<html><div style='background-color:'><DIV class=RTE>
<P>Yeh maybe I sold my sole to the devil a few times... but Im not telemarketing. Im calling back coustomers Ive hade in the past and doing polls in 1 business, and the other is student lone consolidation.<BR><BR></P></DIV>
<DIV></DIV>>From: asterisk-users-request@lists.digium.com
<DIV></DIV>>Reply-To: asterisk-users@lists.digium.com
<DIV></DIV>>To: asterisk-users@lists.digium.com
<DIV></DIV>>Subject: Asterisk-Users digest, Vol 1 #2423 - 14 msgs
<DIV></DIV>>Date: Fri, 09 Jan 2004 16:06:30 -0600
<DIV></DIV>>
<DIV></DIV>>Send Asterisk-Users mailing list submissions to
<DIV></DIV>> asterisk-users@lists.digium.com
<DIV></DIV>>
<DIV></DIV>>To subscribe or unsubscribe via the World Wide Web, visit
<DIV></DIV>> http://lists.digium.com/mailman/listinfo/asterisk-users
<DIV></DIV>>or, via email, send a message with subject or body 'help' to
<DIV></DIV>> asterisk-users-request@lists.digium.com
<DIV></DIV>>
<DIV></DIV>>You can reach the person managing the list at
<DIV></DIV>> asterisk-users-admin@lists.digium.com
<DIV></DIV>>
<DIV></DIV>>When replying, please edit your Subject line so it is more specific
<DIV></DIV>>than "Re: Contents of Asterisk-Users digest..."
<DIV></DIV>>
<DIV></DIV>>
<DIV></DIV>>Today's Topics:
<DIV></DIV>>
<DIV></DIV>> 1. Re: Asterisk-Users digest, Vol 1 #2413 - 13 msgs (dkwok)
<DIV></DIV>> 2. Re: USA dial plan (Tilghman Lesher)
<DIV></DIV>> 3. RE: USA dial plan (Kris Boutilier)
<DIV></DIV>> 4. Re: Why * try to codec translate when it can do
<DIV></DIV>> without during codec negotiation. (Robert Hajime Lanning)
<DIV></DIV>> 5. RE: Cisco Gear (Steven Critchfield)
<DIV></DIV>> 6. Re: USA dial plan (info-lists@robertc.de)
<DIV></DIV>> 7. Re: * as sip b2bua? (Olle E. Johansson)
<DIV></DIV>> 8. RE: Mailing list growth (daryl@introspect.net)
<DIV></DIV>> 9. Re: DTMF in MeetMe (David Burr)
<DIV></DIV>> 10. RE: Screen Pop & Remote Agents = Telemarketing (daryl@introspect.net)
<DIV></DIV>> 11. Re: Cisco Gear (Steve)
<DIV></DIV>> 12. Re: SIP and error talking to voicemail (Steve)
<DIV></DIV>> 13. Re: SIP and error talking to voicemail (Steve)
<DIV></DIV>>
<DIV></DIV>>--__--__--
<DIV></DIV>>
<DIV></DIV>>Message: 1
<DIV></DIV>>Date: Sat, 10 Jan 2004 07:48:27 +0100
<DIV></DIV>>From: dkwok <DKWOK@IWARE.COM.AU>
<DIV></DIV>>Organization: iware.com.au
<DIV></DIV>>To: asterisk-users@lists.digium.com, terence@parker.com.hk
<DIV></DIV>>Subject: [Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #2413 - 13 msgs
<DIV></DIV>>Reply-To: asterisk-users@lists.digium.com
<DIV></DIV>>
<DIV></DIV>>This is a cryptographically signed message in MIME format.
<DIV></DIV>>
<DIV></DIV>>--------------ms030003010805000002020003
<DIV></DIV>>Content-Type: text/plain; charset=us-ascii; format=flowed
<DIV></DIV>>Content-Transfer-Encoding: 7bit
<DIV></DIV>>
<DIV></DIV>> >
<DIV></DIV>> > -- __--__--
<DIV></DIV>> >
<DIV></DIV>> > Message: 1
<DIV></DIV>> > From: Terence Parker <TERENCE@PARKER.COM.HK>
<DIV></DIV>> > Date: Fri, 9 Jan 2004 11:25:23 +0800
<DIV></DIV>> > To: asterisk-users@lists.digium.com
<DIV></DIV>> > Subject: [Asterisk-Users] Problem registering FWD
<DIV></DIV>> > Reply-To: asterisk-users@lists.digium.com
<DIV></DIV>> >
<DIV></DIV>> >
<DIV></DIV>> > --Apple-Mail-1-822243116
<DIV></DIV>> > Content-Transfer-Encoding: 7bit
<DIV></DIV>> > Content-Type: text/plain;
<DIV></DIV>> > charset=US-ASCII;
<DIV></DIV>> > format=flowed
<DIV></DIV>> >
<DIV></DIV>> > I seem to have a problem registering my Asterisk box with the FWD
<DIV></DIV>> > service - I have the following in my sip.conf file:
<DIV></DIV>> >
<DIV></DIV>>
<DIV></DIV>>Have a look at http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions
<DIV></DIV>>
<DIV></DIV>>If you sip client is behind firewall you will not be able to connect to
<DIV></DIV>>FWD. However you can get around by using IAXTEL. check out this page:
<DIV></DIV>>
<DIV></DIV>>www.iaxtel.com/setup.html
<DIV></DIV>>
<DIV></DIV>>David Kwok
<DIV></DIV>>
<DIV></DIV>>--------------ms030003010805000002020003
<DIV></DIV>>Content-Type: application/x-pkcs7-signature; name="smime.p7s"
<DIV></DIV>>Content-Transfer-Encoding: base64
<DIV></DIV>>Content-Disposition: attachment; filename="smime.p7s"
<DIV></DIV>>Content-Description: S/MIME Cryptographic Signature
<DIV></DIV>>
<DIV></DIV>>MIAGCSqGSIb3DQEHAqCAMIACAQExCzAJBgUrDgMCGgUAMIAGCSqGSIb3DQEHAQAAoIIEvjCC
<DIV></DIV>>AlswggHEoAMCAQICARQwDQYJKoZIhvcNAQEEBQAwRDELMAkGA1UEBhMCQVUxDjAMBgNVBAoT
<DIV></DIV>>BWl3YXJlMQswCQYDVQQLEwJDQTEYMBYGA1UEAxQPY2FAaXdhcmUuY29tLmF1MB4XDTAzMTEw
<DIV></DIV>>MTAwMTIyNFoXDTA0MTAzMTAwMTIyNFowfTELMAkGA1UEBhMCQVUxDDAKBgNVBAgTA05TVzEQ
<DIV></DIV>>MA4GA1UEBxMHTlNZRE5FWTEOMAwGA1UEChMFSVdBUkUxGzAZBgNVBAMUEmRrd29rQGl3YXJl
<DIV></DIV>>LmNvbS5hdTEhMB8GCSqGSIb3DQEJARYSZGt3b2tAaXdhcmUuY29tLmF1MIGfMA0GCSqGSIb3
<DIV></DIV>>DQEBAQUAA4GNADCBiQKBgQDCcZEUZbESmEA4zQyeVp+t3Q/PU7Mi0tqOnu2BTBWJZ0sv1aRY
<DIV></DIV>>bEn1q67fxkQ4Q/x0OWyKv7p7tTZNKF2oSp1TRInCmSleyGOfKm7AR3OSNhYUfGF08vefcl3X
<DIV></DIV>>G2Y1nMoXDZUGfas7AbLmKkMgBx0jQ9VKbKzG70ganHgREchvrwIDAQABoyQwIjAgBgNVHREB
<DIV></DIV>>Af8EFjAUghJka3dva0Bpd2FyZS5jb20uYXUwDQYJKoZIhvcNAQEEBQADgYEAbGa9xUwYlpja
<DIV></DIV>>wMGh/L46YhyolmOqJa4a72sVu1wBVRnVLTXVn7Wc4p7SZaKjTdhOmRS7SmKvm3cPx3u9XKCY
<DIV></DIV>>6nuUrkA9SMAtYuJ9UzE+BMV8/MtC1avEtTZebWjdXy4f8dKc4AVN+WP9YAFGh67a1GmTk6M8
<DIV></DIV>>Ilzm/giua4G18qgwggJbMIIBxKADAgECAgEUMA0GCSqGSIb3DQEBBAUAMEQxCzAJBgNVBAYT
<DIV></DIV>>AkFVMQ4wDAYDVQQKEwVpd2FyZTELMAkGA1UECxMCQ0ExGDAWBgNVBAMUD2NhQGl3YXJlLmNv
<DIV></DIV>>bS5hdTAeFw0wMzExMDEwMDEyMjRaFw0wNDEwMzEwMDEyMjRaMH0xCzAJBgNVBAYTAkFVMQww
<DIV></DIV>>CgYDVQQIEwNOU1cxEDAOBgNVBAcTB05TWURORVkxDjAMBgNVBAoTBUlXQVJFMRswGQYDVQQD
<DIV></DIV>>FBJka3dva0Bpd2FyZS5jb20uYXUxITAfBgkqhkiG9w0BCQEWEmRrd29rQGl3YXJlLmNvbS5h
<DIV></DIV>>dTCBnzANBgkqhkiG9w0BAQEFAAOBjQAwgYkCgYEAwnGRFGWxEphAOM0Mnlafrd0Pz1OzItLa
<DIV></DIV>>jp7tgUwViWdLL9WkWGxJ9auu38ZEOEP8dDlsir+6e7U2TShdqEqdU0SJwpkpXshjnypuwEdz
<DIV></DIV>>kjYWFHxhdPL3n3Jd1xtmNZzKFw2VBn2rOwGy5ipDIAcdI0PVSmysxu9IGpx4ERHIb68CAwEA
<DIV></DIV>>AaMkMCIwIAYDVR0RAQH/BBYwFIISZGt3b2tAaXdhcmUuY29tLmF1MA0GCSqGSIb3DQEBBAUA
<DIV></DIV>>A4GBAGxmvcVMGJaY2sDBofy+OmIcqJZjqiWuGu9rFbtcAVUZ1S011Z+1nOKe0mWio03YTpkU
<DIV></DIV>>u0pir5t3D8d7vVygmOp7lK5APUjALWLifVMxPgTFfPzLQtWrxLU2Xm1o3V8uH/HSnOAFTflj
<DIV></DIV>>/WABRoeu2tRpk5OjPCJc5v4IrmuBtfKoMYICWjCCAlYCAQEwSTBEMQswCQYDVQQGEwJBVTEO
<DIV></DIV>>MAwGA1UEChMFaXdhcmUxCzAJBgNVBAsTAkNBMRgwFgYDVQQDFA9jYUBpd2FyZS5jb20uYXUC
<DIV></DIV>>ARQwCQYFKw4DAhoFAKCCAWcwGAYJKoZIhvcNAQkDMQsGCSqGSIb3DQEHATAcBgkqhkiG9w0B
<DIV></DIV>>CQUxDxcNMDQwMTEwMDY0ODI4WjAjBgkqhkiG9w0BCQQxFgQUHoTFM+i9DmiJhNlFgQT5JN5k
<DIV></DIV>>QcgwUgYJKoZIhvcNAQkPMUUwQzAKBggqhkiG9w0DBzAOBggqhkiG9w0DAgICAIAwDQYIKoZI
<DIV></DIV>>hvcNAwICAUAwBwYFKw4DAgcwDQYIKoZIhvcNAwICASgwWAYJKwYBBAGCNxAEMUswSTBEMQsw
<DIV></DIV>>CQYDVQQGEwJBVTEOMAwGA1UEChMFaXdhcmUxCzAJBgNVBAsTAkNBMRgwFgYDVQQDFA9jYUBp
<DIV></DIV>>d2FyZS5jb20uYXUCARQwWgYLKoZIhvcNAQkQAgsxS6BJMEQxCzAJBgNVBAYTAkFVMQ4wDAYD
<DIV></DIV>>VQQKEwVpd2FyZTELMAkGA1UECxMCQ0ExGDAWBgNVBAMUD2NhQGl3YXJlLmNvbS5hdQIBFDAN
<DIV></DIV>>BgkqhkiG9w0BAQEFAASBgIcgZFFpQGww1yZvz1boal6wMRMIkmrRi5Zr2tQn/k6sYuVf0eRS
<DIV></DIV>>RNBQBVtxmNkNuPJMkg2iEyRGCDd5K0r1Nak7whLAjeGTILW70wLnzHIRsdfF5slpg153ihwx
<DIV></DIV>>fGKGT+z/BY872ljrNjXrPi/a0cpaw66a3xxB22vaoT1OvWtxAAAAAAAA
<DIV></DIV>>--------------ms030003010805000002020003--
<DIV></DIV>>
<DIV></DIV>>
<DIV></DIV>>--__--__--
<DIV></DIV>>
<DIV></DIV>>Message: 2
<DIV></DIV>>From: Tilghman Lesher <TILGHMAN@MAIL.JEFFANDTILGHMAN.COM>
<DIV></DIV>>To: asterisk-users@lists.digium.com
<DIV></DIV>>Subject: Re: [Asterisk-Users] USA dial plan
<DIV></DIV>>Date: Fri, 9 Jan 2004 15:10:55 -0600
<DIV></DIV>>Reply-To: asterisk-users@lists.digium.com
<DIV></DIV>>
<DIV></DIV>>On Friday 09 January 2004 13:55, ml@neoninternet.com wrote:
<DIV></DIV>> > > Hi,
<DIV></DIV>> > >
<DIV></DIV>> > > Do the callers in USA dialling from USA Telco lines always have
<DIV></DIV>> > > to prefix the CITY/AREA code with "1" in order
<DIV></DIV>> > > To successfully make a call to other USA destinations?
<DIV></DIV>> > >
<DIV></DIV>> > > ----
<DIV></DIV>> > > I have not been to USA (yet) :)
<DIV></DIV>> > >
<DIV></DIV>> > > Ta
<DIV></DIV>> > > SJ
<DIV></DIV>> >
<DIV></DIV>> > In all cases of long distance, 1 plus the area code is used. In
<DIV></DIV>> > small areas where local only is involved you usually only dial 7
<DIV></DIV>> > digits. In metro areas with multiple area codes, you use 10 digit
<DIV></DIV>> > dialing. Some places you use 10 digit dialing or 1 + area code,
<DIV></DIV>> > depends on the phone company. I've seen this happen on the east
<DIV></DIV>> > coast.
<DIV></DIV>>
<DIV></DIV>>And then you have numbers that you cannot dial, because your local
<DIV></DIV>>provider forces you to dial the 1, but the remote provider refuses to
<DIV></DIV>>complete the call with the 1.
<DIV></DIV>>
<DIV></DIV>>Aren't multiple providers wonderful?
<DIV></DIV>>
<DIV></DIV>>-Tilghman
<DIV></DIV>>
<DIV></DIV>>
<DIV></DIV>>--__--__--
<DIV></DIV>>
<DIV></DIV>>Message: 3
<DIV></DIV>>From: Kris Boutilier <KRIS.BOUTILIER@SCRD.BC.CA>
<DIV></DIV>>To: "'asterisk-users@lists.digium.com'" <ASTERISK-USERS@LISTS.DIGIUM.COM>
<DIV></DIV>>Subject: RE: [Asterisk-Users] USA dial plan
<DIV></DIV>>Date: Fri, 9 Jan 2004 13:11:57 -0800
<DIV></DIV>>Reply-To: asterisk-users@lists.digium.com
<DIV></DIV>>
<DIV></DIV>> Information on the way things are structured here can be gleaned by
<DIV></DIV>>Googling for 'North American Numbering Plan'. Way too much information can
<DIV></DIV>>be found at http://www.nanpa.com/
<DIV></DIV>>
<DIV></DIV>>k.
<DIV></DIV>>
<DIV></DIV>>-----Original Message-----
<DIV></DIV>>From: Senad Jordanovic [mailto:senad@boltblue.com]
<DIV></DIV>>Sent: January 9, 2004 10:50 AM
<DIV></DIV>>To: asterisk-users@lists.digium.com
<DIV></DIV>>Subject: [Asterisk-Users] USA dial plan
<DIV></DIV>>
<DIV></DIV>>
<DIV></DIV>>Hi,
<DIV></DIV>>
<DIV></DIV>>Do the callers in USA dialling from USA Telco lines always have to
<DIV></DIV>>prefix the CITY/AREA code with "1" in order
<DIV></DIV>>To successfully make a call to other USA destinations?
<DIV></DIV>>
<DIV></DIV>>----
<DIV></DIV>>I have not been to USA (yet) :)
<DIV></DIV>>
<DIV></DIV>>Ta
<DIV></DIV>>SJ
<DIV></DIV>>
<DIV></DIV>>
<DIV></DIV>>_______________________________________________
<DIV></DIV>>Asterisk-Users mailing list
<DIV></DIV>>Asterisk-Users@lists.digium.com
<DIV></DIV>>http://lists.digium.com/mailman/listinfo/asterisk-users
<DIV></DIV>>To UNSUBSCRIBE or update options visit:
<DIV></DIV>> http://lists.digium.com/mailman/listinfo/asterisk-users
<DIV></DIV>>
<DIV></DIV>>--__--__--
<DIV></DIV>>
<DIV></DIV>>Message: 4
<DIV></DIV>>Date: Fri, 9 Jan 2004 13:20:15 -0800 (PST)
<DIV></DIV>>Subject: Re: [Asterisk-Users] Why * try to codec translate when it can do
<DIV></DIV>> without during codec negotiation.
<DIV></DIV>>From: "Robert Hajime Lanning" <LANNING+ASTERISK@MONSOONWIND.COM>
<DIV></DIV>>To: asterisk-users@lists.digium.com
<DIV></DIV>>Reply-To: asterisk-users@lists.digium.com
<DIV></DIV>>
<DIV></DIV>><QUOTE who="SamW">
<DIV></DIV>> > case 1
<DIV></DIV>> > ------
<DIV></DIV>> > [sip-a]
<DIV></DIV>> > allow=g729
<DIV></DIV>> > disallow=all
<DIV></DIV>> > allow=alaw
<DIV></DIV>>
<DIV></DIV>>Try:
<DIV></DIV>>[sip-a]
<DIV></DIV>>disallow=all
<DIV></DIV>>allow=g729
<DIV></DIV>>allow=alaw
<DIV></DIV>>
<DIV></DIV>>The "disallow=all" clears your previous setting of "allow=g729"
<DIV></DIV>>
<DIV></DIV>>--
<DIV></DIV>>END OF LINE
<DIV></DIV>> -MCP
<DIV></DIV>>
<DIV></DIV>>--__--__--
<DIV></DIV>>
<DIV></DIV>>Message: 5
<DIV></DIV>>Subject: RE: [Asterisk-Users] Cisco Gear
<DIV></DIV>>From: Steven Critchfield <CRITCH@BASESYS.COM>
<DIV></DIV>>To: asterisk-users@lists.digium.com
<DIV></DIV>>Date: Fri, 09 Jan 2004 15:23:36 -0600
<DIV></DIV>>Reply-To: asterisk-users@lists.digium.com
<DIV></DIV>>
<DIV></DIV>>On Fri, 2004-01-09 at 14:20, Arnold Ligtvoet wrote:
<DIV></DIV>> > message posted on behalf Of Adthrawn
<DIV></DIV>> > [SNIP cisco stuff]
<DIV></DIV>> > > I'll now feel ashamed, and sink into my seat :-)
<DIV></DIV>> > >
<DIV></DIV>> > > Best,
<DIV></DIV>> > > Ad.
<DIV></DIV>> >
<DIV></DIV>> > Perhaps it would have been better to provide an email address or phonenumber
<DIV></DIV>> > where people can contact you directly. Now everybody who is interested has
<DIV></DIV>> > to reply to the list.
<DIV></DIV>>
<DIV></DIV>>Maybe you should learn how to use your email client. I emailed this
<DIV></DIV>>person at the email address used for the original message. I have
<DIV></DIV>>already exchanged several messages with him. So I can vouch for the fact
<DIV></DIV>>that the email address is good and checked.
<DIV></DIV>>--
<DIV></DIV>>Steven Critchfield <CRITCH@BASESYS.COM>
<DIV></DIV>>
<DIV></DIV>>
<DIV></DIV>>--__--__--
<DIV></DIV>>
<DIV></DIV>>Message: 6
<DIV></DIV>>Date: Fri, 9 Jan 2004 22:24:55 +0100 (CET)
<DIV></DIV>>Subject: Re: [Asterisk-Users] USA dial plan
<DIV></DIV>>From: info-lists@robertc.de
<DIV></DIV>>To: asterisk-users@lists.digium.com
<DIV></DIV>>Reply-To: asterisk-users@lists.digium.com
<DIV></DIV>>
<DIV></DIV>> > Hi,
<DIV></DIV>> >
<DIV></DIV>> > Do the callers in USA dialling from USA Telco lines always have to
<DIV></DIV>> > prefix the CITY/AREA code with "1" in order
<DIV></DIV>> > To successfully make a call to other USA destinations?
<DIV></DIV>> >
<DIV></DIV>> > ----
<DIV></DIV>> > I have not been to USA (yet) :)
<DIV></DIV>> >
<DIV></DIV>> > Ta
<DIV></DIV>> > SJ
<DIV></DIV>>
<DIV></DIV>>For comprehensive info by area code (and as pointed out it does differ
<DIV></DIV>>from location to location) check the North American Numbering Plan website
<DIV></DIV>>at http://www.nanpa.com/. Left menu click on Dialing Plan and then go to
<DIV></DIV>>the location of your choice.
<DIV></DIV>>
<DIV></DIV>>Robert
<DIV></DIV>>
<DIV></DIV>>--__--__--
<DIV></DIV>>
<DIV></DIV>>Message: 7
<DIV></DIV>>Date: Fri, 09 Jan 2004 22:28:17 +0100
<DIV></DIV>>From: "Olle E. Johansson" <OEJ@EDVINA.NET>
<DIV></DIV>>Organization: Edvina AB
<DIV></DIV>>To: asterisk-users@lists.digium.com
<DIV></DIV>>Subject: Re: [Asterisk-Users] * as sip b2bua?
<DIV></DIV>>Reply-To: asterisk-users@lists.digium.com
<DIV></DIV>>
<DIV></DIV>>Thilo Salmon wrote:
<DIV></DIV>>
<DIV></DIV>> > Hi everyone,
<DIV></DIV>> >
<DIV></DIV>> > any chance * could be used as a b2bua without forcing the media stream
<DIV></DIV>> > through the same box? I would love to do some computing on incoming
<DIV></DIV>> > calls, do things like setting another callerid and the forward the call
<DIV></DIV>> > to another sip UA - all without any audio traversing the * box. Any
<DIV></DIV>> > ideas?
<DIV></DIV>>Thilo,
<DIV></DIV>>Isn't the definition of a b2bua that the media streams pass it?
<DIV></DIV>>back-to-back-user-agent.
<DIV></DIV>>
<DIV></DIV>>Anyway, not to be picky, Asterisk by default wants to be in the media
<DIV></DIV>>path. There are ways to release the signalling and media path
<DIV></DIV>>back to the clients, with canreinvite=yes, but that's not the default behaviour.
<DIV></DIV>>
<DIV></DIV>>A SIP proxy like SIP express router from iptel.org fits your
<DIV></DIV>>description better. And yes, SER works together with Asterisk.
<DIV></DIV>>
<DIV></DIV>>
<DIV></DIV>>/Olle
<DIV></DIV>>
<DIV></DIV>>
<DIV></DIV>>--__--__--
<DIV></DIV>>
<DIV></DIV>>Message: 8
<DIV></DIV>>Subject: RE: [Asterisk-Users] Mailing list growth
<DIV></DIV>>Date: Fri, 9 Jan 2004 16:29:42 -0500
<DIV></DIV>>From: <DARYL@INTROSPECT.NET>
<DIV></DIV>>To: <ASTERISK-USERS@LISTS.DIGIUM.COM>
<DIV></DIV>>Reply-To: asterisk-users@lists.digium.com
<DIV></DIV>>
<DIV></DIV>> > -----Original Message-----
<DIV></DIV>> > From: asterisk-users-admin@lists.digium.com=20
<DIV></DIV>> > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of=20
<DIV></DIV>> > Philipp von Klitzing
<DIV></DIV>> > Sent: Friday, January 09, 2004 6:52 AM
<DIV></DIV>> > To: asterisk-users@lists.digium.com
<DIV></DIV>> > Subject: Re: [Asterisk-Users] Mailing list growth
<DIV></DIV>> >=20
<DIV></DIV>>[...]
<DIV></DIV>> > "higher-level implementation" list that deals specifically with=20
<DIV></DIV>> > channelbanks & T1 issues (=3Dlarger installations). VoIP will remain =
<DIV></DIV>>on=20
<DIV></DIV>> > asterisk-users.
<DIV></DIV>>[...]
<DIV></DIV>>
<DIV></DIV>>That doesn't quite sound right. Maybe it is from your perspective, but
<DIV></DIV>>are you telling me that the NOCs with 80+ 7960's running VoIP don't
<DIV></DIV>>count as a large installation?
<DIV></DIV>>
<DIV></DIV>>Of course, the term large is also relative. A 4-port T1 card on its
<DIV></DIV>>own....even 2 or 3 of them, could never by any stretch of my imagination
<DIV></DIV>>be considered a large installation......but I deal with (among other
<DIV></DIV>>things) Definity's that service near entire buildings in mid-town
<DIV></DIV>>Manhattan with multiple DS3s....so it's all relative.
<DIV></DIV>>
<DIV></DIV>>The problem with splitting VoIP and T1/TDM/whatever you want to call it
<DIV></DIV>>is that the crossover is huge, and where the problems lie often aren't
<DIV></DIV>>clear to those looking for help.
<DIV></DIV>>Daryl G. Jurbala
<DIV></DIV>>BMPC Network Operations
<DIV></DIV>>Tel: +1 215 825 8401 x235
<DIV></DIV>>Fax: +1 508 526 8500
<DIV></DIV>>INOC-DBA: 26412*DGJ
<DIV></DIV>>
<DIV></DIV>>PGP Key: http://www.introspect.net/pgp=20
<DIV></DIV>>
<DIV></DIV>>--__--__--
<DIV></DIV>>
<DIV></DIV>>Message: 9
<DIV></DIV>>Date: Fri, 09 Jan 2004 14:33:09 -0700
<DIV></DIV>>From: David Burr <LIST@T1.BZ>
<DIV></DIV>>To: asterisk-users@lists.digium.com
<DIV></DIV>>Subject: Re: [Asterisk-Users] DTMF in MeetMe
<DIV></DIV>>Reply-To: asterisk-users@lists.digium.com
<DIV></DIV>>
<DIV></DIV>>the * and # are hard coded.
<DIV></DIV>>unless "b' -- run AGI script specified in ${MEETME_AGI_BACKGROUND}
<DIV></DIV>>is what your refering to.. which doesnt say how to use it.
<DIV></DIV>>
<DIV></DIV>>
<DIV></DIV>>Jeremy McNamara wrote:
<DIV></DIV>>
<DIV></DIV>> > David Burr wrote:
<DIV></DIV>> >
<DIV></DIV>> >> Does the MeetMe monitor for DTMF tones to trigger an AGI?
<DIV></DIV>> >> If not is this a planned feature?
<DIV></DIV>> >
<DIV></DIV>> >
<DIV></DIV>> >
<DIV></DIV>> > show application MeetMe
<DIV></DIV>> >
<DIV></DIV>> >
<DIV></DIV>> > Jeremy McNamara
<DIV></DIV>> >
<DIV></DIV>> >
<DIV></DIV>> > _______________________________________________
<DIV></DIV>> > Asterisk-Users mailing list
<DIV></DIV>> > Asterisk-Users@lists.digium.com
<DIV></DIV>> > http://lists.digium.com/mailman/listinfo/asterisk-users
<DIV></DIV>> > To UNSUBSCRIBE or update options visit:
<DIV></DIV>> > http://lists.digium.com/mailman/listinfo/asterisk-users
<DIV></DIV>> >
<DIV></DIV>> >
<DIV></DIV>>
<DIV></DIV>>
<DIV></DIV>>--__--__--
<DIV></DIV>>
<DIV></DIV>>Message: 10
<DIV></DIV>>Subject: RE: [Asterisk-Users] Screen Pop & Remote Agents = Telemarketing
<DIV></DIV>>Date: Fri, 9 Jan 2004 16:38:52 -0500
<DIV></DIV>>From: <DARYL@INTROSPECT.NET>
<DIV></DIV>>To: <ASTERISK-USERS@LISTS.DIGIUM.COM>
<DIV></DIV>>Reply-To: asterisk-users@lists.digium.com
<DIV></DIV>>
<DIV></DIV>>-----Original Message-----
<DIV></DIV>>From: asterisk-users-admin@lists.digium.com
<DIV></DIV>>[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of empire
<DIV></DIV>>underground
<DIV></DIV>>Sent: Friday, January 09, 2004 1:32 PM
<DIV></DIV>>To: asterisk-users@lists.digium.com
<DIV></DIV>>Subject: [Asterisk-Users] Screen Pop & Remote Agents
<DIV></DIV>> > can I put a .csv file in the sql DB and have it dial from there? and
<DIV></DIV>>will I be able to set a
<DIV></DIV>> > Dial Plan to only call certin area codes? stuff like that. The reason
<DIV></DIV>>I ask all this is because
<DIV></DIV>> > all of these over priced dialers do just that. Also can Asterisk be
<DIV></DIV>>set with the FTC laws to 3%
<DIV></DIV>> > droped call ratio?
<DIV></DIV>> > If all of the questions I have asked here have allready be answered
<DIV></DIV>>some point in time... Can
<DIV></DIV>> > someone pl ease point me in the right direction to get all the
<DIV></DIV>>answers.
<DIV></DIV>>
<DIV></DIV>>So you're setting up a telemarketing rig? That's certainly not the kind
<DIV></DIV>>of thing I'd expect to get much help or sympathy for ANYWHERE other than
<DIV></DIV>>in telemarketing circles.
<DIV></DIV>>
<DIV></DIV>>I think the cost of a proper commercial predictive dialer would be
<DIV></DIV>>relatively cheap after already having sold your soul.
<DIV></DIV>>
<DIV></DIV>>Daryl
<DIV></DIV>>
<DIV></DIV>>--__--__--
<DIV></DIV>>
<DIV></DIV>>Message: 11
<DIV></DIV>>From: Steve <STEVE@SZMIDT.ORG>
<DIV></DIV>>To: asterisk-users@lists.digium.com
<DIV></DIV>>Subject: Re: [Asterisk-Users] Cisco Gear
<DIV></DIV>>Date: Fri, 9 Jan 2004 16:41:20 -0500
<DIV></DIV>>Reply-To: asterisk-users@lists.digium.com
<DIV></DIV>>
<DIV></DIV>>On Friday 09 January 2004 02:40 pm, Iain Stevenson wrote:
<DIV></DIV>> > Prices? Are we talking a 7960 for the price of a SNOM?
<DIV></DIV>> >
<DIV></DIV>> > Iain
<DIV></DIV>>
<DIV></DIV>>Oops, just realized I replied to the wrong person....
<DIV></DIV>>
<DIV></DIV>> > --On Friday, January 9, 2004 6:00 pm +0000 Adthrawn
<DIV></DIV>> >
<DIV></DIV>> > <ADTHRAWN@ADTHRAWN.FREESERVE.CO.UK>wrote:
<DIV></DIV>> > > Hi,
<DIV></DIV>> > >
<DIV></DIV>> > > I know it's not really the place, but if anybody in the UK (or US) is
<DIV></DIV>> > > interested, I'm clearing out lots of new Cisco stock...
<DIV></DIV>> > >
<DIV></DIV>> > > 7970G's (colour LCD), 7960G's, 7940G's, 7920G's (wireless IP phone),
<DIV></DIV>> > > 7935's (conference phone) and 3550-24-PWR switches.
<DIV></DIV>> > >
<DIV></DIV>> > > I also have boxes of 7914's, the single-7914 foot stand and double-7914
<DIV></DIV>> > > foot stand (these are required to connect a 7914 to a 7960G).
<DIV></DIV>> > >
<DIV></DIV>> > > And some useful locking and non-locking wallmount brackets for 79xx
<DIV></DIV>> > > range.
<DIV></DIV>> > >
<DIV></DIV>> > > We also have lots of PSU's for the whole 79xx range.
<DIV></DIV>> > >
<DIV></DIV>> > > I'll now feel ashamed, and sink into my seat :-)
<DIV></DIV>> > >
<DIV></DIV>> > > Best,
<DIV></DIV>> > > Ad.
<DIV></DIV>> > >
<DIV></DIV>> > > _______________________________________________
<DIV></DIV>> > > Asterisk-Users mailing list
<DIV></DIV>> > > Asterisk-Users@lists.digium.com
<DIV></DIV>> > > http://lists.digium.com/mailman/listinfo/asterisk-users
<DIV></DIV>> > > To UNSUBSCRIBE or update options visit:
<DIV></DIV>> > > http://lists.digium.com/mailman/listinfo/asterisk-users
<DIV></DIV>> >
<DIV></DIV>> > _______________________________________________
<DIV></DIV>> > Asterisk-Users mailing list
<DIV></DIV>> > Asterisk-Users@lists.digium.com
<DIV></DIV>> > http://lists.digium.com/mailman/listinfo/asterisk-users
<DIV></DIV>> > To UNSUBSCRIBE or update options visit:
<DIV></DIV>> > http://lists.digium.com/mailman/listinfo/asterisk-users
<DIV></DIV>>
<DIV></DIV>>--
<DIV></DIV>>Steve
<DIV></DIV>>
<DIV></DIV>>__________________________________________________
<DIV></DIV>>You actually need to constantly be alert
<DIV></DIV>> and willing to handle things, or life
<DIV></DIV>> will find a way to get you good!
<DIV></DIV>>
<DIV></DIV>>--__--__--
<DIV></DIV>>
<DIV></DIV>>Message: 12
<DIV></DIV>>From: Steve <STEVE@SZMIDT.ORG>
<DIV></DIV>>To: asterisk-users@lists.digium.com
<DIV></DIV>>Subject: Re: [Asterisk-Users] SIP and error talking to voicemail
<DIV></DIV>>Date: Fri, 9 Jan 2004 16:47:14 -0500
<DIV></DIV>>Reply-To: asterisk-users@lists.digium.com
<DIV></DIV>>
<DIV></DIV>>On Thursday 08 January 2004 12:03 pm, Dave Cotton wrote:
<DIV></DIV>> > On Thu, 2004-01-08 at 17:28, Steve wrote:
<DIV></DIV>> > > On Thursday 08 January 2004 03:22 am, Dave Cotton wrote:
<DIV></DIV>> > > > I just downloaded my mail to start the day, SIPphone had emailed me
<DIV></DIV>> > > > with a firmware update for GS, having had exactly the problem you
<DIV></DIV>> > > > outline, I've down loaded the new firmware (1.0.4.38 from TFTP
<DIV></DIV>> > > > 130.94.123.253) because their email states:-
<DIV></DIV>> > >
<DIV></DIV>> > > This sounds good! But, how did you come to have that version and their
<DIV></DIV>> > > website still only has 1.0.3.81?
<DIV></DIV>> >
<DIV></DIV>> > 130.94.123.253 came from SIPphone not Grandstream, but even
<DIV></DIV>> > http://www.grandstream.com/TEMP/FIRMWARE/ only has 1.0.4.30
<DIV></DIV>> >
<DIV></DIV>> > The only thing I can say is it's cleared my problems, making my GS
<DIV></DIV>> > usable again.
<DIV></DIV>>
<DIV></DIV>>Yes, that plus ensuring I was not using 723 on the Grandstream got it working.
<DIV></DIV>>With 723 it could not sync up with Asterisk.
<DIV></DIV>>--
<DIV></DIV>>Steve
<DIV></DIV>>
<DIV></DIV>>__________________________________________________
<DIV></DIV>>You actually need to constantly be alert
<DIV></DIV>> and willing to handle things, or life
<DIV></DIV>> will find a way to get you good!
<DIV></DIV>>
<DIV></DIV>>--__--__--
<DIV></DIV>>
<DIV></DIV>>Message: 13
<DIV></DIV>>From: Steve <STEVE@SZMIDT.ORG>
<DIV></DIV>>To: Asterisk List <ASTERISK-USERS@LISTS.DIGIUM.COM>
<DIV></DIV>>Subject: Re: [Asterisk-Users] SIP and error talking to voicemail
<DIV></DIV>>Date: Fri, 9 Jan 2004 16:47:45 -0500
<DIV></DIV>>Reply-To: asterisk-users@lists.digium.com
<DIV></DIV>>
<DIV></DIV>>On Thursday 08 January 2004 03:22 am, Dave Cotton wrote:
<DIV></DIV>> > On Thu, 2004-01-08 at 07:42, Steve wrote:
<DIV></DIV>> > > Hi,
<DIV></DIV>> > >
<DIV></DIV>> > > I used to have a Grandstream phone connected to Asterisk a few months
<DIV></DIV>> > > ago. Worked just great!
<DIV></DIV>> > >
<DIV></DIV>> > > Then today I do a new install, rather than an upgrade, and all of a
<DIV></DIV>> > > sudden I cannot check voicemail with it. No problem calling or receiving
<DIV></DIV>> > > call. It simply speeds through the vm greetings but I cannot hear them.
<DIV></DIV>> > > If I check the same VM with an analog phone it works fine.
<DIV></DIV>> > >
<DIV></DIV>> > > So I wanted to check if there's something known going on in these
<DIV></DIV>> > > particular areas?
<DIV></DIV>> >
<DIV></DIV>> > I just downloaded my mail to start the day, SIPphone had emailed me with
<DIV></DIV>> > a firmware update for GS, having had exactly the problem you outline,
<DIV></DIV>> > I've down loaded the new firmware (1.0.4.38 from TFTP 130.94.123.253)
<DIV></DIV>> > because their email states:-
<DIV></DIV>> >
<DIV></DIV>> > (1) Fix for voice echo problem during calls
<DIV></DIV>> > (2) Problem with dialing numbers
<DIV></DIV>> > (3) Speaker phone volume set to a higher volume
<DIV></DIV>> >
<DIV></DIV>> > Rebooted and for the first time recently * proudly announced "no
<DIV></DIV>> > messages" instead of "login incorrect". And I can hear it from the
<DIV></DIV>> > speaker.
<DIV></DIV>> >
<DIV></DIV>> > YMMV
<DIV></DIV>>
<DIV></DIV>>I forgot to say Thanks!
<DIV></DIV>>
<DIV></DIV>>--
<DIV></DIV>>Steve
<DIV></DIV>>
<DIV></DIV>>__________________________________________________
<DIV></DIV>>You actually need to constantly be alert
<DIV></DIV>> and willing to handle things, or life
<DIV></DIV>> will find a way to get you good!
<DIV></DIV>>
<DIV></DIV>>
<DIV></DIV>>--__--__--
<DIV></DIV>>
<DIV></DIV>>_______________________________________________
<DIV></DIV>>Asterisk-Users mailing list
<DIV></DIV>>Asterisk-Users@lists.digium.com
<DIV></DIV>>http://lists.digium.com/mailman/listinfo/asterisk-users
<DIV></DIV>>
<DIV></DIV>>
<DIV></DIV>>End of Asterisk-Users Digest
<DIV></DIV></div><br clear=all><hr> <a href="http://g.msn.com/8HMAENUS/2743??PS=">Expand your wine savvy — and get some great new recipes — at MSN Wine.</a> </html>