[Asterisk-Users] Multiline / Console / Receptionist phone
I have been looking to see if this type of phone can be implimented in *. I have found nothing conclusive. Is any out there using a multiline / mutlifunction phone typically used by a receptionist for transfering / routing calls? I need to know how this is accomplished or what alternative exists for this. Thanx! Gary P. -- Signature Prototypes Patterns Models Dies Fixtures Please visit www.jppattern.com for more information about J.P. Pattern, Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Multiline / Console / Receptionist phone
Does this phone have LEDs showing lines in-use? Thanx! Gary P. Tracy R Reed wrote: On Mon, Dec 13, 2004 at 12:50:54PM -0600, Gerald J. Puhl spake thusly: I have been looking to see if this type of phone can be implimented in *. I have found nothing conclusive. Is any out there using a multiline / mutlifunction phone typically used by a receptionist for transfering / routing calls? I need to know how this is accomplished or what alternative exists for this. I am using the Snom 220 with the hint extension priority with success. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Signature Prototypes Patterns Models Dies Fixtures Please visit www.jppattern.com for more information about J.P. Pattern, Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Granstream phones message button
Using: Software Version: Program--1.0.5.16 Bootloader--1.0.0.21 HTML--1.0.0.41 VOC--1.0.0.6 I will pursue changin firmware. Is this difficult? Gary P. Derek Conniffe wrote: I meant to say "downgrade to 1.0.5.11 Derek Derek Conniffe wrote: It sounds like you might be using the broken BT firmware - are you using 1.0.5.16? If you are you'll find that the message button and the auto-dial feature do not work. You can downgrade to 1.0.5.16 but I've read that the newer [BETA?] 1.0.5.18 works too. Derek Gerald J. Puhl wrote: To all: (newbie) I have setup a BT 100 phone and mostly everthing is working pretty good except for the message button. I have place value in the appropiate field in the web configuration but nothing seems to work. When I press the button the speakerphone led goes on but the phone does nothing else (no dialtone, no sip request to *). Does anyone have this buttton working? I would like to go right to voicemail. Gary P. -- * Prototypes * * Patterns * * Models * * Dies * * Fixtures * * Please visit /www.jppattern.com http://www.jppattern.com// for more information about J.P. Pattern, Inc. * ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Signature Prototypes Patterns Models Dies Fixtures Please visit www.jppattern.com for more information about J.P. Pattern, Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Ripping CD audio for MOH
Tom: I downloaded this freeware @ http://www.eusing.com/ (free CD to MP3). It converted my CD (CDA tracks) to MP3. Copied them onto my * server mohmp3 dir. You will need mpg123 for moh. I downloaded the rpm and installed. Everthing worked the first time I set it up. Gary P. Thomas Johnson wrote: Hello- I've got some audio CDs that I'd like to use for MOH. What's the best way to do this? I don't care if it's mp3 or some other format - whatever will work best. What applications (osx or linux) are best? Optimal settings? Thanks- Tom ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Signature Prototypes Patterns Models Dies Fixtures Please visit www.jppattern.com for more information about J.P. Pattern, Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Granstream phones message button
To all: (newbie) I have setup a BT 100 phone and mostly everthing is working pretty good except for the message button. I have place value in the appropiate field in the web configuration but nothing seems to work. When I press the button the speakerphone led goes on but the phone does nothing else (no dialtone, no sip request to *). Does anyone have this buttton working? I would like to go right to voicemail. Gary P. -- Signature Prototypes Patterns Models Dies Fixtures Please visit www.jppattern.com for more information about J.P. Pattern, Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SoftPhone on * with X-Lite or iaxComm (1 X100P card)
To all: I am having an echo problem with X-Lite and iaxComm. I am using the monitor speakers and a desktop microphone. My problem is that the sound from the speakers is repeated by the microphone causing an echo that is annoying. Is this correctable? I have searched throught this archives and played with various settings but have not been able to fix this. I have purchased a couple if different microphones that I thought may cure this problem, but nothing changed. Is this just simply a bad chioce and I need to use a headset rather than an open speaker microphone setup? Thanx, Gary P. PS: I am an * newbie, please go easy on me. -- Signature Prototypes Patterns Models Dies Fixtures Please visit www.jppattern.com for more information about J.P. Pattern, Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X101P interface (asterisk newbie)
To all: I am researching the feasibility of replacing our current PBX (ATT Partner Plus) with an * PBX. I have purchased an X101P card and I have * running of a FC2 machine. The X101P is connected to an extension on our current PBX. Many archives exists regarding X101P cards but I just need some guidance. First the X101P did not detect a ring, so I found an email where someone reconfigured the MINPEGTIME by decreasing the amount (did that and it seamed to work). Now sometimes it seams to get stuck and not detect a ring unless is repeatedly stop and restart *. Output form * in CLI shows a hang-up: Hang-up: channel: 1 index = 0, normal = 12, callwait = -1, thirdcall = -1 disabled echo cancellation on channel 1 Set option TDD MODE, value: OFF(0) on Zap/1-1 Updated conferencing on 1, with 0 conference users -- Hungup 'Zap/1-1' I assume that this means I am not having a trouble detecting a hangup? Is the X101P a finicky card (and perhaps a bad choice)? Or am I just not setting things right (and stupid). -- Signature Prototypes Patterns Models Dies Fixtures Please visit www.jppattern.com for more information about J.P. Pattern, Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
