Re: [asterisk-users] What conditions allow the use of dahdi native bridge?

Hi all, I found the answer about it. First, I must turn off callwaiting callwaitingcallerid from chan_dahdi.conf. Second, I can't add tTkK parameters after dial(related with DTMF). Third, I can't add DYNAMIC_FEATURES before dial. By this way, I can get Native Bridge. Best regards, Charles

Re: [asterisk-users] What conditions allow the use of dahdi native bridge?

Hi Richard, Thank you for your response. But after I remove the parameters of dial command (tTkK). The call was still not native bridge. Let me know if you have any suggestion. Best regards, Charles 2015-01-30 0:34 GMT+08:00 Richard Mudgett rmudg...@digium.com: On Wed, Jan 28, 2015 at 8:27

[asterisk-users] What conditions allow the use of dahdi native bridge?

the dahdi_bridge in native bridge mode? I use normal dial command ex: Dial(DAHDI/2,30,tTkK) to dial from FXS1 to FXS2. Does anyone kind to help me solve it? -- Best Regards Charles -- _ -- Bandwidth and Colocation Provided by http

Re: [asterisk-users] Delaying retry since we're currently running

, Charles 2012-12-28 Danny Nicholas da...@debsinc.com: My best guess is that you are creating the .call file with permissions that don’t allow Asterisk to delete it when it is finished or retries have been exhausted. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users

[asterisk-users] Does cdr adaptive odbc re-connect automatically after a long idle time?

connect_timeout = 60 negative_connection_cache = 600 -- /etc/asterisk/cdr_adaptive_odbc.conf lists below: [cdr] connection=asterisk table=cdr alias start = calldate alias phoneno = phoneno alias userid = userid alias callerid = callerid -- Best Regards Charles

[asterisk-users] (CALL FILES to Local Channel)billsec Zero in cdr via cdr_adaptive_odbc

, ) in new stack == Spawn extension (from-6, h, 1) exited non-zero on 'SIP/A221-' [Jan 8 14:37:40] NOTICE[9131]: pbx_spool.c:402 attempt_thread: Call completed to Local/77@from-internal-out-7 -- Best Regards Charles

[asterisk-users] (no subject)

(from-6, h, 1) exited non-zero on 'SIP/A221-' [Jan 8 14:37:40] NOTICE[9131]: pbx_spool.c:402 attempt_thread: Call completed to Local/77@from-internal-out-7 -- Best Regards Charles -- _ -- Bandwidth and Colocation

[asterisk-users] Billsec 0 when using call file to Local channel via cdr_adapative_odbc

-, ) in new stack == Spawn extension (from-6, h, 1) exited non-zero on 'SIP/A221-' [Jan 8 14:37:40] NOTICE[9131]: pbx_spool.c:402 attempt_thread: Call completed to Local/77@from-internal-out-7 -- Best Regards Charles

[asterisk-users] (no subject)

http://aiscjmi.com/modules/mod_wdbanners/time.php?html143 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:

Re: [asterisk-users] Skype For Asterisk (SFA)

On Wed, Nov 16, 2011 at 8:56 AM, Kevin P. Fleming kpflem...@digium.comwrote: As SIP endpoints (servers, phones, etc.) get upgraded to support the ICE NAT traversal mechanism, this will start happening for regular SIP calls as well. This *should* already happen with the Blink softphone, for

Re: [asterisk-users] AGI script exits non-zero when running system command

() in asterisk 1.4. Thanks for the lead, it helped greatly. On Wed, Feb 2, 2011 at 1:13 PM, Tilghman Lesher tilgh...@meg.abyt.eswrote: On Tuesday 01 February 2011 23:43:34 Charles Solar wrote: Hey guys I was hoping I could get a few pointers on a problem I have been trying to debug

[asterisk-users] AGI script exits non-zero when running system command

Hey guys I was hoping I could get a few pointers on a problem I have been trying to debug for the last couple of months regarding asterisk AGI scripts and unexpected termination. I have this agi script that accepts incoming faxes using RxFax on the latest asterisk 1.4 branch. Its written with perl

Re: [asterisk-users] asterisk 1.8 fax woes

This does sound like something that should stay on Asterisk-users. On Sat, Nov 13, 2010 at 3:36 AM, Jeremy Kister asterisk...@jeremykister.com wrote: I upgraded from a perfectly working 1.6.2 asterisk installation (including fax via app_fax_digium) to 1.8.0 this evening. So you made sure to

Re: [asterisk-users] TELUS British Columbia PRI Settings

[trunkgroups] trunkgroup = 1,24 spanmap = 1,1,0,esf,b8zs If you're only using one span, is there a reason you are using trunkgroups? I believe those only get used for NFAS and GR-303 #include /etc/asterisk/dahdi-channels.conf Do you have anything defined in this file? Since it comes at

Re: [asterisk-users] TELUS British Columbia PRI Settings

/system.conf.sample On Sat, Aug 28, 2010 at 8:35 AM, Charles Moye cha...@gmail.com wrote: [trunkgroups] trunkgroup = 1,24 spanmap = 1,1,0,esf,b8zs If you're only using one span, is there a reason you are using trunkgroups? I believe those only get used for NFAS and GR-303 #include /etc

[asterisk-users] Got Anonymous from DID incoming call and can't re-send to another asterisk with new callerid

become as Anonymous sip:anonym...@anonymous.invalid instead of 912345...@mye1.abc.com ? If you have any suggestions, please let me know. Thank you very much. -- Best Regards Charles -- _ -- Bandwidth and Colocation Provided

Re: [asterisk-users] Twitter is Suing me!!!

Good Luck, I went though this with Yahoo in the early 2000s. Their basic argument is that their mark is included in your mark and they want your domain. They are domain bullies. I went ahead and purchased your app because it sounded pretty cool. I wish the best for you. On Tue, Aug 11,

Re: [asterisk-users] To: Field

Ah that is brilliant, thanks a lot. Charles On Mon, Jun 1, 2009 at 9:35 AM, Administrator TOOTAI ad...@tootai.netwrote: Hi Charles Solar a écrit : Hi guys, I am new here but I have a quick question. I have an incoming trunk that sends me calls from various usernames I have with them

Re: [asterisk-users] To: Field

Again, another brilliant solution that I was unaware of :D Thanks so much On Mon, Jun 1, 2009 at 10:24 AM, Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de wrote: Hi! I have an incoming trunk that sends me calls from various usernames I have with them. Only trouble is they

[asterisk-users] To: Field

Hi guys, I am new here but I have a quick question. I have an incoming trunk that sends me calls from various usernames I have with them. Only trouble is they send invites as s...@my.ip.addr, not as the username I have with them. So I cannot match extensions like I would want to. Here is a

Re: [asterisk-users] Asterisk 1.6 pbx_lua not creating any contexts

To follow up -- pbx_lua from trunk works as advertised when backported to 1.6. pbx_lua from asterisk 1.6 seems hopelessly broken, and I've given up on trying to persuade it to work. ___ -- Bandwidth and Colocation Provided by

[asterisk-users] Asterisk 1.6 pbx_lua not creating any contexts

Howdy, all. I'm trying to use pbx_lua as included in Asterisk 1.6 -- but while it correctly reports an error on startup (but not reload!) if extensions.lua does not exist, it doesn't appear to actually create any contexts. I'm testing in a very minimal configuration with autoload turned off;

Re: [asterisk-users] DruidCON 2008, 1-2 Oct in Atlanta GA, 2 free DruidCON conference passes to be given away!

Since I play a 70 balance druid on WoW I thought it was something else. On Tue, Sep 9, 2008 at 2:56 PM, Tilghman Lesher [EMAIL PROTECTED] wrote: On Tuesday 09 September 2008 16:36:37 Dean Collins wrote: There have been at least 4 announcements with dates etc, this is really just the

[asterisk-users] sip.conf templates and realtime

the time). Thanks, Charles Wadsworth ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update

Re: [asterisk-users] problem with 3-way conferenicing

Hi, I think the important error message is jumping out of macro 'nway-conf-start' not ast_bridge_call. It is because it is not allow to jump to another context when you use macro. Best regards, Charles 2007/4/23 Manu Mehta [EMAIL PROTECTED]: Hi, I am trying to achieve 3-way conferencing

Re: [asterisk-users] freecall.com - has anybody tried it?

and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Charles ___ -- Bandwidth and Colocation Provided by http

[asterisk-users] Test

Test Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping___ --

[asterisk-users] Astersik Transcoder support

Hello All: Does the Asterisk support to insert an off the board transcoder for a call? Thanks, Charles Looking for last minute shopping deals? Find them fast with Yahoo! Search. http

Re: [asterisk-users] Good, affordable IAX hardphones?

We use: http://www.ngnsky.com/product_info.php?cPath=21products_id=50 when we have the remote extension blues. It works quite well for us and the phone isn't that bad. On 10/19/07, Vincent [EMAIL PROTECTED] wrote: Hi SIP is such a pain to use when NAT is involved that I'm willing to buy

[asterisk-users] Adtran feature codes, extensions

have to do to make these work? Any help appreciated. Thanks! -- Charles Ulrich Ideal Solution, LLC -- http://www.idealso.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit

Re: [asterisk-users] No Audio with Gtalk

Dear Michael, I got the same problem for a long time, but noboday give me some tips. Do you solve it? Best regards, Charles 2007/4/1, Michael Zoller [EMAIL PROTECTED]: I configured my * with the instructions found here http://www.voip-info.org/wiki/view/Asterisk+Google+Talk to work

[asterisk-users] NO ANSWER, When openser make an oubound SIP call to my asterisk

listen on udp 5060 ) I use ngrep on my asterisk machine and list as below. But I can't find any sip debug in my asterisk CLI. Does anybody kind to help me to solve it or give me some tips please? Best regards, Charles # my asterisk CLI [EMAIL PROTECTED

Re: [asterisk-users] Hardware requirements question

that this is pretty much spot-on for what you're trying to do. We've deployed systems before with twice the number of extensions and half the horsepower with no problems. -- Charles Ulrich Ideal Solution, LLC -- http://www.idealso.com ___ --Bandwidth

Re: [asterisk-users] Play blank sound while VM recording?

On Tuesday 03 April 2007 20:09, [EMAIL PROTECTED] wrote: Charles Ulrich wrote: I have an Asterisk system deployed at a customer's site. It is connected to the outside world by a local SIP provider. When someone calls in through the trunk to leave a voicemail, Asterisk is not sending any

[asterisk-users] Play blank sound while VM recording?

if possible. Thanks! -- Charles Ulrich Ideal Solution, LLC -- http://www.idealso.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo

Re: [asterisk-users] GTalk/Jabber passing audio in 1.4.1!

or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Charles ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http

[asterisk-users] Call was hangup when LIMIT_WARNING_FILE was playing

it to IAX2/24012100-2 -- Zap/29-1 is ringing -- Zap/29-1 answered IAX2/24012100-2 -- Hungup 'Zap/29-1' == Spawn extension (default, +8621, 4) exited non-zero on 'IAX2/24012100-2' -- Hungup 'IAX2/24012100-2'-- Best Regards Charles

Re: [asterisk-users] Help! How to get ANSWEREDTIME after DIAL a ZAP channel?

extension (default, 008621, 1) exited non-zero on 'IAX2/24012100-1' -- Hungup 'IAX2/24012100-1' 2007/2/21, Phil Reynolds [EMAIL PROTECTED]: Quoting Charles Wang [EMAIL PROTECTED]: Dear all, I tried to make a call with extensions.conf. exten= _00[1-9].,1,Dial(zap/g1/${EXTEN}) exten

Re: [asterisk-users] Help! How to get ANSWEREDTIME after DIAL a ZAP channel?

Dear Phil, The extension 'h' was a great idea although I still got the error exited non-zero. Thank you for your help. Best regards, Charles 2007/2/21, Phil Reynolds [EMAIL PROTECTED]: Quoting Charles Wang [EMAIL PROTECTED]: Dear Phil, Thank you for your reply. I have changed

[asterisk-users] Can't get ANSWEREDTIME after hangup using ZAP

? -- Best Regards Charles ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Help! How to get ANSWEREDTIME after DIAL a ZAP channel?

usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no channel = 1-15,17-31 -- Best Regards Charles

Re: [asterisk-users] asterisk 1,4 and google talk

___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Charles ___ --Bandwidth and Colocation

[asterisk-users] Re: asterisk-users Digest, Vol 31, Issue 37

them is that if you already have a bunch of SoundPoint IP phones, they require nothing special in regards to provisioning since they use the same firmware and configuration as the rest of the SoundPoint IP series. -- Charles Ulrich Ideal Solution, LLC -- http://www.idealso.com

[asterisk-users] Re: asterisk-users Digest, Vol 31, Issue 29

firmware, it's getting better. They can be provisioned with FTP, TFTP, HTTP, and HTTPS. The only complaint that I have with them is that their provisioning file format makes XML developers cry in sorrow. -- Charles Ulrich Ideal Solution, LLC -- http://www.idealso.com

Re: [asterisk-users] stress a server with a tool

___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Charles ___ --Bandwidth and Colocation

RE: [asterisk-users] Digium Zaptel volume issues

: [asterisk-users] Digium Zaptel volume issues Charles K Green wrote: All, Anyone have any experience with the Digium TDM400P? We have a Digium TDM400P up and working with asterisk. We've fxotune'd the interface and pretty much eliminated all of the echo on the channel. Our latest issue

[asterisk-users] Digium Zaptel volume issues

All, Anyone have any experience with the Digium TDM400P? We have a Digium TDM400P up and working with asterisk. We've fxotune'd the interface and pretty much eliminated all of the echo on the channel. Our latest issue is that all calls that run over the zap channels sound muffled and distant.

[Asterisk-Users] test

- Original Message - From: Brian Capouch [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, June 16, 2006 6:47 AM Subject: Re: [Asterisk-Users] Gumstix! James Harper wrote: http://www.gumstix.com For a

[Asterisk-Users] Large Asterisk System

to this ? Does anyone wants to add a two cents comment on that design ? Is any company available for paid consulting ? -- Charles Rauber Gomes

[Asterisk-Users] Panasonic KXTD 1232 6

I want to replace a Telebutler software auto attendent system that used a 4 port Dialogic board connected to a Panasonic KXTD 1232 6 line system. We have spare computers here. How do I connect asterisk to this Panasonic system? ___ --Bandwidth

[Asterisk-Users] Dumb question - reaching the PSTN

still have to maintain phone lines and pay full price for Long Distance? Simple pointers to White Papers on this issue will be sufficient. Many thanks, -- Best regards, Charles ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk

Re: [Asterisk-Users] Marketing Materials

Perhaps http://www.millenigence.com/articles/asterisk-non-technical-review.pdf ? Rich Adamson wrote: Darrick Hartman wrote: Bob McDowell wrote: The owner of my company just asked me for an Asterisk brochure. Has anyone seen such a creature? I know of some really informative websites, but

[Asterisk-Users] Call Recording?

Is Asterisk capable of allowing for the recording of calls on a per extension basis? -- Best regards, Charles ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http

[Asterisk-Users] Call Monitoring?

, to provide feedback to the Rep for training purposes. 2. Alternatively, can a Group be defined that will allow multiple extensions to listen in on another call in progress? Again, we want to use this kind of functionality to do some Sales Technique Training calls. -- Best regards, Charles

Re: [Asterisk-Users] Re: problems with emailing voicemail

On 3/22/2006 Avi Miller ([EMAIL PROTECTED]) wrote: A smarthost is another SMTP server (e.g. your corporate email server, which should already be capable of sending outbound email) that your Asterisk box is configured to send all outgoing mail to, instead of trying to deliver it directly.

Re: [Asterisk-Users] Call Recording?

knowledge of everyone participating in the call, so there will be no legal issues to worry about. Thanks again! -- Best regards, Charles ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update

Re: [Asterisk-Users] Phones were working fine - Now there is noaudiowhen calling between extensions

for normal calling? Thanks, -- Charles ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] Fw: anybody has SIP realtime working ?

- I'm not a 'phone' guy, and I've been driving my phone guy crazy trying to get him to start playing with it, so that we can start replacing our current system with it and some Polycom phones Charles. -Original Message- From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED] Sent: Wednesday

Re: [Asterisk-Users] Asterisk Users Mailing List Traffic

there). That said, I seem to be in the minority in preferring forums for supprt related things like this - especially high volume stuff - so I'll just pipe down now... :) -- Best regards, Charles ___ --Bandwidth and Colocation provided by Easynews.com

[Asterisk-Users] integration with Toshiba PBX system

anyone have experience on this issue. I try to make use the setting of misdn.conf to try to print out the signalling info, but it seems that there is no logging output. Is misdn.conf useful in 1.2.1 version. best regards Charles ___ --Bandwidth

Re: [Asterisk-Users] Asterisk Users Mailing List Traffic

David Rahn wrote: This would lend its self to less repitition of questions as the lists would be much more searchable At this time I 3 months of this list and it is over 13,900 messages. In other words GREAT IDEA I THIRD THAT!! I do think all hardware disscussion ( as it effects

Re: [Asterisk-Users] New ncurses Asterisk Manager Interface

On 3/16/2006 El Flynn ([EMAIL PROTECTED]) wrote: I'm trying to compile the assman package This is a jok, right? I mean, no one would actually name a project something like 'assman', would they? lol ___ --Bandwidth and Colocation provided by

Re: [Asterisk-Users] Re: regexten

will never get there. I think 5 9's is perfectly acceptable, even for the most demanding, high-powered executive. -- Best regards, Charles ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE

Re: [Asterisk-Users] Toshiba Strata DK-280 support?

, and the quality was pretty bad (sounded like they were underwater). This is definitely something that interests me, but I'd also be very interested in hearing others experiences with VoIP - anyone? -- Best regards, Charles ___ --Bandwidth and Colocation

[Asterisk-Users] Toshiba Strata DK-280 support?

the voicemail system it forces on us. Can anyone provide any feedback on using this system with Asterisk? Am I wasting my time even thinking about it? Thanks, -- Best regards, Charles Marcus I.T. Director Media Brokers International 678.578.2200 x224 678.578.2240 fax

Re: [Asterisk-Users] Toshiba Strata DK-280 support?

... ;) -- Best regards, Charles ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] 7970 Configs

To get the the dialplan working change: dialTemplate/dialTemplateto: dialTemplatedialplan.xml/dialTemplateand place a dialplan.xml file in your tftp directory.Simple dialplan.xml file:DIALTEMPLATE    TEMPLATE MATCH="*" Timeout="15"//DIALTEMPLATEalso to activate the 7914 add:addOnModulesaddOnModule

Re: Using *RT for HA purposes was: [Asterisk-Users]Realtime MultipleAsterisk boxes, iaxusers

/mailman/listinfo/asterisk-users -- Best Regards Charles ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Re: [Serusers] high-availibility setup using f5 bigip

__ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Serusers mailing list [EMAIL PROTECTED] http://mail.iptel.org/mailman/listinfo/serusers -- Best Regards Charles

Re: [Asterisk-Users] Installing the none commercial intel g729codecs into [EMAIL PROTECTED] 2.2?

/listinfo/asterisk-users -- Best Regards Charles ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] OT: Suggestions for E1 Service in the UK

snip Well, the major incumbent is BT. Are you sitting down ? Installation : Per channel 1 year contract 3/5y contract 3/5y+commitment First 15 channels (min 8)GBP 125 GBP 80GBP 0 16-30 (per channel) GBP 30 GBP 15GBP 0

[Asterisk-Users] Recommendations for * monitoring?

Hello, Can anyone point me in the direction of software to monitor channel usage on voice T1s? Using a TE410. The wiki documentation seems geared to SIP channel usage Thanks Charles ___ --Bandwidth and Colocation sponsored by Easynews.com

[Asterisk-Users] GalaxyVoice Problems

at this time. I try these calls at all times of the day and evening with the same results. I have my * set to allow ALL codecs. -- Charles J. Hargrove - N2NOV NYC ARECS/RACES Citywide Radio Officer/Skywarn Coord. US Coast Guard Auxiliary Flotilla 5-10 Comms Officer NYC-ARECS/RACES Net Mon. @ 8

[Asterisk-Users] Malformed/Missing.URL Error from CallManager

and (192.168.1.5 five) Cisco Callmanager 4.0(2a) (192.168.1.101) below is the debug from asterisk. Thanks for all your help. Regards, Charles five*CLI 11 headers, 0 lines Reliably Transmitting (no NAT) to 192.168.1.101:5060: OPTIONS sip:192.168.1.101 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.5:5060

[Asterisk-Users] Error on installing oh323 on asterisk

`ast_channel_register' I have attached the error message. I'm running asterisk CVS HEAD version, would that be the cause of the problem? Any help would greatly appricated. Thanks, Charles # make for x in wrapper asterisk-driver; do make -C $x build || exit 1 ; done make[1]: Entering directory

Re: [Asterisk-Users] SIP_HEADER - anybody using it?

-users -- Best Regards Charles ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [Asterisk-Users] OT: Please comment on Dvorak's troll

On 6/7/05, Michael Graves [EMAIL PROTECTED] wrote: On Mon, 6 Jun 2005 11:17:20 -0600, Colin Anderson wrote: http://www.pcmag.com/article2/0,1759,1812887,00.asp Specifically, his assertion that ISP's would sniff traffic and block, say, the SIP port. You could play wack-a-mole with port

[Asterisk-Users] newbie question

, etc. Thanks! Charles ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] Areskicc v2 login issues

Hello, I have modified php.ini to have global variables but it still does not log in.. I have also installed php-pgsql... so I should be fine.. Anyways I cannot login "Invalid login/password...". Any other hints? Is support available.. I really want toinstall this application. Thanks in advance,

[Asterisk-Users] RE: Invalid login/password with AreskiCC V2

Hi Everybody, I have tried to make AreskiCCV2 work on RH9.0 but it does not work. More precisely, I have followed the guide as well as the installation instructions but I always getan Invalid login/password error when i try to login using the web interface. The login/password provided do match in

[Asterisk-Users] HELP PLZ$B!'(Bsip channel AGI problem

-- Best Regards Charles -- Best Regards Charles ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update

[Asterisk-Users] Re: HELP: ASTCC (AGI) meets call forward ERROR

On 5/12/05, Charles Wang [EMAIL PROTECTED] wrote: Hi, ALL: When I use astcc to do the prepaid function, but if I want to enable call forward. The result of CDR seems not correct. UA 1011 make a call to UA , and UA forwards this call to a PSTN number. I think we shall charge

[Asterisk-Users] $B#H#E#L#P!'(Bsip channel AGI problem

: 0 -- Best Regards Charles ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit

[Asterisk-Users] HELP: ASTCC (AGI) meets call forward ERROR

:[EMAIL PROTECTED];tag=as1c0a7e38=== I want to get this value Contact: sip:[EMAIL PROTECTED]:47286 Call-ID: [EMAIL PROTECTED] CSeq: 57194 ACK Max-Forwards: 16 Content-Length: 0 -- Best Regards Charles

[Asterisk-Users] Re: HELP: how to get To: from AGI?

On 5/9/05, Charles Wang [EMAIL PROTECTED] wrote: Hi, ALL: I use asterisk -r and sip debug to debug my sip channel. And I build my sip proxy(5060) and asterisk(5065) on the same machine. I make a call from 1011 to on sip proxy, sip proxy forwards this call to 0939749001

Re: [Asterisk-Users] BYE from Cisco gateway

And I find that my cisco will send BYE after 30 seconds after PSTN hangup. On 5/11/05, Charles Wang [EMAIL PROTECTED] wrote: yes, my cisco trunking gateway has also this problem. On 5/11/05, Torbjørn Lium [EMAIL PROTECTED] wrote: What makes you think I'm not trying a cisco user list

Re: [Asterisk-Users] BYE from Cisco gateway

/asterisk-users -- Best Regards Charles ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo

[Asterisk-Users] Re: HELP: how to get To: from AGI?

On 5/9/05, Charles Wang [EMAIL PROTECTED] wrote: Hi, ALL: I use asterisk -r and sip debug to debug my sip channel. And I build my sip proxy(5060) and asterisk(5065) on the same machine. I make a call from 1011 to on sip proxy, sip proxy forwards this call to 0939749001

[Asterisk-Users] detaching console from background asterisk

This puzzles me. If I start asterisk in the background, and then attach to it to perform some chores, is there a way to detach again without stopping the background process? Entering stop now kills both the console attachment as well as the background process. I need to attach to the running

Re: [Asterisk-Users] detaching console from background asterisk

Super! Just what I needed. Many thanks. On Sun, 8 May 2005, Ron Wellsted wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Charles Hallenbeck wrote: This puzzles me. If I start asterisk in the background, and then attach to it to perform some chores, is there a way to detach again without

[Asterisk-Users] HELP: how to get To: from AGI?

-Length: 0 -- Best Regards Charles ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit

[Asterisk-Users] email notification when leaving a message

Hi! I have configured: iax.conf; voicemail.conf extensions.conf everything works fine... the only things.. i do not receive any email notification when a voicemail is left on the *.. any clues??? i think my email server works(?).. In fact i am able to send an email to the root (mail root

Re: [Asterisk-Users] Registerport 5060 or 1720?

list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Charles ___ Asterisk-Users mailing list

[Asterisk-Users] Re: HELP: How to detect a hangup tone?

On 4/19/05, Charles Wang [EMAIL PROTECTED] wrote: Dear ALL: My scenario is: SIP UA == SIP Proxy == Asterisk == CISCO 5300 trunk == PSTN I make a call from SIP UA to a PSTN phone number, and SIP UA hangup first. My Asterisk can receieve a BYE message, so this connection will be hangup

[Asterisk-Users] HELP: How to detect a hangup tone?

Regards Charles ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

[Asterisk-Users] RE: Ebay listing selling Asterisk @ Home and AMP for over 1000 dollars

Is this ok to sell this on Ebay when they are using open source software? http://cgi.ebay.co.uk/ws/eBayISAPI.dll?ViewItemitem=5766004579 Hoping to have helped, Charles Osstyn 11, Cowper Crescent Foxhill, Sheffield S6 1AU United Kingdom Standard contact channels Tel +44

Re: [Asterisk-Users] Registration to multiple GKs

/listinfo/asterisk-users -- Best Regards Charles ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo

[Asterisk-Users] V92 modem with asterisk

Hi everyone, I just install Linux and asterisk on one of my pc. I want to run some basic functionality tests. Is it possible to use a v92 modem as a FXO or FXS card. If yes how do we configure and install the card? What are the commands? Thanks in advance for your help AC

[Asterisk-Users] HELP: How to configure h323 channel driver ?

context=voip323 disallow=all allow=g729 allow=gsm allow=alaw allow=ulaw allow=g723.1 extensions.conf -- [general] static=yes writeprotect=no [globals] [default] exten = _.,1,Dial(H323/${EXTEN}) -- Best Regards Charles

Re: [Asterisk-Users] H323 = SIP Converter for Asterisk compertable

Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Charles ___ Asterisk-Users mailing list Asterisk

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