search for: ekn

Anyone try the new Cisco firmware for the 79xx sip phones? In my test it seems to work fine for a little and than soon the phone looses its time. At first the status shows clear, and then it appears to get confused about the ntp time source and the time goes away on it. No features, just bug fixes. Perhaps I have something wrong with my ntp settings, but the other firmwares work just fine. --

Does anyone have any experience with an Asterisk compatible softphone application which meets the following criteria: 1) Is able to use touch screen rather than mouse for on-screen functions. 2) Has an API which can be used to export Caller ID info to another App on the same compuer. Thanks Bill

Hi, No one seems to have any issue with the following posting. Can any one suggest how to install/configure channel drivers to work. Basically I am trying to send the SIP calls to GNUGK but Asterisk reports the error "No channel driver found". >>> I was trying to compile the oh323 channel driver but unable to compile the openh323_1_13_5 (which is the only required version as

Is there a way to get the OnHold music to restart without restarting asterisk? -- respectfully, Joseph - (606) 477-2355 x140 ------=============

Does anyone know if the 3rd patch listed on this bug fixes the seg fault problem related to voicemail? http://bugs.digium.com/view.php?id=4800 -- respectfully, Joseph --------------------

...ext = main callwaiting = 1 mailbox = 5000 ; Another line. [79140] id = 5929 pin = label = Craig context = main [79141] id = 5933 pin = label = Andy context = main Message: 4 Date: Fri, 05 Aug 2005 14:53:39 -0400 From: Joseph <tech@ekn.com> Subject: Re: [Asterisk-Users] Cisco 7914 To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <1123268019.6987.132.camel@localhost> Content-Type: text/plain On Fri, 2005-08-05 at 14:09 -0400, Craig Bruenderman wrote: > Ho...

What third party programs are available for parsing the queue_log file and CDR file? I know about XC-AST, but management would prefer a php based solution. What have other admins done to retrieve detailed call information about the queue system? Anyone develop their own that they don't mind sharing? --johann

Was in the process of upgrading a 7960 to SIP and in advertently applied a skinny image (P003G302.bin), now no matter what i put in OS79XX.TXT and OS7960.TXT it simply wont upgrade. The phone is pulling down OS79XX.TXT from my TFTP server but then goes on to repetedly ask for SEP<MACaddr>.cnf. Help!! -------------- next part -------------- An HTML attachment was scrubbed... URL:

...-sodon5E(a zZ*6=Ofp3;M3QEiXr9wn7n(k#j)-PS7k)9+GgirB2pOp*9E{eLy7XMYFe~Rx3BWAH& zh7g@T966=TK4iY9O6S8`1IE~VEW5t8Qg=6Oo~{9Tl98#i!r@3Wi4^n5u!J40_bKR{ z4!bk3u%J!Y=;l$ll)xS~Wd~CWHeS2vofQL}p?<<K(XHiy9D?n?tHZEVF|PP{>_%Ec zf|Am1iIcSnEan7i0@-=F6i;y_^^B2!29lb)+7I@vIM+|-SC58#yvzcX{5!{u{EsdU z0#&ekN^Y}H?F=m(nqKR7r=K|TN9e2stzgn)e@twA8XlP(_A)^dq%=X@VZ{uea>>p% z*DTMZ-om-sp9bwuZR(+B+y2>?x);6|ua_{j=g}tn^TJ{izMLc*q;u*#9}XdGW4d0q zfH@uSnt`^Ej0-Gbx4*k68vz4#lNv5Me|8lyYc*Keh#%xZb_$Q?V(hGI;Kgv=okd!7 zmKvvTtElv+_LP@$`zEUt_TE0jmChq=L2cu{*ZgkKptPlVDZnkFAH?5?A4&o5o5e9p zQw##izuG4&0gVaS9invhG...

...t;l*T0*N|6jpz1D{ePS<{eIl02KH>PLt6Nm^TIbNn+N33-8#J1t z2B|7n)K#>GR$kMV_LT>B at PxI{h<;@FcW}8!2ij%cxIkQC<z4g8DcQDdNhhVNi at WFt zaf|EWF8*QM;!NBnKg@%T+sTw{ZWtS^(gIjJ6fr3-sHhcNGQ?VEimpbeAb0CzTI)=8 zEvlotWmK(^qV7*GAxfoMVR<QUTdKE(%GcSxd_b(5b(Z3t>2#qlQC6CjJBHR1vm!pR zo|uTJ(WM5ZMLhJsveKN~zY;=DB>S3jItLNqKxkW_GL{>nbGPPjy#Pv#oeiBVMbJnQ zR9vbTBj|b&^p+b$SmMC0IYOtsX>A2oD%dGV+p1NG6g^ZyGH_6lZ_dfiE32zBa*XVo zvwV4v<2B2mTkM92+M#uJ(Y)og3CAm%>?L+OiN-0f73Uw_fJGF5!JQOmQbd!Ki=&&b zsZa~FyU{zh at q~XHm&$9c8lIwhNYMIxUh{g3q>$Q(ji}+JMnt9c(xtSJG8Foes{KK1 z>(i+6`S(yOY...

I am trying to get an understanding of how line appearances work like on the cisco 7960 phones. Is there a wiki somewhere about how this works? Also, the 7960 phones let you register more than one ext. Why would you want more than one or is this connected to line appearances? Is there a way to have phones use more than one codec, say use g.711 to talk with * and g.729 to talk with another

Has anyone tried to get dialogic ss7 trunking to work with Asterisk? I did some googling but nothing helpful turned up... Could a person get an intel dialogic and get * to see it like a zap channel? And use it for incoming and outgoing trunk access? -- respectfully, Joseph ------=============

There were a number of messages about using a digium card as a data t1 (router) kind of thing. Is there a howto out there or a status that I can see whether this is feasible or not? Or is it feasible but very difficult? -- respectfully, Joseph - (606) 477-2355 x140 ------=============

I made a little php script that creates a 79xx config if you give it the mac address, ext, etc. Is this something that would be of interest to anyone? Likely it could be improved on. And there may be some variations that I have not thot of. -- respectfully, Joseph =============== ---------------------= ********** =

Is there a way to get the que logs to go to a postgresql database? And/or are they any documents that explain the layout and field meanings? The logs look like they will be very helpful. Especially to show how long calls are held and who is taking them. I would like to make simple web reporting tool to show a daily summary status of queue activity. Also, there is not a manager command to do

We have an Adit 600 with some fxo lines. Callers calling in on the fxo ports do not hear echo, but the party answering hears bad echo. Here is the gain settings on the Adit: Corporate Adit 600> show 1 SLOT 1: Settings for FXO: Channel 1: Signaling: LSCPD RxGain: -2 dB TxGain: -3 dB And here are the zap settings: busydetect=no callprogress=no

Has anyone tried compiling the zaptel stuff under the 2.6.11 kernel? I get all kinds of errors when doing that. On the 2.6.10 it works fine. This is using cvs. Also, any news on when a 1.2 will be released? Or even available for download? :) -- respectfully, Joseph =============== ---------------------= ********** =

I notice these notices running cvs head: NOTICE[6154] chan_sip.c: No field 'From' present to copy Any ideal what make this notice? -- respectfully, Joseph =============== ---------------------= ********** =

When sending a call to a line defined on chan_sccp, there is an error on the console that says: Jun 7 08:22:29 WARNING[3924]: sccp_channel.c:79 sccp_channel_send_callinfo: Incoming call SCCP/Line1-00000008 doesn't have CallerId name Is this because of the changes in the callerid name from stable to head? Or does someone know of a way to correct this? -- respectfully, Joseph

This says the latest stable version is 1.0.7... http://www.voip-info.org/tiki-index.php?page=Asterisk+Download But it look like the latest stable version is 1.0.9. Am I missing something? -- respectfully, Joseph =============== ---------------------= ********** =