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.
> > case 1 > > ------ > > [sip-a] > > allow=g729 > > disallow=all > > allow=alaw > >Try: >[sip-a] >disallow=all >allow=g729 >allow=alaw > >The "disallow=all" clears your previous setting of "allow=g729" > >-- >END OF LINE > -MCP > >--__--__-- > >Message: 5 >Subject: RE: [Asterisk-Users] Cisco Gear >From: Steven Critchfield <[EMAIL PROTECTED]> >To: [EMAIL PROTECTED] >Date: Fri, 09 Jan 2004 15:23:36 -0600 >Reply-To: [EMAIL PROTECTED] > >On Fri, 2004-01-09 at 14:20, Arnold Ligtvoet wrote: > > message posted on behalf Of Adthrawn > > [SNIP cisco stuff] > > > I'll now feel ashamed, and sink into my seat :-) > > > > > > Best, > > > Ad. > > > > Perhaps it would have been better to provide an email address or phonenumber > > where people can contact you directly. Now everybody who is interested has > > to reply to the list. > >Maybe you should learn how to use your email client. I emailed this >person at the email address used for the original message. I have >already exchanged several messages with him. So I can vouch for the fact >that the email address is good and checked. >-- >Steven Critchfield <[EMAIL PROTECTED]> > > >--__--__-- > >Message: 6 >Date: Fri, 9 Jan 2004 22:24:55 +0100 (CET) >Subject: Re: [Asterisk-Users] USA dial plan >From: [EMAIL PROTECTED] >To: [EMAIL PROTECTED] >Reply-To: [EMAIL PROTECTED] > > > Hi, > > > > Do the callers in USA dialling from USA Telco lines always have to > > prefix the CITY/AREA code with "1" in order > > To successfully make a call to other USA destinations? > > > > ---- > > I have not been to USA (yet) :) > > > > Ta > > SJ > >For comprehensive info by area code (and as pointed out it does differ >from location to location) check the North American Numbering Plan website >at http://www.nanpa.com/. Left menu click on Dialing Plan and then go to >the location of your choice. > >Robert > >--__--__-- > >Message: 7 >Date: Fri, 09 Jan 2004 22:28:17 +0100 >From: "Olle E. Johansson" <[EMAIL PROTECTED]> >Organization: Edvina AB >To: [EMAIL PROTECTED] >Subject: Re: [Asterisk-Users] * as sip b2bua? >Reply-To: [EMAIL PROTECTED] > >Thilo Salmon wrote: > > > Hi everyone, > > > > any chance * could be used as a b2bua without forcing the media stream > > through the same box? I would love to do some computing on incoming > > calls, do things like setting another callerid and the forward the call > > to another sip UA - all without any audio traversing the * box. Any > > ideas? >Thilo, >Isn't the definition of a b2bua that the media streams pass it? >back-to-back-user-agent. > >Anyway, not to be picky, Asterisk by default wants to be in the media >path. There are ways to release the signalling and media path >back to the clients, with canreinvite=yes, but that's not the default behaviour. > >A SIP proxy like SIP express router from iptel.org fits your >description better. And yes, SER works together with Asterisk. > > >/Olle > > >--__--__-- > >Message: 8 >Subject: RE: [Asterisk-Users] Mailing list growth >Date: Fri, 9 Jan 2004 16:29:42 -0500 >From: <[EMAIL PROTECTED]> >To: <[EMAIL PROTECTED]> >Reply-To: [EMAIL PROTECTED] > > > -----Original Message----- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf Of=20 > > Philipp von Klitzing > > Sent: Friday, January 09, 2004 6:52 AM > > To: [EMAIL PROTECTED] > > Subject: Re: [Asterisk-Users] Mailing list growth > >=20 >[...] > > "higher-level implementation" list that deals specifically with=20 > > channelbanks & T1 issues (=3Dlarger installations). VoIP will remain = >on=20 > > asterisk-users. >[...] > >That doesn't quite sound right. Maybe it is from your perspective, but >are you telling me that the NOCs with 80+ 7960's running VoIP don't >count as a large installation? > >Of course, the term large is also relative. A 4-port T1 card on its >own....even 2 or 3 of them, could never by any stretch of my imagination >be considered a large installation......but I deal with (among other >things) Definity's that service near entire buildings in mid-town >Manhattan with multiple DS3s....so it's all relative. > >The problem with splitting VoIP and T1/TDM/whatever you want to call it >is that the crossover is huge, and where the problems lie often aren't >clear to those looking for help. >Daryl G. Jurbala >BMPC Network Operations >Tel: +1 215 825 8401 x235 >Fax: +1 508 526 8500 >INOC-DBA: 26412*DGJ > >PGP Key: http://www.introspect.net/pgp=20 > >--__--__-- > >Message: 9 >Date: Fri, 09 Jan 2004 14:33:09 -0700 >From: David Burr <[EMAIL PROTECTED]> >To: [EMAIL PROTECTED] >Subject: Re: [Asterisk-Users] DTMF in MeetMe >Reply-To: [EMAIL PROTECTED] > >the * and # are hard coded. >unless "b' -- run AGI script specified in ${MEETME_AGI_BACKGROUND} >is what your refering to.. which doesnt say how to use it. > > >Jeremy McNamara wrote: > > > David Burr wrote: > > > >> Does the MeetMe monitor for DTMF tones to trigger an AGI? > >> If not is this a planned feature? > > > > > > > > show application MeetMe > > > > > > Jeremy McNamara > > > > > > _______________________________________________ > > 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 > > > > > > >--__--__-- > >Message: 10 >Subject: RE: [Asterisk-Users] Screen Pop & Remote Agents = Telemarketing >Date: Fri, 9 Jan 2004 16:38:52 -0500 >From: <[EMAIL PROTECTED]> >To: <[EMAIL PROTECTED]> >Reply-To: [EMAIL PROTECTED] > >-----Original Message----- >From: [EMAIL PROTECTED] >[mailto:[EMAIL PROTECTED] On Behalf Of empire >underground >Sent: Friday, January 09, 2004 1:32 PM >To: [EMAIL PROTECTED] >Subject: [Asterisk-Users] Screen Pop & Remote Agents > > can I put a .csv file in the sql DB and have it dial from there? and >will I be able to set a > > Dial Plan to only call certin area codes? stuff like that. The reason >I ask all this is because > > all of these over priced dialers do just that. Also can Asterisk be >set with the FTC laws to 3% > > droped call ratio? > > If all of the questions I have asked here have allready be answered >some point in time... Can > > someone pl ease point me in the right direction to get all the >answers. > >So you're setting up a telemarketing rig? That's certainly not the kind >of thing I'd expect to get much help or sympathy for ANYWHERE other than >in telemarketing circles. > >I think the cost of a proper commercial predictive dialer would be >relatively cheap after already having sold your soul. > >Daryl > >--__--__-- > >Message: 11 >From: Steve <[EMAIL PROTECTED]> >To: [EMAIL PROTECTED] >Subject: Re: [Asterisk-Users] Cisco Gear >Date: Fri, 9 Jan 2004 16:41:20 -0500 >Reply-To: [EMAIL PROTECTED] > >On Friday 09 January 2004 02:40 pm, Iain Stevenson wrote: > > Prices? Are we talking a 7960 for the price of a SNOM? > > > > Iain > >Oops, just realized I replied to the wrong person.... > > > --On Friday, January 9, 2004 6:00 pm +0000 Adthrawn > > > > <[EMAIL PROTECTED]>wrote: > > > Hi, > > > > > > I know it's not really the place, but if anybody in the UK (or US) is > > > interested, I'm clearing out lots of new Cisco stock... > > > > > > 7970G's (colour LCD), 7960G's, 7940G's, 7920G's (wireless IP phone), > > > 7935's (conference phone) and 3550-24-PWR switches. > > > > > > I also have boxes of 7914's, the single-7914 foot stand and double-7914 > > > foot stand (these are required to connect a 7914 to a 7960G). > > > > > > And some useful locking and non-locking wallmount brackets for 79xx > > > range. > > > > > > We also have lots of PSU's for the whole 79xx range. > > > > > > I'll now feel ashamed, and sink into my seat :-) > > > > > > Best, > > > Ad. > > > > > > _______________________________________________ > > > 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 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 > >-- >Steve > >__________________________________________________ >You actually need to constantly be alert > and willing to handle things, or life > will find a way to get you good! > >--__--__-- > >Message: 12 >From: Steve <[EMAIL PROTECTED]> >To: [EMAIL PROTECTED] >Subject: Re: [Asterisk-Users] SIP and error talking to voicemail >Date: Fri, 9 Jan 2004 16:47:14 -0500 >Reply-To: [EMAIL PROTECTED] > >On Thursday 08 January 2004 12:03 pm, Dave Cotton wrote: > > On Thu, 2004-01-08 at 17:28, Steve wrote: > > > On Thursday 08 January 2004 03:22 am, Dave Cotton wrote: > > > > I just downloaded my mail to start the day, SIPphone had emailed me > > > > with a firmware update for GS, having had exactly the problem you > > > > outline, I've down loaded the new firmware (1.0.4.38 from TFTP > > > > 130.94.123.253) because their email states:- > > > > > > This sounds good! But, how did you come to have that version and their > > > website still only has 1.0.3.81? > > > > 130.94.123.253 came from SIPphone not Grandstream, but even > > http://www.grandstream.com/TEMP/FIRMWARE/ only has 1.0.4.30 > > > > The only thing I can say is it's cleared my problems, making my GS > > usable again. > >Yes, that plus ensuring I was not using 723 on the Grandstream got it working. >With 723 it could not sync up with Asterisk. >-- >Steve > >__________________________________________________ >You actually need to constantly be alert > and willing to handle things, or life > will find a way to get you good! > >--__--__-- > >Message: 13 >From: Steve <[EMAIL PROTECTED]> >To: Asterisk List <[EMAIL PROTECTED]> >Subject: Re: [Asterisk-Users] SIP and error talking to voicemail >Date: Fri, 9 Jan 2004 16:47:45 -0500 >Reply-To: [EMAIL PROTECTED] > >On Thursday 08 January 2004 03:22 am, Dave Cotton wrote: > > On Thu, 2004-01-08 at 07:42, Steve wrote: > > > Hi, > > > > > > I used to have a Grandstream phone connected to Asterisk a few months > > > ago. Worked just great! > > > > > > Then today I do a new install, rather than an upgrade, and all of a > > > sudden I cannot check voicemail with it. No problem calling or receiving > > > call. It simply speeds through the vm greetings but I cannot hear them. > > > If I check the same VM with an analog phone it works fine. > > > > > > So I wanted to check if there's something known going on in these > > > particular areas? > > > > I just downloaded my mail to start the day, SIPphone had emailed me with > > a firmware update for GS, having had exactly the problem you outline, > > I've down loaded the new firmware (1.0.4.38 from TFTP 130.94.123.253) > > because their email states:- > > > > (1) Fix for voice echo problem during calls > > (2) Problem with dialing numbers > > (3) Speaker phone volume set to a higher volume > > > > Rebooted and for the first time recently * proudly announced "no > > messages" instead of "login incorrect". And I can hear it from the > > speaker. > > > > YMMV > >I forgot to say Thanks! > >-- >Steve > >__________________________________________________ >You actually need to constantly be alert > and willing to handle things, or life > will find a way to get you good! > > >--__--__-- > >_______________________________________________ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users > > >End of Asterisk-Users Digest
Expand your wine savvy � and get some great new recipes � at MSN Wine. _______________________________________________ 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
