November 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] DAHDI issue
by Russell.Mosemann＠cune.org 05 Nov '09

05 Nov '09

> 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!

1 0

DAHDI issue
by Russell.Mosemann＠cune.org 05 Nov '09

05 Nov '09

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!

1 0

[ERR] mod_portaudio.c:974 Cannot find an input device
by Frank Carmickle 05 Nov '09

05 Nov '09

Hello I updated to 15376 added some build depends and still no joy. Any more pointers. Thanks. --FC

1 0

Transfer call to group
by Albano Daniele Salvatore - Lavoro 05 Nov '09

05 Nov '09

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/15(a)192.168.0.77 bridge() Data for bridge application is ${group_call(${dialed_extension}+F@${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

1 0

Calling more than 1 variable in condition
by Ahmed Munir 05 Nov '09

05 Nov '09

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> Kindly advise for this issue. -- Regards, Ahmed Munir

2 1

FS hangup
by Lars Zeb 05 Nov '09

05 Nov '09

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

2 4

Re: [Freeswitch-users] no REINVITE on Blind Transfer with bypass_media
by Humberto Quintana 05 Nov '09

05 Nov '09

>> -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/514xxxxxx 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.com>wrote: >>>>> >>>>>> 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

1 0

Java example
by dipen＠entvoice.com 05 Nov '09

05 Nov '09

Hi, Can you please paste me your sample java dialplan code that work for you ? ..coz m also facing the same problem. My mod_java is loaded properly. Also /usr/lib/jvm/java-1.5.0-gcj-4.3-1.5.0.0/jre/lib/i386/client/libjvm.so and freeswitch.jar in java.conf.xml is specified properly. I have written a java code to print HIIIIIIIIII on the console but its not printing. Level mentioned is INFO. on FS console it just prints EXECUTE sofia/internal/1004@192.168.1.144:5061 java(testing.class) I am attaching my java code herewith. Can u just tell me where more i should do the modification to get my dialplan work. Waiting for your kind reply. Thanks & Regards, Dipen Velani On Fri 19/12/08 4:09 AM , kriko <kristjan.ugrin(a)gmail.com> wrote: > Seems like my dialplan was a bit problematic, it works now. > Thanks. > On Thu, 18 Dec 2008 15:19:22 +0100, Anthony Minessale > wrote: > > did you turn up your console log level high enough to see it? The > default > > level is "INFO" > > > > > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > > > -- > kriko > _______________________________________________ > Freeswitch-users mailing list > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ---- Msg sent via @Mail - http://atmail.com/

3 4

Filtering a particular event.
by Nagalenoj H. 05 Nov '09

05 Nov '09

Hi, I've tried to filter the events like below to filter a particular event. 1) register for all events 2) filter for one unique-id 3) filter only one/more events(ex: DTMF & CHANNEL_EXECUTE) So, I want to receive only these events for the specific unique-id. But, I am receiving other events too. I'm using perl ESL outbound. Is it possible to do like this?! -- Regards, Nagalenoj H.

2 1

Skypiax load error
by 大泥人 05 Nov '09

05 Nov '09

Hi All, I once meet the Skypiax load error issue and some guys infomed me that there is no configuration file for Skypiax. When I follow these intructions -> http://wiki.freeswitch.org/wiki/Skypiax#Config_files_location_and_script_to… I still have some difficulties unstanding this: ." So, go and copy src\mod\endpoints\mod_skypiax\configs/skypiax.conf.xml to Debug\conf\autoload_configs." I did not find such directories in my freeswitch folder. Did not understand what "src" means, I checked the freeswitch folder and did not find such a folder named"src". There is a folder named"mod" under freeswitch while look into "mod" folder there are only some DLL files and can not find endpoints and etc. 2.You'll probably build the "Debug" version I just build this from the precompiled binaries and then launched FS. I'm not sure what I launched is in debug mode or not. If anyone can offer some help that really be appriciated. Thanks Daniel Zeng From: freeswitch-users-request(a)lists.freeswitch.org Subject: FreeSWITCH-users Digest, Vol 41, Issue 40 To: freeswitch-users(a)lists.freeswitch.org Date: Thu, 5 Nov 2009 06:57:44 -0800 Send FreeSWITCH-users mailing list submissions to freeswitch-users(a)lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request(a)lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner(a)lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." --附转发的邮件-- From: anthony.minessale(a)gmail.com To: freeswitch-users(a)lists.freeswitch.org Date: Thu, 5 Nov 2009 08:13:46 -0600 Subject: Re: [Freeswitch-users] no REINVITE on Blind Transfer with bypass_media 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 <hjqlopez(a)hotmail.com> 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/514xxxxxx 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.com>wrote: >>> >>>> 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 >> >> >> _________________________________________________________________ >> Windows Live: Friends get your Flickr, Yelp, and Digg updates when they >> e-mail you. >> http://go.microsoft.com/?linkid=9691817 >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 > > _________________________________________________________________ > Ready. Set. Get a great deal on Windows 7. See fantastic deals on Windows 7 now > http://go.microsoft.com/?linkid=9691818 _________________________________________________________________ Windows Live: Make it easier for your friends to see what you’re up to on Facebook. http://go.microsoft.com/?linkid=9691816 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)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_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 --附转发的邮件-- From: rob4manhere(a)gmail.com To: freeswitch-users(a)lists.freeswitch.org Date: Thu, 5 Nov 2009 08:52:05 -0600 Subject: Re: [Freeswitch-users] Setting up Conference with Moderator Hi UK, From what I've done and read, the caller-controls (in conference.conf.xml) can be modified to almost anything you can think of, BUT, they are mapped 1-to-1 to a conference- ie you can't map a caller control just for those with the moderator flag. So unless you want everyone able to mute/kick everyone then you can't do it. The wiki seems to indicate this as well: "Be aware that the caller-controls are applied across the entire conference. You cannot enter one member of the conference using caller- controls ABC and then enter a second member using caller-controls XYZ." http://wiki.freeswitch.org/wiki/Mod_conference I think this might be a limitation of mod_conference. Perhaps one of the pros can chime in if I'm off-base or there's some nifty way to accomplish this. Cheers, Rob On Nov 4, 2009, at 8:09 PM, Ujjval Karihaloo wrote: > Any ideas on the below...has anyone implemented the below: > > Once I have the Moderator and Participants logged on, how do I > invoke the moderator previlidges, LIk esay muting everyone/someone > or kicking someone out of the Conf and the like? > > > > -----Original Message----- > From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org > ] On Behalf Of Ujjval Karihaloo > Sent: Monday, November 02, 2009 12:52 PM > To: freeswitch-users(a)lists.freeswitch.org > Subject: Re: [Freeswitch-users] Setting up Conference with Moderator > > Rob: > > Once I have the Moderator and Participants logged on, how do I > invoke the moderator previlidges, LIk esay muting everyone/someone > or kicking someone out of the Conf and the like? > > > > -----Original Message----- > From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org > ] On Behalf Of Rob Forman > Sent: Friday, October 30, 2009 9:34 AM > To: freeswitch-users(a)lists.freeswitch.org > Subject: Re: [Freeswitch-users] Setting up Conference with Moderator > > Hm, strange. I haven't seen that before. Can you pastebin your logs > at debug level? > > On Oct 30, 2009, at 9:43 AM, Ujjval Karihaloo wrote: > >> It's strange... a tcpdump tells me that there is no DTMF from my >> provider when using IVR, but when I call into a TN that goes >> directly into the Conference App, I see DTMF from the provider. >> >> >> >> -----Original Message----- >> From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org >> ] On Behalf Of Rob Forman >> Sent: Friday, October 30, 2009 7:23 AM >> To: freeswitch-users(a)lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >> >> I've never had any problem with that. Is your logging at debug level >> so you can see the RECV DTFM in the log/fs_cli? Are you calling from >> a SIP phone on the pbx, or via a PSTN provider? Maybe your provider >> isn't passing them through. >> >> Make sure your logging is turned up then try something simpler, like >> calling the echo application, and see if DTFM comes through. >> >> Rob >> >> On Oct 29, 2009, at 11:34 PM, Ujjval Karihaloo wrote: >> >>> Rob: >>> >>> For some reason, I don't see the DTMF appear on the fs_CLI when >>> using the below configuration....so it basically timesout. >>> >>> UK >>> >>> >>> >>> -----Original Message----- >>> From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org >>> ] On Behalf Of Ujjval Karihaloo >>> Sent: Monday, October 26, 2009 9:21 AM >>> To: freeswitch-users(a)lists.freeswitch.org >>> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >>> >>> Thx a lot Rob, reading the wiki your way or using IVR seems >>> correct.. >>> =============== >>> The wiki also says that the wait-mod might be "used in conjunction >>> with an IVR where the moderators are authenticated with an extra >>> pass- >>> code", which is what I did. I guess that's why I didn't understand >>> the point of the +pin. >>> ====================== >>> >>> I will try it out. >>> >>> Again thx a lot for your help. Will keep everyone posted. >>> >>> ________________________________________ >>> From: freeswitch-users-bounces(a)lists.freeswitch.org [freeswitch-users-bounces(a)lists.freeswitch.org >>> ] On Behalf Of Rob Forman [rob4manhere(a)gmail.com] >>> Sent: Friday, October 23, 2009 12:22 PM >>> To: freeswitch-users(a)lists.freeswitch.org >>> Subject: Re: [Freeswitch-users] Setting up Conference with Moderator >>> >>> I just re-tested with the pin in my dial plan: >>> >>> <action application="conference" data="conference 123456@default >>> +flags{}+1234" /> >>> >>> And it doesn't challenge me for the pin. I just drop right in. I >>> figured this is how it was intended, since the wiki says the pin is >>> set initially and only challenged in later attempts [by future >>> callers]: >>> >>> "The first time a conference name (confname) is used, it will be >>> created on demand, and the pin will be set to what ever is specified >>> at that time: the pin in the data string if specified, or if not, >>> the >>> "pin" setting in the conference profile, and if that is also >>> unspecified, then there is no pin protection. Any later attempt to >>> join the conference must specify the same pin number, if one existed >>> when it was created. " >>> >>> >>> The wiki also says that the wait-mod might be "used in conjunction >>> with an IVR where the moderators are authenticated with an extra >>> pass- >>> code", which is what I did. I guess that's why I didn't understand >>> the point of the +pin. >>> >>> I'm sure there's a scenario where its used and useful, the wiki just >>> doesn't explain it. >>> >>> Rob >>> >>> On Oct 23, 2009, at 12:43 PM, Brian West wrote: >>> >>>> Well first off you're not defining a pine here... >>>> >>>> confname@profilename+flags{mute|deaf|waste|moderator}+[conference >>>> pin >>>> number] >>>> >>>> That might be why its not asking for a pin. >>>> >>>> /b >>>> >>>> On Oct 23, 2009, at 12:30 PM, Rob Forman wrote: >>>> >>>>> <entry action="menu-exec-app" digits="1" param="conference >>>>> 123456@default+flags{}" /> >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users(a)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(a)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(a)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(a)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(a)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(a)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(a)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(a)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(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org --附转发的邮件-- From: rob4manhere(a)gmail.com To: freeswitch-users(a)lists.freeswitch.org Date: Thu, 5 Nov 2009 08:57:08 -0600 Subject: [Freeswitch-users] Wideband / HD phones 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 _________________________________________________________________ 约会说不清地方？来试试微软地图最新msn互动功能！ http://ditu.live.com/?form=TL&swm=1

3 2