RE: [asterisk-users] Extensions +International
I pick up the handset and get a dialtone. I press 9011331234567 or something international. Before I can finish, the local option kicks in because it saw 9. Is there a way to say _9[2-9]NXXX or something like that? Are you sure the handset is not processing that call string, making it local before it even gets submitted to Asterisk? I know in the Snom's that that can happen if your number matching is hokey... Stefano ___ --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] freepbx with ASTERISK 1.4
it's possible to configure freepbx 2.2 with asterisk 1.4? Look here for the archives: http://lists.digium.com/pipermail/asterisk-users/ Search for the subject FreePBX 2.2.0 and Asterisk 1.4.0. You'll find EXACTLY what you're looking for. :-) Stefano ___ --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] Feeding digit input to PauseQueueMember
Hello, I'm trying to figure out how to do something that I hope is pretty easy. I have a remote phone system (Definity ProLogix) connected to my Asterisk system via a T1 cable (all onsite). I'd like to get some of these users on a queue hosted on the Asterisk. I've got it setup so that it seems to work OK (calls flow normally), but I'd like the users to be able to dial one extension to run PauseQueueMember, and another to do UnpauseQueueMember. Is something like this possible? Answer Playback (what extension to pause) Get input --- how do I do that? PauseQueuemember (input from user) Playback (agent paused) Hangup I have done most of this already in other contexts, but I cant figure out how to get input from the user? Is there a function for that? What is it? Thanks, Stefano ___ --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] Feeding digit input to PauseQueueMember
Is something like this possible? Answer Playback (what extension to pause) Get input --- how do I do that? PauseQueueMember (input from user) Playback (agent paused) Hangup Eventually I found it: The Read Application http://www.asteriskguru.com/tutorials/read.html Or http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Read Stefano ___ --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] Toll-free dialing via PRI problem
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Irvin Sent: Wednesday, January 31, 2007 10:09 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Toll-free dialing via PRI problem Sheepishly, that was the magic bullet. Thanks Trevor!! Tim Trevor Peirce [EMAIL PROTECTED] wrote: Jerry Jones wrote: From asterisk, you do not hear anything other than ringing as it does not cut the audio path through until it receives the answer from the far end, hence the steady ringing. So instead of Dial(Zap/g1/1800xxx,,r) just do Dial(Zap/g1/1800xxx,,) so early audio can make it through. Unless there's more to the puzzle? My setup was also fixed. Thanks Trevor! :-D Stefano ___ --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] Talkoff
Hello all, I'm trying to see if I can finally get rid of a talkoff problem that I've been having with my Asterisk server since I started messing with it over a year ago. Currently, I'm running it on SuSE 10.1 box with Asterisk 1.4. I'm using Snom 360s with the set. My setup is one where the PSTN connects to a legacy PBX (Definity), then connects, via a T1 cable, to a 4 port T1 card in the Asterisk. Doing research over the past few months, I've seen that I should use dtmfmode=rfc2833 and ensure that I use ulaw. The sip.conf section referring to my extension looks like this: [5257] type=friend secret=5257 record_out=Adhoc record_in=Adhoc qualify=no port=5060 nat=never mailbox=5257 host=dynamic dtmfmode=rfc2833 disallow=gsm,alaw dial=SIP/5257 context=from-internal canreinvite=no callerid=device 5257 My Snom 360 is set to use G.711u first, which I presume is ulaw. I don't notice the problem on every call, just some calls. It always seems to be to calls going outside, but that's most of the calls I deal with( as opposed to staying on the Asterisk or bridging to the Definity). The person calling me never hears the tones, just me. What can I look at to get more clues? Cheers, Stefano ___ --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] Talkoff
What is the manufacturer and model of the 4-port T1 card? I have had talkoff with the TE406 (1st gen echo canceller), and have heard of talkoff occurring with relaxdtmf=yes in zapata.conf. Hey there. I do believe it it a Digium TE406 with Echo Canceller. I can't remember how many times I've read about relaxdtmf :-) Yes, that's set to no in zapata.conf. So: - if you have a TE406, disable hardware DTMF detection Interesting about the TE406 though. How does one turn off the hardware DTMF detection? I imagine it's in the driver/module config somewhere. Thanks, Stefano ___ --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] Toll-free dialing via PRI problem
Outgoing calls to certain toll-fee (8XX) numbers fail -- we hear ringing but the calls are never answered. All other calls, and most toll-free numbers are not affected. The numbers that are affected are all travel related companies (United Airlines, American Airlines, US Air, Starwood Hotels, etc.) we cannot connect to any of these numbers. Hey Tim, All I can offer you is the fact that I see the exact same thing on my setup that uses * and a TE411P. I've also seen it when calling Lenovo tech support and Sirius Satellite Radio. On the latter two, it bypasses the auto-attendant when I call and connects me straight to an operator/technician. When you call on regular PBX or cell phone, you are greeted by an auto-attendant, press 1, yada-yada. Let us know what you find out. Cheers, Stefano ___ --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] Toll-free dialing via PRI problem
This is a common issue with large inbound call center operations. They like to cheat. They actually start sending prompts to the caller without actually signalling their carrier that they have answered the line. Typically they do not answer until a phone is ringing or you are in a queue. I do believe this is illegal per the FCC. From asterisk, you do not hear anything other than ringing as it does not cut the audio path through until it receives the answer from the far end, hence the steady ringing. Forgive my impetuousness, but what's the differentiator that allows calls to work from my legacy PBX and cell phone that precludes the Asterisk from working as the others? Just askin' is all... ;-) Stefano ___ --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] Erratic Snom MWI lights
Snom's ... Retrieve button... works when MWI is *NOT* lit but does *NOT* work when it is lit. Any advice Do you have an asterisk extension in your dialplan? See http://www.voip-info.org/wiki-Asterisk+phone+snom, especially the part about: Making the MWI work with ASTERISK Asterisk sends notifications on voicemail messages (if you configured the mailbox option in sip.conf. The messages are sent by default from [EMAIL PROTECTED], which can be modified using vmexten= in sip.conf. When pressing the MWI or Vmail soft button on the SNOM phones the phone calls this extension to connect to the voicemail application. If you haven't configured an extension named asterisk in the context of the phone, the MWI/VMail button will not work. Cheers, Stefano ___ --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] Erratic Snom MWI lights
exten = asterisk,1,VoicemailMain(${CALLERIDNUM})
I use that one myself. Does the Snom attempt to dial asterisk when
you hit Retrieve? What error do you get? Sure it's in the right
context (I screw that up ALL the time)...
Stefano
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[Asterisk-Users] Experience with SetTransferCapability
Does anyone have exporience with the SetTransferCapability application? I'm trying to use it, but it does not give the expected result. My configuration is like this: Telco---Definity---Asterisk---Brooktrout PRI card The Definity communicates with the Asterisk using the Bearer 3.1K audio setting. This is because the Asterisk was placed in between an already working setup. In default situations, Calls from SIP --- Asterisk --- Brooktrout seem to work OK. Calls from Definity -- Asterisk --- Brooktrout do not. All I get is silence and it seems as though (looking at pri debug span 2), the call terminates immediately with the following output: **BEGUN DEBUG OUTPUT** asterisk*CLI -- Making new call for cr 32819 Protocol Discriminator: Q.931 (8) len=46 Call Ref: len= 2 (reference 51/0x33) (Originator) Message type: SETUP (5) [04 02 88 90] Bearer Capability (len= 4) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Unrestricted digital information (8) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 0 User information layer 1: Unknown (24) [18 03 a1 83 82] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Preferred Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 2 ] [28 10 b1 4d 63 47 68 65 65 2c 20 53 74 65 66 61 6e 6f] Display (len=16) Charset: 31 [ McGhee, Stefano ] [6c 05 21 83 32 35 37] Calling Number (len= 7) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation allowed of network provided number (3) '257' ] [70 05 80 35 31 33 33] Called Number (len= 7) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) '5133' ] Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 51/0x33) (Terminator) Message type: CALL PROCEEDING (2) [18 03 a9 83 82] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 2 ] -- Processing IE 24 (cs0, Channel Identification) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 51/0x33) (Terminator) Message type: DISCONNECT (69) [08 02 80 d8] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0) Ext: 1 Cause: Unknown (88), class = Invalid message (5) ] -- Processing IE 8 (cs0, Cause) NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication, peerstate Disconnect Request Protocol Discriminator: Q.931 (8) len=16 Call Ref: len= 2 (reference 51/0x33) (Originator) Message type: RELEASE (77) [08 02 81 d8] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Unknown (88), class = Invalid message (5) ] [7e 05 04 d8 1d 15 08] User-User Information (len= 7) [ 04 58 1d 15 08 ] Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 51/0x33) (Terminator) Message type: RELEASE COMPLETE (90) NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null asterisk*CLI * END DEBUG OUTPUT Now, changing on calls to be routed to the Brooktrout to be explicitly SetTransferCapability(SPEECH) works fine. However, the default value is SPEECH and the debug logs bear that out. NOT setting the value doesn't work. Setting SetTransferCapability to DIGITAL and 3K1AUDIO seem to have the same effect as not setting SetTransferCapability at all. Oddly, the Info transfer capability I get for DIGITAL and 3K1AUDIO are the same: Unrestricted digital information (8). Anyone have an idea what I might be missing? Stefano McGhee Manager of Information Systems StudentUniverse.com 100 Talcott Avenue East Watertown, MA, 02472 Email: [EMAIL PROTECTED] Tel: 617.321.3257 StudentUniverse.com Students Fly Cheaper ___ --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
