[Asterisk-Users] problems with dialing
Hi, We're having problems dialing out to Asterisk from our Grandstream GXP-200 phones. About 2 of 3 times, when we dial, nothing happens. Looking at the console in max debug mode, there are no messages except the following: Feb 24 10:29:20 WARNING[2475]: chan_sip.c:1208 retrans_pkt: Maximum retries exceeded on transmission 9913b47bcd7[EMAIL PROTECTED] for seqno 4524 (Critical Response) Note: Early dial is set to Yes. DTMF is via SIP info. The phones are connected via a wireless bridge, range extender, and router to the asterisk box. Pinging the phone from the Asterisk box reveals a fairly long latency: 64 bytes from 192.168.10.100: icmp_seq=1 ttl=250 time=1110 ms64 bytes from 192.168.10.100: icmp_seq=2 ttl=250 time=114 ms64 bytes from 192.168.10.100: icmp_seq=3 ttl=250 time=21.8 ms64 bytes from 192.168.10.100: icmp_seq=4 ttl=250 time=33.4 ms64 bytes from 192.168.10.100: icmp_seq=5 ttl=250 time=4.46 ms64 bytes from 192.168.10.100: icmp_seq=6 ttl=250 time=57.4 ms Could this be the source of the problem? If so, would appreciate tips on how to work around this. Thanks in advance, WILL ___ --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] sound file installation problem
I downloaded asterisk-sounds-1.2.0-beta1, superused, then typed "make install". The installation stopped with the following error: No description for sounds/access-code.gsmmake: *** [datafiles] Error 1 Does anyone have any useful tips? I'm running Debian 3.0. Thanks, WILL ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] wifi phones - desk
Hi, I'm provisioning an office with limited cabling. I'm looking for a desk based wifi phone. Most of the ones I've seen are handsets. Any ideas? Thanks, WILL ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] "Please Press Any Key to Accept a Call"
Hi, I'd like to add a feature to my asterisk system that tries to find a user among a couple of locations, and then goes to internal voicemail if the user doesn't pick up. (e,g, an internal extension and a cell phone). The catch is that I want the user to manually accept the call to prevent it from going (for example) to the voice mail on my cell phone. Scenario * Call comes in, outside caller dials "100" * Desk phone for user Joe rings. No answer * Joe's house phone rings. * Joe's wife picks up and hears a voice "Please press any key to accept a call for extension 100." * Joe's wife hangs up. * Joe's cell phone rings. * Joe picks up and hears a voice "Please press any key to accept a call for extension 100." * Joe presses 1 and says "Hello this is Joe". Alternately, in the penultimate step * Cell voice mail picks up. * Voice says "Please press any key to accept a call for extension 100". No keys pressed since it's a voice mail * Call is routed to Asterisk voicemail. It seems straight forward to try multiple locations, but I'm not seeing how to only patch the call through if the user responds with a key press. Thanks, WILL ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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: "Please Press Any Key to Accept a Call"
BJ, Great news - glad to hear it. I think the key thing I'm looking for is that this is all transparent to the caller. I want them to hear nice hold music while the user is searched for, and only be directed to the physical extension if the correct person picks up. (e.g. no third party voice mail). CF - That sounds good, I'll search the archives. Possibly an application would still be of service if it simplifies the dial plan. Look forward to trying both approaches. Best, WILL ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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: Editing Asterisk config files with WORD Pad
On Windows, I really like TextPad (shareware - www.textpad.com ) for editing text files. It handles the line ending issue well (you can save in PC or unix format). It has great search and replace functionality. And you can edit a bunch of files at once. Best, WILL Message: 1 Date: Tue, 15 Nov 2005 09:19:24 -0700 From: Chuck Bunn <[EMAIL PROTECTED]> Subject: [Asterisk-Users] Editing Asterisk config files with WORD Pad To: Asterisk - Users Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Hi, I have tried editing some Asterisk config files (ie sip.conf) in MS Word Pad and I have saved the files as 'Unicode Text Document' with quotes around the full file name => "sip.conf" and then uploaded the files to a Linux server using FileZilla. When I do this the config files fail to work. Although I am somewhat proficient at using 'vi' I find it easier to cut and paste with Word Pad (I know I can cut and paste with 'vi'... ). Is this possible to do or am I all wet... Thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] problems with international dialing
Hi,I'm struggling to set up a plan to allow international dialing from my US location. As I understand it, an international call can have 9 to 15 digits including country code. The problem is that the call always goes through after I've entered the 9th digit.
My service provider is BroadVoice, my phones are Grandstream GXP-200's. DTMF is set on the phone to be via SIP INFO and Early Dial is set to "Yes".I have this in my plan:[outbound-long-distance-forio]
; this is how outbound long distance calls are handled; international numbers are max 15 digits included 2-3 digit country code (E-164 ITU protocol); we assume 9-15 digitsexten => _011X., 1, Macro(forio-dial-outbound,default,${EXTEN})
; long distanceexten => _1NXXNXX, 1, Macro(forio-dial-outbound,default,${EXTEN})Appreciate any tips. Thanks! WILL-- Forio Business Simulations Will Glass-Husain
[EMAIL PROTECTED]www.forio.com
___
--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] problems with DTMF
Hi,I'm struggling a bit with DTMF. It seems to work fine on my internal network, but when I call outside lines with telephone trees, some systems understand the DTMF and some ignore it. Anyone have tips on solving this? Thanks in advance. My local phone is a Grandstream GXP-200mailbox=89username=89secret=xtype=friendhost=dynamiccontext=internal-foriodtmfmode=infocanreinvite=noAnd my provider is BroadVoice host=sip.broadvoice.comfromdomain=sip.broadvoice.comfromuser=4153730175secret=username=4153730175authname=4153730175 type=peeruser=phonecontext=inbound-forioinsecure=verycanreinvite=no-- Forio Business Simulations Will Glass-Husain[EMAIL PROTECTED] www.forio.com ___ --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] disconnect with mute
Hi, I'm having problems with an occasional disconnect from phone calls while my phone is on mute. This is a problem with long conference calls, for example. I've a GrandStream GXP-2000 and Asterisk 1.2.1. Anyone have experience with similar issues? Best, WILL ___ --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
