCH using latest trunk (1.0.4 does not help me) 1) start FreeSWITCH 2) run the cli command: console loglevel debug 3) run the cli command: sofia profile internal siptrace on 4) reproduce your issue and put the trace on freeswitch pastebin http://pastebin.freeswitch.org (login and pass are stated in the auth dialog) Also please answer brian's question. What phones and/or sip devices are involved in this call. On Wed, Nov 4, 2009 at 3:39 PM, Humberto Quintana wrote: Thanks for your time, -The scenario is still the same: Always bypass media. Environment 100% NAT free :-) Call established from A to B through FS. Then... Blind transfer from B to C (Refer-to: C) RTP should go directly between A and C. -With 1.0.4 and 1.0.5pre3, FS actually INVITEs C after receiving the REFER-to:C, BUT there is no 2-way audio. Only RTP from C to A (due to the lack of reINVITE to A, after C answers). Please check SIP diagram here: http://provision.netcelerate.net/ngrep/blindxfer2009-11-04-v1.0.5pre3.html -What it's wrong with r15332 is there is not such call to C. For sure I know SIP is a protocol, may be my description was not clear but this SIP diagram speaks by itself ;-) http://provision.netcelerate.net/ngrep/blindxfer2009-11-04rev15332.html -You could check the sofia debug for r15332 here: http://pastebin.com/m6f2b3836 Best regards, Humberto I don't know what you are talking about anymore. The scenario I had tested is when a call is bridged in bypass_media=true bridge and you blind transfer that call back to the dialplan as soon as it hits the routing state it will resume media. it has been confirmed to not work and confirmed to have been fixed several time and if you are still having a problem you must have something blocking some of your packets or something . You have to understand that sip is a protocol and your description is completely non-standard. Perhaps you should get a console trace and attach it to a jira. The trace probably makes more sense to me. sofia profile internal siptrace on console loglevel debug reproduce and attach the whole capture. On Tue, Nov 3, 2009 at 6:05 PM, Humberto Quintana wrote: Hi, I tried r15332 and set in the sofia profile: a) bypass_media_after_bridge=true only b) bypass_media_after_bridge=true, param name=media-option value=resume-media-on-hold/ In both cases FS is hanging up the initial call (A to FS) after accepting the REFER to C: A - reINVITE with FS' SDP - FS A - 200 - FS A - ACK - FS A - BYE - FS The call to C is not even tried. I found this line is the logs that could give some idea: 2009-11-03 18:29:41.280707 [NOTICE] mod_sofia.c:733 Hangup sofia/external/514xx at a.b.c.d [CS_ROUTING] [RECOVERY_ON_TIMER_EXPIRE] after sending the ACK for the reINVITE Regards, Humberto please try r15326 I think i have it working. I recommend for optimal results you set bypass_media_after_bridge=true either as a global or in your DP in place of bypass_media=true On Mon, Nov 2, 2009 at 4:30 PM, Humberto Quintana hotmail.comwrote: Hi Mike, I re-tried with trunk rev 15319 but I got almost the same behavior: There is now a reINVITE (with FS' SDP) going to A when the REFER is accepted. But still there is no reINVITE for A (with C's SDP) after the call from FS to C is established. Anyway, we decided for now to do a different implementation but if you want to explore more in this issue count me in ;-) Thank you very much! Humberto __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com _ Ready. Set. Get a great deal on Windows 7. See fantastic deals on Windows 7 now http://go.microsoft.com/?linkid=9691818 ___ 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] FS hangup
fixed in 15376 On Thu, Nov 5, 2009 at 12:35 PM, Anthony Minessale anthony.miness...@gmail.com wrote: yes sounds like a bug. I think i redid it and forgot to check for true still =0 On Thu, Nov 5, 2009 at 12:17 PM, Lars Zeb larc...@yahoo.com wrote: Thanks for the help. Yes, I am using a lua script to handle inbound calls with continue_on_fail set to true: session:execute(set, continue_on_fail=true); I changed it to: session:execute(set, continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,NO_ROUTE_DESTINATION); and it works OK now. Did something change between v15311 to v15372 to make this behave differently? I ask because it worked with “true” in the earlier version. *From:* freeswitch-users-boun...@lists.freeswitch.org [mailto: freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Anthony Minessale *Sent:* Thursday, November 05, 2009 9:48 AM *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] FS hangup do you have continue_on_fail set? if you do you have to include no_answer,busy etc once you set it, you have to set *everything* you want. On Thu, Nov 5, 2009 at 11:27 AM, Lars Zeb larc...@yahoo.com wrote: I just updated to v15372 from v15311. When calling into FreeSWITCH, it hangs up the call rather than going to voicemail (line 262 in pastebin). I don’t know what might be causing this. Can anyone help? Thanks, Lars http://pastebin.freeswitch.org/11006 Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 i386 GNU/Linux ___ 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/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org 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] FS hangup
yes sounds like a bug. I think i redid it and forgot to check for true still =0 On Thu, Nov 5, 2009 at 12:17 PM, Lars Zeb larc...@yahoo.com wrote: Thanks for the help. Yes, I am using a lua script to handle inbound calls with continue_on_fail set to true: session:execute(set, continue_on_fail=true); I changed it to: session:execute(set, continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,NO_ROUTE_DESTINATION); and it works OK now. Did something change between v15311 to v15372 to make this behave differently? I ask because it worked with “true” in the earlier version. *From:* freeswitch-users-boun...@lists.freeswitch.org [mailto: freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Anthony Minessale *Sent:* Thursday, November 05, 2009 9:48 AM *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] FS hangup do you have continue_on_fail set? if you do you have to include no_answer,busy etc once you set it, you have to set *everything* you want. On Thu, Nov 5, 2009 at 11:27 AM, Lars Zeb larc...@yahoo.com wrote: I just updated to v15372 from v15311. When calling into FreeSWITCH, it hangs up the call rather than going to voicemail (line 262 in pastebin). I don’t know what might be causing this. Can anyone help? Thanks, Lars http://pastebin.freeswitch.org/11006 Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7 10:39:21 EDT 2009 i686 i686 i386 GNU/Linux ___ 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/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org 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
[Freeswitch-users] Does OpenZap support CTR21?
Hello As an alternative to more expensive alternatives like OpenVox or Sangoma, I'd like to order an X100P clone from www.x100p.com for use in France. According to a PDF on the site, the reason this card gets bad reviews is that the Silicon labs Si3012/Si3035 DAA chip used in the original Digium X100P card and low cost X100P clone cards only supports FCC mode. However, the Si3014/Si3034 DAA chip used on the X100P SE supports global line standards. As for software, the Silicon labs Si3014/Si3034 DAA chip used in the X100P SE supports 600 Ohm impedance and complex impedance to meet CTR21 line standards. However, the Zaptel wcfxo driver only supports CTR21 mode with 600 Ohm AC termination, which may or may not be the correct setting depending on the country and the phone system in use. So... does someone know if OpenZap, which is apparently required in addition to Zaptel/Dahdi for FreeSwitch to work PCI TDM cards, supports CTR21? Thank you. -- View this message in context: http://old.nabble.com/Does-OpenZap-support-CTR21--tp26217371p26217371.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ 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] Calling more than 1 variable in condition
On Wed, Nov 4, 2009 at 10:22 PM, Ahmed Munir ahmedmunir...@gmail.comwrote:
Hi,
In my dial plan I've created a variable named SIP_CALL, PSTN_CALL. If
SIP_CALL = true, it dials out to sip call, when PSTN_CALL=true, it dials out
to landline call, as I provide sample below;
condition field=${SIP_CALL} expression=^true$
action application=log data=INFO SIP CALL/
action application=bridge data=sofia/${domain}/$1/
/condition
condition field=${PSTN_CALL} expression=^true$
action application=log data=INFO PSTN CALL/
action application=bridge data=sofia/gateway/$1/
/condition
The problem I'm facing is how can I apply condition when I've to call more
than 1 variables? Like if there are no records in SIP numbering plan table
and PSTN numbering plan table so it get the digits and dial out the to
carrier (how to apply AND operation in condition?) i.e.
condition field=*${SIP_CALL} and ${PSTN_CALL}* expression=^false$
action application=log data=INFO GET DTMF/
action application=getdigits data=values for getdigits/
action application=bridge data=sofia/gateway/$1/
/condition
AND operations are very simple - just stack the conditions:
condition field=${Field1} expression=expr1/
condition field=${Fiedl2} expression=expr2
-- actions here --
/condition
note that you must close the first condition's tag! BTW, this is covered in
the dialplan section of the wiki. :)
-MC
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[Freeswitch-users] Transfer call to group
Hi,
actually i'm trying to setup an IVR that, when the choice is done,
transfer the call to a group, really simply.
Here the dialplan in default context to handle call to group (four
extensions, one for group, from 2001 to 2004)
http://pastebin.freeswitch.org/11014
Here the output log
http://pastebin.freeswitch.org/11015
When i call the group directly from a telephone in the default context
or when the ivr transfer me to the group i didn't get nothing, looking
to log you can see (line 153)
EXECUTE sofia/internal/1...@192.168.0.77 bridge()
Data for bridge application is
${group_call(${dialed_extension...@${domain_name})}
It's, probably, a stupid error, but the only other way to accomplish
this is to bridge individually phones using | as separator but i would
to mantain a single extension to handle this stuff.
Thanks for your support!
Best Regards,
Daniele
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Re: [Freeswitch-users] Dialpan: try.. finally analogs
It cleans up after itself fine, but it is an indication of some issue in the code we need to address. if you can reproduce this in svn trunk, please file a bug on jira.freeswitch.org with details how to reproduce. mike On Nov 5, 2009, at 12:44 PM, Artem Shiyanov wrote: Hello! I have to deal with classic problem: Leaking stream handle in FS console. I also know the reason - firstly channel is sent to the extension with playback and later it is transfered to another extensions with execute_extension or, another trouble-case - channel is bridged to some addres. I do not ask (but I wish to) why FS doesn't close stream automatically when channel is gone. I ask whether it is possible to use some try.. finally construction in diaplan? If yes then I can simply stop playback in the finally block.. Any thoughs? ___ 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
Re: [Freeswitch-users] Does OpenZap support CTR21?
This would be specific to the zaptel driver for that card, not openzap. mike On Nov 5, 2009, at 1:43 PM, Fred-145 wrote: Hello As an alternative to more expensive alternatives like OpenVox or Sangoma, I'd like to order an X100P clone from www.x100p.com for use in France. According to a PDF on the site, the reason this card gets bad reviews is that the Silicon labs Si3012/Si3035 DAA chip used in the original Digium X100P card and low cost X100P clone cards only supports FCC mode. However, the Si3014/Si3034 DAA chip used on the X100P SE supports global line standards. As for software, the Silicon labs Si3014/Si3034 DAA chip used in the X100P SE supports 600 Ohm impedance and complex impedance to meet CTR21 line standards. However, the Zaptel wcfxo driver only supports CTR21 mode with 600 Ohm AC termination, which may or may not be the correct setting depending on the country and the phone system in use. So... does someone know if OpenZap, which is apparently required in addition to Zaptel/Dahdi for FreeSwitch to work PCI TDM cards, supports CTR21? ___ 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] DAHDI issue
Debian 5.0.3 FreeSWITCH Version 1.0.trunk (15376M) openzap and libpri-1.4.10.2 dahdi-linux-complete-2.2.0.2+2.2.0 Digium Wildcard TE110P T1/E1 Card (running as a T1) This was working with zaptel. I thought that I would upgrade from zaptel to DAHDI, but it's generating no such device or address errors. FS is running as root but can't seem to see the channels. I have unloaded and loaded the drivers. Permissions look fine. The dahdi tools can see the card. Any insights? http://pastebin.freeswitch.org/11016 -- Russell Mosemann Concordia University, Nebraska See http://www.cune.edu/ for the latest news and events! ___ 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] [ERR] mod_portaudio.c:974 Cannot find an input device
Hello I updated to 15376 added some build depends and still no joy. Any more pointers. Thanks. --FC ___ 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] DAHDI issue
I thought that I would upgrade from zaptel to DAHDI, After I send the message, the answer comes to me. I guess that's the way things work. :-) I had forgotten to define the channels in /etc/dahdi/system.conf. Here are the settings, and things are working. Thanks for listening. :-) span=1,1,0,esf,b8zs bchan=1-23 dchan=24 loadzone = us defaultzone=us echocanceller=mg2,1-23 -- Russell Mosemann Concordia University, Nebraska See http://www.cune.edu/ for the latest news and events! ___ 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] sip profile question
The internal.xml also has an ext-rtp-ip variable and in trying to understand what this is for (my version of fs is 1) I noticed in trunks conf file it is explained. So the available options that I have given my setup is multihomed with a lan/wan setup where the wan interface is dynamic would be a fqdn for fs to lookup, or auto/auto-nat. How exactly does auto and auto-nat work so I may know of its going to work properly/reliably in my scenario. Thanks! jlc ___ 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] Bug in Freeswitch/scripts/gentls_cert.in build file?
Here is what is believed to be a bug found by Robert Hadley found in
Freeswitch1.0.4/scripts/gentls_cert.in build file:
Fix for gentls_cert remove to work:
[scripts]# diff gentls_cert.in gentls_cert.in~
129c129
if [ -d ${CONFDIR}/CA ]; then
---
if [ ! -d ${CONFDIR}/CA ]; then
Best Regards,
Jerry
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Re: [Freeswitch-users] sip profile question
auto-nat tries to use upnp/nat-pmp to figure it out... auto will just put your IP in there. The other values can be stun:host or an IP. The docs in trunk show this now... its really simple to understand but you should NEVER have to set that unless you have a nat scenario that requires you to lie about your IP and such to traverse the nat. /b On Nov 5, 2009, at 4:54 PM, Joseph L. Casale wrote: The internal.xml also has an ext-rtp-ip variable and in trying to understand what this is for (my version of fs is 1) I noticed in trunks conf file it is explained. So the available options that I have given my setup is multihomed with a lan/wan setup where the wan interface is dynamic would be a fqdn for fs to lookup, or auto/auto-nat. How exactly does auto and auto-nat work so I may know of its going to work properly/reliably in my scenario. Thanks! jlc ___ 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] Bug in Freeswitch/scripts/gentls_cert.in build file?
In the future please post issues to jira.freeswitch.org along with a
diff -u from the root freeswitch source directory.
This already seems to be fixed in svn trunk can you verify.
Thanks,
Brian
On Nov 5, 2009, at 5:00 PM, Jerry Richards wrote:
Here is what is believed to be a bug found by Robert Hadley found
in Freeswitch1.0.4/scripts/gentls_cert.in build file:
Fix for gentls_cert remove to work:
[scripts]# diff gentls_cert.in gentls_cert.in~
129c129
if [ -d ${CONFDIR}/CA ]; then
---
if [ ! -d ${CONFDIR}/CA ]; then
Best Regards,
Jerry
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Re: [Freeswitch-users] sip profile question
auto-nat tries to use upnp/nat-pmp to figure it out... auto will just put your IP in there. The other values can be stun:host or an IP. The docs in trunk show this now... its really simple to understand but you should NEVER have to set that unless you have a nat scenario that requires you to lie about your IP and such to traverse the nat. Thanks for the fast reply Brian, so bear with me here... I am just about to go live w/ my first fs box as I move away from a year or two with Asterisk. So I don't have upnp/nat-pmp, I guess auto would be my next choice, but if the box is multihomed, how does it decide which of the two (well more as I am going to use vlans) ip's to stick in there? I guess I could use a public stun server, but if there is a self contained way for me to handle it, I would rather do that so that I don't have to worry about someone else's stun server being up so my fs box functions. Thanks! jlc ___ 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 profile question
If you're on a public IP you have no need for ext-rtp-ip or ext-sip-ip REMOVE them. If your multi homed then you'll need to set them.. we don't listen on 0.0.0.0 you'll have to start a profile for each IP you wish to listen on. /b On Nov 5, 2009, at 5:20 PM, Joseph L. Casale wrote: Thanks for the fast reply Brian, so bear with me here... I am just about to go live w/ my first fs box as I move away from a year or two with Asterisk. So I don't have upnp/nat-pmp, I guess auto would be my next choice, but if the box is multihomed, how does it decide which of the two (well more as I am going to use vlans) ip's to stick in there? I guess I could use a public stun server, but if there is a self contained way for me to handle it, I would rather do that so that I don't have to worry about someone else's stun server being up so my fs box functions. Thanks! jlc ___ 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] FusionPBX
FYI, the latest Windows SVN build now includes the option to configure FusionPBX, a port of the pfsense/FreeSWITCH gui: http://fusionpbx.com/index.php If you plan to install it someplace other than the default location of C:/FreeSWITCH just make sure to update the paths in Admin, System Settings from the FusionPBX web interface. The default username for the GUI is *admin*, password *fusionpbx* Here's the link: http://files.freeswitch.org/windows_installer/freeswitch.exe At this time FusionPBX utilizes sqlite for its data store. The author, mcrane, plans to release a new version soon with support for a MySQL, or PostgreSQL backend. regards, Carlos ___ 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] evaluate variable through cli
How does one show the assigned value that a variable such as
$${local_ip_v4} or $${domain} might have through the cli?
Thanks,
jlc
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Re: [Freeswitch-users] evaluate variable through cli
global_getvar local_ip_v4
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca
On 5-Nov-09, at 3:40 PM, Joseph L. Casale wrote:
How does one show the assigned value that a variable such as
$${local_ip_v4} or $${domain} might have through the cli?
Thanks,
jlc
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[Freeswitch-users] Want 183 w/SDP, but Get 200 w/SDP
I am trying to make a call through a Gateway that sends the INVITE with no SDP and ONLY wants the 200 OK w/SDP when the callee answers. For some reason, Freeswitch answers the call with 200 OK w/SDP even before the callee answers the phone. Is this to provide ringback? Can I disable that action? Best Regards, Jerry ___ 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 profile question
If you're on a public IP you have no need for ext-rtp-ip or ext-sip-ip REMOVE them. If your multi homed then you'll need to set them.. we don't listen on 0.0.0.0 you'll have to start a profile for each IP you wish to listen on. I am multihomed, and the wan nic is dynamic. Is there any way for me to control how it guesses the IP of a `specific` interface without the use of a third party (stun etc). Thanks, jlc ___ 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 profile question
Just use ${local_ip_v4} then.
and enable auto-restart on the sofia.conf.xml
/b
On Nov 5, 2009, at 5:45 PM, Joseph L. Casale wrote:
I am multihomed, and the wan nic is dynamic. Is there any way for me
to
control how it guesses the IP of a `specific` interface without the
use
of a third party (stun etc).
Thanks,
jlc
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Re: [Freeswitch-users] Want 183 w/SDP, but Get 200 w/SDP
This all depends on what you're doing in your dialplan if you do stuff like record it requires media and will trigger it. A sip trace or some such debug would be more helpful then a terse description of a problem. /b On Nov 5, 2009, at 5:49 PM, Jerry Richards wrote: I am trying to make a call through a Gateway that sends the INVITE with no SDP and ONLY wants the 200 OK w/SDP when the callee answers. For some reason, Freeswitch answers the call with 200 OK w/SDP even before the callee answers the phone. Is this to provide ringback? Can I disable that action? Best Regards, Jerry ___ 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
Re: [Freeswitch-users] evaluate variable through cli
vars.xml
but watch out the core will trump local_ip_v4 if it happens to change.
/b
On Nov 5, 2009, at 5:40 PM, Joseph L. Casale wrote:
How does one show the assigned value that a variable such as
$${local_ip_v4} or $${domain} might have through the cli?
Thanks,
jlc
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Re: [Freeswitch-users] mod_skypiax for OSX?????
2009/11/6 Giovanni Maruzzelli gmar...@celliax.org On Thu, Nov 5, 2009 at 6:57 PM, Seven Du dujinf...@gmail.com wrote: Ciao Giovanni, Do you still plan to merge this? Sorry Seven, I've lost track of this, and now I'm so sick I'm completely un-useful ;). That's OK, we all have a lot of things to do each day. But yes, I would like to do it, if you think it is in a useful state. Can you please create a Jira and attach an svn diff, so in the next days I can merge it? I'd like to create a jira and I think it would be easier if you can directly merge from branch. However the branch is a bit old and it would need some days if you need svn diff based on the current trunk. Thanks. -giovanni 2009/9/5 Giovanni Maruzzelli gmar...@celliax.org Seven, thanks a lot for your efforts. I will merge it in the next days, and I will take care that it will not breaks Windows or Linux. If I find problems I will wait for you having more time in the future. I send you my super best wishes for your personal things to go well and solves in the best of the possible ways. ciao for now, -giovanni Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 On Sat, Sep 5, 2009 at 1:13 PM, Seven Dudujinf...@gmail.com wrote: gm, Thanks a lot you can merge into the mainline. I check into my branch because it's currently not as useful as on Linux and Windows and the solution is not good. But it works and it is a good start that mod_skypiax runs on OSX. Sure it would be easier for people want to test and improve it if it been merged into trunk. I think you can make a diff by svn diff -r 14472:14772 http://svn.freeswitch.org/svn/freeswitch/branches/seven/src/mod/endpoints/mod_skypiax FYI for personal reason I won't have much time put on this in the coming month. Actually the code was done a few weeks ago but i only got a chance to commit it yesterday. Sure that is not to say I cannot do but fixes. But can you please make sure it won't break Linux/ windows build when you merge the code? I haven't have a chance to test all of them yet. -7- On Sep 5, 2009, at 4:49 PM, Giovanni Maruzzelli wrote: Seeeven! I saw the modification you made on the wiki page... You made it, mod_skypiax runs on OSX Let's merge your mods on the mainline, plese ;-))) -giovanni Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 ___ 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 ___ 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 -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 ___ 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
Re: [Freeswitch-users] Wideband / HD phones
If you need a really cheap entry-level phone that does Polycom's HD Siren codecs then check out the IP 335 that just came out. It's very basic but I'm hearing good things from people who've used them. -MC On Thu, Nov 5, 2009 at 6:57 AM, Rob Forman rob4manh...@gmail.com wrote: Hey all, Looking at buying some high def phones. Any recommendations (preferably based on experience) for hardware based on product quality, standards compliance, features integration with Freeswitch, etc? Thank you! Rob Forman ___ 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
Re: [Freeswitch-users] sip profile question
Just use ${local_ip_v4} then.
and enable auto-restart on the sofia.conf.xml
Cool, it seems to always use the public ip, quite reliably.
That is what I am after (why), is there something in the code that
forces it to favor for example, non RFC 1918 addresses?
It works, I just want to understand exactly how and why rather
than be oblivious:)
Thanks for all the advice!
jlc
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Re: [Freeswitch-users] FusionPBX
Screenshots for the FusionPBX graphical interface http://fusionpbx.com/files/fusionpbx_com/screenshots/index.php --- On Thu, 11/5/09, Carlos Talbot carlos.tal...@gmail.com wrote: From: Carlos Talbot carlos.tal...@gmail.com Subject: [Freeswitch-users] FusionPBX To: freeswitch-users@lists.freeswitch.org Date: Thursday, November 5, 2009, 4:28 PM FYI, the latest Windows SVN build now includes the option to configure FusionPBX, a port of the pfsense/FreeSWITCH gui: http://fusionpbx.com/index.php If you plan to install it someplace other than the default location of C:/FreeSWITCH just make sure to update the paths in Admin, System Settings from the FusionPBX web interface. The default username for the GUI is admin, password fusionpbx Here's the link: http://files.freeswitch.org/windows_installer/freeswitch.exe At this time FusionPBX utilizes sqlite for its data store. The author, mcrane, plans to release a new version soon with support for a MySQL, or PostgreSQL backend. regards, Carlos -Inline Attachment Follows- ___ 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
[Freeswitch-users] Events in mod_perl
Hi all, Is there any way to receive events while running a perl program with the help of mod_perl?? I've seen some functions related to sending and receiving events in the mod_perl wiki. But I don't know how to use that. Any 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
