Re: [asterisk-users] mysql CDRs in web based tool

2013-09-25 Thread Bob Kyeyune
.php /var/log/asterisk/cdr-csv/Master.csv path is the location where your cdrmysql.php is located second is the database already truncated for you hope this helps Regards. Kyeyune Bob VOIP Engineer +256 774 702 258 On Wed, Sep 25, 2013 at 5:02 PM, Doug Lytle wrote: > >> but

Re: [asterisk-users] WebRTC softphone for Asterisk - any suggestion?

2013-05-31 Thread Bob Kyeyune
hello; hopefully u can help me i have asterisk vanilla installation 11 and i have also managed to install webrtc2sip how do i make asterisk to communicate with webrtc2sip c'se right now both run independently Regards. Kyeyune Bob Network & IT Engineer +256 774 702 258 bob.kyey...@onesol

Re: [asterisk-users] Can't register to Asterisk 1.6 with old Aastra phones

2013-04-30 Thread Bob Kyeyune
am also stuck with Alcatel lucent IP Touch 4018 any one connected them to Asterisk thanks Regards. Kyeyune Bob Network & IT Engineer +256 774 702 258 bob.kyey...@onesolutions.ug Integrated IT services from Plot 57B Luthuli Avenue Bugolobi, Kampala On Sun, Apr 28, 2013 at 11:56 PM, Ca

[asterisk-users] Asterisk 1.8 Streaming MOH timing interface

2013-02-04 Thread Bob Pierce
t timing source. I suspec that if I restart the Asterisk service everything will come back up the way that it did last time. However, I'm wondering if there would be a way to switch the Music On Hold module back to using pthread timing without restarting the Asteris

[asterisk-users] send record file to email

2013-02-01 Thread Bob Kyeyune
Hello; how do i embed and send the recorded file to email automagically exten => _1XXX,3,MixMonitor(${CALLFILENAME}|b|/usr/sbin/wav2mp3 ${CALLFILENAME} ${peeremail} ${EXTEN} ${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)} ) Regards. Kyeyune Bob Network & IT Engineer +256 774 702 258 b

Re: [asterisk-users] Problem with outbound dialing from remote phone

2011-10-14 Thread Bob Bosiljevac
Did anything else change on her home network that could correlate to the time this started flaking on you? (eg: a new router/gateway) BB -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk

Re: [asterisk-users] Polycom and auto answer

2011-08-08 Thread Bob Pierce
500,n,Hangup This is working fine in an environment with many 330s, some 450s, some 335s and a 550 all running 3.3.1 Hope this helps you out. Bob -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to A

Re: [asterisk-users] use ILBC installed from asterisk yum repositories

2011-08-02 Thread Bob Pierce
On Tue, Aug 2, 2011 at 4:39 PM, Lefteris Zafiris wrote: > You can write a short makefile for just codec_ilbc module, build it and > install it on your running asterisk system. You will have to install the > asterisk18-devel package and get the asterisk source code either from > a tar or from the s

[asterisk-users] use ILBC installed from asterisk yum repositories

2011-08-02 Thread Bob Pierce
Asterisk from source? Thanks, Bob -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk

Re: [asterisk-users] Macro issue under 1.8.5

2011-07-15 Thread Bob Pierce
I'm still using macro with Asterisk 1.8.5.0 On Fri, Jul 15, 2011 at 3:17 PM, Paul Belanger wrote: > On 11-07-15 02:18 PM, Doug Lytle wrote: >> >> --[ UxBoD ]-- wrote: >>> >>> I back leveled to 1.8.3 and that works fine. What am I missing as >>> app_macro has been installed okay? >> >> Macro was d

Re: [asterisk-users] Asterisk 1.8 Paging without DAHDI or MOH Shoutcast with DAHDI

2011-07-11 Thread Bob Pierce
ads and DAHDI timing modules and force MOH to use pthreads and force Page() to use DAHDI? Bob On Tue, Jun 28, 2011 at 12:03 AM, Faisal Hanif wrote: > Call file are not suitable for you as asterisk process these files in serial > mode (single threaded) and in case of large number of files p

[asterisk-users] Asterisk 1.8 Paging without DAHDI or MOH Shoutcast with DAHDI

2011-06-27 Thread Bob Pierce
nyone have insight into how we could accomplish this paging feature or of anything that we may have missed? I suspect we could get this all to work with the original Page() application if there was a way to force MusicOnHold to use the pthread timing module instead of the Dahdi timing module. I

Re: [asterisk-users] Cannot install dahdi-linux on (old) PAE kernel.

2011-05-06 Thread Bob Beers
On Fri, May 6, 2011 at 2:12 PM, Bob Beers wrote: > On Fri, May 6, 2011 at 1:27 PM, Bob Beers wrote: >> Hi Steven, >> >> Can you put the .spec file from dahdi-linux-kmod package up? > >  Nevermind, I got it. :-) > > Looking at it now. Not sure if this will work

Re: [asterisk-users] Cannot install dahdi-linux on (old) PAE kernel.

2011-05-06 Thread Bob Beers
On Fri, May 6, 2011 at 1:27 PM, Bob Beers wrote: > Hi Steven, > > Can you put the .spec file from dahdi-linux-kmod package up? Nevermind, I got it. :-) Looking at it now. Did you CC [Packager: Jason Parker

Re: [asterisk-users] Cannot install dahdi-linux on (old) PAE kernel.

2011-05-06 Thread Bob Beers
package to recognise the kmod PAE package as > the right one for this kernel? Hi Steven, Can you put the .spec file from dahdi-linux-kmod package up? I can get the srpm, but I'm stuck on a weak machine at the moment. Maybe I can help you to mo

Re: [asterisk-users] asterisk 1.8 question

2011-03-25 Thread Bob Beers
els concise" "core show channels verbose" >From my experience, they all "work" in 1.8, but do give different output. -- HTH, - Bob Beers -- _ -- Bandwidth and Colocation Provided by http://www.ap

Re: [asterisk-users] Failover Routing

2011-03-01 Thread Bob Beers
ooked at SIP_HEADER() dialplan function? <https://wiki.asterisk.org/wiki/display/AST/Function_SIP_HEADER> Maybe you can parse Reason header in 4xx or 5xx response? HTH, -Bob -- _ -- Bandwidth and Colocation Provided by http://www.a

Re: [asterisk-users] Question about EuroBRI final 2 digits

2011-02-10 Thread Bob Beers
iki/display/AST/Application_DISA> cheers, -- -Bob -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hel

Re: [asterisk-users] How to disable srtp in asterisk 1.8.2.3?

2011-02-01 Thread Bob Beers
uld also work. Can you show your sip.conf and a trace of the call with 'sip set debug on' that shows the a=crypto: line in the INVITE SDP? It doesn't happen for me. I am able to send INVITEs with or without the a=crypto: line by setting e

Re: [asterisk-users] Mailing list question

2011-01-20 Thread Bob Beers
On Thu, Jan 20, 2011 at 12:03 PM, Danny Nicholas wrote: > Putting the "--" in front of it might make it go away. If I am not mistaken it should be exactly two dashes followed by a space on a line alone to indicate the end of the mail content. But not all mail readers will honor

Re: [asterisk-users] Go from CALLINGout to just CALLING

2011-01-04 Thread Bob Beers
er, (hyphen, pipe, comma, etc.) I can't really tell what you are trying to achieve, but if CALLINGout is the value of a variable, say X, and you want just the first 6 characters, you could use (maybe): exten => s,n,Set(newVAR=${X:0:6}) HTH, -Bob --

Re: [asterisk-users] How to append custom option to Contact: header on outgoing SIP INVITE msgs?

2010-11-19 Thread Bob Beers
The linewrapping by gmail of the patch file makes it difficult to read. So, I added it as an attachment for any interested readers. -- -Bob --- asterisk-1.8.0-beta2.orig/channels/chan_sip.c 2010-07-26 15:59:03.0 -0400 +++ asterisk-1.8.0-beta2/channels/chan_sip.c 2010-11-05 12:18

Re: [asterisk-users] How to append custom option to Contact: header on outgoing SIP INVITE msgs?

2010-11-19 Thread Bob Beers
On Fri, Nov 5, 2010 at 10:58 AM, Bob Beers wrote: > Hi list, > > My need is to append a site specific parameter to the >  Contact: header on all INVITEs exiting * via a SIP trunk. > I'd like it to look something like this: > > Contact: > > where SITE-ID=us.h

[asterisk-users] How to append custom option to Contact: header on outgoing SIP INVITE msgs?

2010-11-05 Thread Bob Beers
ing contact header parameters from arriving requests/registrations, but nothing about creating any such parameters. Thanks for any help/hints, -- -Bob Beers -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com

Re: [asterisk-users] Gotoif changed in 1.8?

2010-11-03 Thread Bob Beers
On Wed, Nov 3, 2010 at 4:32 PM, Danny Nicholas wrote: > TAM, Bob! Guess I've got to go through now and "unquote" my literals... Hi Danny, Glad that helped. But on second thought, maybe the better fix is to remove the double quotes in the Gotoif()'s, like thi

Re: [asterisk-users] Gotoif changed in 1.8?

2010-11-03 Thread Bob Beers
exten => s,n(ok),Set(BALCOUNT=0) > > > > -- Executing [...@tb-account-balance:7] GotoIf("SIP/134-", > "0?tb-account-balance,s,reset_bc") in new stack > > [Nov  3 14:23:02] WARNING[20937]: ast_expr

Re: [asterisk-users] Opensource Speech recognition for Asterisk

2010-08-24 Thread Bob Kleiner
> Thanks guys. A lot of info here :-) > > I am wondering if anyone followed this and it was working for them: > > http://scribblej.com/svn/ > > ??? Hello Bruce We successfully deployed it and now saving thousands on commercial ASR ports. It seems users are rather happy with it. The recognition se

Re: [asterisk-users] Calls Dropping

2010-05-07 Thread Bob Smither
On Sun, 2010-05-02 at 09:52 -0400, Dan Journo wrote: > Hi Bob, > > Thanks for that. Is there any way I can make the task run in the > background and free up the console? Also so that I can disconnect my > ssh session without losing the task. > > Thanks > Dan Matthieu

Re: [asterisk-users] Calls Dropping

2010-05-02 Thread Bob Smither
On Sun, 2010-05-02 at 08:34 -0400, Dan Journo wrote: > How can i log a continuous ping test to a file and include the date > and time of each ping? Try this: #!/bin/sh for (( ; ; )) do NOW=$(date +"%T %m/%d/%Y") PING=$(ping -qc 1 example.com) echo $NOW: $PING >> pinger.log done exit 0

Re: [asterisk-users] IAX Call Rejected (was IAX Problem)

2010-04-06 Thread bob gailer
On 4/6/2010 10:31 AM, Steve Edwards wrote: > On Tue, 6 Apr 2010, bob gailer wrote: > > >> I have2 Trixbox Servers. Each has an IAX trunks to the other. One works >> the other fails: >> >> -- Executing [...@macro-dialout-trunk:19] Dial("SIP/526-09ee

Re: [asterisk-users] IAX Problem

2010-04-06 Thread bob gailer
Apr 2010, bob gailer wrote: > > >> I have2 Trixbox Servers. Each has an IAX trunks to the other. One works >> the other fails: >> >> -- Executing [...@macro-dialout-trunk:19] Dial("SIP/526-09eec7c8", >> "IAX2/InterOffice/210,300,tr") in ne

[asterisk-users] IAX Problem

2010-04-06 Thread bob gailer
e-7578' == Everyone is busy/congested at this time (1:0/0/1) The only difference I am aware of is that one server has a public IP address, the other is behind a NAT. The trunk from the server with the public address works fine. I added nat=yes to the other's peer details - did no

Re: [asterisk-users] Aastra, Asterisk 1.4 and Voicemail

2010-03-08 Thread Bob Pierce
On Mon, Mar 8, 2010 at 7:08 PM, Mike wrote: > This seems like a basic thing to set up, so I have no doubt many people have > done this. Anyone care to point me in the right direction? In our config files, we have: softkey1 type: speeddial softkey1 label: "Voice Mail" softkey1 value: *97 This se

Re: [asterisk-users] Ringing for incoming call

2009-12-19 Thread Bob Smither
On Sat, 2009-12-19 at 08:26 -0500, cov...@ccs.covici.com wrote: > I have a strange suggestion -- have one extension answer the call and > dial the extension you want -- then it should ring before dialing the > second one. Actually, that is pretty close to what I do on a *1.6 box and it works. Her

Re: [asterisk-users] Ringing for incoming call

2009-12-19 Thread Bob Smither
mode=auto host= context=inbound username= secret= allow=all insecure=very nat=yes Context in extensions.conf: [inbound] exten => 8772709688,1,Goto(cci,s,1) The context [cci] is shown above. I appreciate the help, as I am confused! -- Bob Smither, PhD Circuit Concepts,

Re: [asterisk-users] Asterisk 1.6.2.0 Now Available!

2009-12-18 Thread Bob Smither
http://downloads.asterisk.org/pub/telephony/asterisk/ > > And any plan for Skype for Asterisk? > > Ira http://www.digium.com/en/products/software/skypeforasterisk.php (but it is not free) -- Bob Smither ___ -- Bandwidth and Colo

Re: [asterisk-users] Ringing for incoming call

2009-12-18 Thread Bob Smither
On Fri, 2009-12-18 at 19:58 -0600, Steve Johnson wrote: > Try putting the wait before the Answer. > > ... > exten => s,n,Wait(10) > exten => s,n,Answer > ... Thanks Steve. I tried that: > On Fri, Dec 18, 2009 at 5:10 PM, Bob Smither wrote: > > Dear All, &g

[asterisk-users] Ringing for incoming call

2009-12-18 Thread Bob Smither
ny useful information - for example the Ringing app is shown to run, but the caller does not hear it. Any suggestions? Many thanks! -- Bob Smither ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing li

Re: [asterisk-users] Getting multiple phones to ring ...

2009-12-15 Thread Bob Smither
On Mon, 2009-12-14 at 11:49 -0600, Bob Smither wrote: > This has to be easy, but I have spent a fair amount of time looking for > a solution to no avail. I am trying to get multiple phones to ring when > a call comes into an Asterisk box from a particular phone number. What > hap

[asterisk-users] Getting multiple phones to ring ...

2009-12-14 Thread Bob Smither
. Although phone bt112 is called, it does not ring. The phone bt116 is called and it does ring. I have set the call-limit parameter to 10 within the [general] context in sip.conf. What am I missing? Many

Re: [asterisk-users] Astricon

2009-10-21 Thread Bob Pierce
> Or charge for full access! Leave a few teasers, and charge some amount to > see them all. I would pay - even close to attendance price... could only > help you get past break even ;) I agree, I would be quite willing to pay for full access to all the videos from the Confere

Re: [asterisk-users] Asterisk 1.6.0.14 and 1.6.1.5 Now Available - pbx/lua.c changes

2009-08-29 Thread Bob Gustafson
The file pbx/pbx_lua.c needed to be changed slightly so as to find lua on my Fedora9 system [r...@hoho4 asterisk-1.6.1.5]# diff -u pbx/pbx_lua.c.orig pbx/pbx_lua.c --- pbx/pbx_lua.c.orig 2009-08-29 14:39:46.0 -0500 +++ pbx/pbx_lua.c 2009-08-29 14:40:20.0 -0500 @@ -42,9 +42,9

[asterisk-users] Switchvox HA options

2009-06-19 Thread Bob Pierce
What are the HA options for Switchvox systems? Is it possible to set up redundant systems with DRBD? I know on the digium website they talk about "Optional cold spare failover" What does this mean? Is this an active spare ready for some sort of automated failover? Thanks for you

Re: [asterisk-users] Dail in modem

2009-06-19 Thread Bob Pierce
On Fri, 2009-06-19 at 11:45 +0500, ABBAS SHAKEEL wrote: > I am required to do some thing like Dail in modem . > User will have to call a modem just like we do in dail up connection > now we need to handle that request and retrieve some parameters > from that send a HTTp request to a web serve

Re: [asterisk-users] show pri usage

2009-03-26 Thread Bob Pierce
On Thu, 2009-03-26 at 07:19 -0700, Vieri wrote: > Maybe I could simply do something like: > asterisk -rx "show channels" | grep -c -i zap > to get the number of zap/dahdi channels in use. I was actually using a command similar to that up until a few months ago. /usr/sbin/asterisk -rx 'show cha

Re: [asterisk-users] Aastra 480i repair?

2009-03-06 Thread Bob Pierce
and see if the fix is really a fix or if they are going to fail again. I dealt with Linda at Aastra: Linda Berendt Needless to say, we're not buying Aastra phones anymore. We're pretty happy with Polycom phones right now. Bob ___ -- Bandwidt

Re: [asterisk-users] Polycom Spectralink 8002 Configuration

2009-02-25 Thread Bob Pierce
'm not sure what you should do with the first line in that case. Bob ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Polycom Spectralink 8002 Configuration

2009-02-25 Thread Bob Pierce
555; secret LINE2 = LINE2_PROXY = 1 LINE2_CALLID = NOC Tech LINE2_AUTH= ; secret Bob ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: h

Re: [asterisk-users] Polycom Spectralink 8002 Configuration

2009-02-25 Thread Bob Pierce
As soon as I trimmed the config file down to just the necessary components, the phone started to work! Bob On Mon, 2009-02-23 at 21:07 -0500, M Hulber wrote: > I have a new Polycom Spectralink 8002 and am having trouble with the > configuration or the unit but I can't see what&#x

Re: [asterisk-users] don't get 2.0 gui to run on asterisk 1.6.0.5

2009-02-23 Thread Bob Pierce
I found that after moving to Asterisk 1.6 and the latest SVN of ASterisk-GUI, the link changed from: http://localhost:8088/asterisk/static/config/cfgbasic.html to: http://localhost:8088/static/config/cfgbasic.html I don't know if you'll find the same or not... Bob On Mon, 2009-02-2

[asterisk-users] Swift - detection of multiple digits unreliable on my system

2009-02-17 Thread Bob Hartwig
familiar to anyone? I am open to the possibility of using swift -o to generate to a WAV file, then using that file with read(), but I would like to avoid the delays and additional complexity associated with that technique, if possible. Thanks! Bob Hartwig ___

Re: [asterisk-users] GUI interface to manage business edition

2009-02-13 Thread Bob Pierce
On Fri, 2009-02-13 at 15:24 -0500, Miguel Martinez wrote: > I am new to the Asterisk world, but have decided to use the business > edition, but am looking for a cost effective gui interface to manage > the software. Does the Asterisk-GUI work with Asterisk Business Edition? ___

[asterisk-users] Broadcast Phone system (for radio)

2009-01-15 Thread Bob Pierce
this link: http://www.telos-systems.com/techtalk/digiphones/digiphones_4.htm States the following: "Generic PBXs will not do for our broadcast application – they just don’t have the features necessary. For example, while lines may certainly be shared to multiple phones, there is no way to switch g

[asterisk-users] Asterisk Appliance

2009-01-13 Thread Bob Pierce
phones? Some older specs stated they would be appropriate for businesses with 2-50 users, while the current spec on the Digium site states they are appropriate for 2-20 users. The application I'm thinking of would be VoIP only with a g.711 SIP trunk and g.711 phones

[asterisk-users] Asterisk as MGCP client

2008-12-29 Thread Bob Pierce
Asterisk system register to a local phone service using MGCP. Thanks for your help. Bob ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/m

[asterisk-users] SendImage() to Polycom ip550 or ip670

2008-12-10 Thread Bob Pierce
ith Polycom Phones? Bob ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] top posting again [was: Re: CDR Design]

2008-12-05 Thread Bob Gustafson
et me know. If there is a technological fix, perhaps these threads will die down. Bob G On Sat, 2008-12-06 at 14:47 +1300, Duncan Turnbull wrote: > I like the discussion, I doubt it will end. > > I prefer top posting because I reply to all my customers that way, my > mail client isn

Re: [asterisk-users] Call quality issue across VPN-> POTS vs SIP

2008-11-03 Thread Bob Pierce
On Mon, 2008-11-03 at 13:17 -0500, Lincoln King-Cliby wrote: > It's conceivable, but how would I verify this and how would I change > it if that was the problem? There's a few things you can do here. 1) Check the sip.conf on both sides to see what is defined there for the trunk. Look for some dis

Re: [asterisk-users] Call quality issue across VPN-> POTS vs SIP

2008-11-03 Thread Bob Pierce
line are going across the VPN as uLaw while the calls from the sip phones are using a compressed codec? Bob ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit

Re: [asterisk-users] hex b1 in CallerID sent by Asterisk On PRI

2008-10-22 Thread Bob Pierce
On Wed, 2008-10-22 at 12:11 -0500, Bob Pierce wrote: > On Tue, 2008-10-21 at 13:56 -0500, Bob Pierce wrote: > > Does anyone know what the significance is of the b1 being sent here? > > > > Or, is there a way to make Asterisk not send the b1 character as a > > test?

Re: [asterisk-users] hex b1 in CallerID sent by Asterisk On PRI

2008-10-22 Thread Bob Pierce
On Tue, 2008-10-21 at 13:56 -0500, Bob Pierce wrote: > Does anyone know what the significance is of the b1 being sent here? > > Or, is there a way to make Asterisk not send the b1 character as a > test? As an update to this, I noticed the following lines in libpri.h near line 236:

[asterisk-users] hex b1 in CallerID sent by Asterisk On PRI

2008-10-21 Thread Bob Pierce
t the significance is of the b1 being sent here? Or, is there a way to make Asterisk not send the b1 character as a test? I've pasted a portion of the PRI debug trace below. Thanks for your help. Bob -- Making new call for cr 32985 > Protocol Discriminator: Q.931 (8) len=55 >

Re: [asterisk-users] FAX over T1 Question

2008-09-05 Thread Bob Pierce
ls. If I'm not mistaken, that can mess up fax machine communications. Bob ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-us

Re: [asterisk-users] is shared_lastcall available in 1.4

2008-08-27 Thread Bob Pierce
y this in a few weeks when I start rebuilding the permanent system that will replace our current temporary system. That should give us the opportunity to test it on the bench instead of playing around with the production box. I'll probably be back

Re: [asterisk-users] is shared_lastcall available in 1.4

2008-08-26 Thread Bob Pierce
On Tue, 2008-08-26 at 17:53 +0300, Atis Lezdins wrote: > > Are there any plans to back port this feature into upcoming 1.4 > > releases? > > > > No, new features are added only in trunk, and released in next major > release (1.6). So what would be involved in back porting this feature for our sy

Re: [asterisk-users] is shared_lastcall available in 1.4

2008-08-26 Thread Bob Pierce
On Tue, 2008-08-26 at 17:30 +0300, Atis Lezdins wrote: > I'd say - go for backport instead. shared_lastcall is commited in > http://svn.digium.com/view/asterisk/trunk/apps/app_queue.c?r1=86820&r2=86985 > and it seems that there are no bugfixes for it since. So, backporting > should be fairly simpl

Re: [asterisk-users] is shared_lastcall available in 1.4

2008-08-26 Thread Bob Pierce
On Mon, 2008-08-25 at 17:47 -0500, Bob Pierce wrote: > > I'm glad to hear that you're enjoying Asterisk 1.4! Unfortunately, > the > > shared_lastcall option is only in versions 1.6.0 and up. > > > > Does anybody have a workaround for this in 1.4? Or maybe

Re: [asterisk-users] is shared_lastcall available in 1.4

2008-08-25 Thread Bob Pierce
On Mon, 2008-08-25 at 17:46 -0500, Mark Michelson wrote: > I'm glad to hear that you're enjoying Asterisk 1.4! Unfortunately, the > shared_lastcall option is only in versions 1.6.0 and up. > Does anybody have a workaround for this in 1.4? ___ -- Bandw

[asterisk-users] is shared_lastcall available in 1.4

2008-08-25 Thread Bob Pierce
h better now, and the AutoFill parallel call distribution is really nice. Bob ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users

Re: [asterisk-users] Reverse Scenario

2008-07-17 Thread Bob Pierce
oom on box 1, but I can't think of how to do that right now. Bob ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-user

Re: [asterisk-users] Problem compiling Zaptel

2008-07-14 Thread Bob Smither
On Sun, 2008-07-13 at 22:12 -0400, Noah Miller wrote: > Hi Bob - > Are you using any Xorcom hardware? If not, you can avoid this issue > by disabling the appropriate items when you run "make menuselect" > before compiling Zaptel. Thanks! That allowed me to compile. Re

[asterisk-users] Problem compiling Zaptel

2008-07-13 Thread Bob Smither
ere make[4]: *** [/projects/asterisk/zaptel-1.4.11/kernel/xpp/card_fxo.o] Error 1 ... I am not sure how to correct this. Any suggestions? Thanks! -- Bob Smither <[EMAIL PROTECTED]> ___ -- Bandwidth and Colocation Provided by http://www.api-digital.

Re: [asterisk-users] Asterisk video alternatives

2008-06-09 Thread Bob G
http://67.169.112.100/openmeetings/ Its OSS, runs on Linux and is not buggy - Original Message - From: "Sanjoy Rath" To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: Re: [asterisk-users] Asterisk video alternatives Date: Mon, 9 Jun 2008 04:41:06 + Hel

Re: [asterisk-users] Asterisk video alternatives

2008-06-06 Thread Bob G
I just got off IM with the owner of Vmukti.Hardik said the Yate woint be ready untillate August its Astrisk only for nowHe is also planning on moving to Sliverlight so the servcie will work on all browsers in August.I build a Free conferencing at 1ezphone.netIm doing any development on it but its b

Re: [asterisk-users] Asterisk video alternatives

2008-06-06 Thread Bob G
look at 1ezphone.net Its based off another OSS and runs on linux the user Interface is flash like 1ezphone.com - Original Message - From: "Matias Surdi" To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk video alternatives Date: Fri, 06 Jun 2008 10:57:37 +

Re: [asterisk-users] Browser based VoIP client?

2008-06-05 Thread Bob G
very full featured Date: Wed, 4 Jun 2008 20:59:28 -0500 On Wed, Jun 4, 2008 at 5:52 PM, Bob G wrote: > None of them have features like hold, transfer, voice mail, dtmf, conference > as far as I know none of them has caller ID > > Only 1ezphone.com has all that and t

Re: [asterisk-users] Browser based VoIP client? None of them are very full featured

2008-06-05 Thread Bob G
" Subject: Re: [asterisk-users] Browser based VoIP client? None of them are very full featured Date: Wed, 4 Jun 2008 20:59:28 -0500 On Wed, Jun 4, 2008 at 5:52 PM, Bob G wrote: > None of them have features like hold, transfer, voice mail, dtmf, conference > as far as I know

Re: [asterisk-users] Avaya IP Phones with *

2008-06-04 Thread Bob G
IP G450/S8700 system.) From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bob G Sent: Wednesday, June 04, 2008 3:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Avaya IP Phones with * Yes we do everyday here at Go

Re: [asterisk-users] http://1ezphone.com/download = sorry no "s"

2008-06-04 Thread Bob G
sorry its http://1ezphone.com/download - Original Message - From: "Bob G" To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: Re: [asterisk-users] Browser based VoIP client? - http://1ezphone.com/downloads Date: Wed, 4 Jun 2008 17:46

Re: [asterisk-users] Browser based VoIP client? None of them are very full featured

2008-06-04 Thread Bob G
None of them have features like hold, transfer, voice mail, dtmf, conferenceas far as I know none of them has caller ID Only 1ezphone.com has all that and the buttons are programmable for CRM features. - Original Message - From: "Tim Panton" To: "Asterisk Users Mailing List - Non-Com

Re: [asterisk-users] Avaya IP Phones with *

2008-06-04 Thread Bob G
Yes we do everyday here at Google - Original Message - From: "Mark Best" To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: [asterisk-users] Avaya IP Phones with * Date: Wed, 4 Jun 2008 15:24:16 -0700 Does anyone have any experience getting Avaya phones wo

Re: [asterisk-users] Browser based VoIP client?

2008-06-04 Thread Bob G
You can download a FREE browser softphone and or cliick to call that supports UDP athttp://1ezphone.com/download It works well with Asterisk I use it everyday - Original Message - From: "Hilary Miller" To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: Re: [aste

Re: [asterisk-users] Browser based VoIP client? - http://1ezphone.com/downloads

2008-06-04 Thread Bob G
you can download a FREE browser softphone and or clcik to call at http://1ezphone.com/downloads Let me know if you have any porblems and I can help you - Original Message - From: "Hilary Miller" To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: [asterisk-users]

Re: [asterisk-users] Asterisk - CRM Integration

2008-04-30 Thread Bob G
I just finished an integartion with SalesForce.com's connector and it worked pretty good. - Original Message - From: "Fernando Berretta" To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: [asterisk-users] Asterisk - CRM Integration Date: Tue, 29 Apr 2008 18:16

Re: [asterisk-users] Best Click-to-call client

2008-04-24 Thread Bob G
I use 1ezphone because its not activex and works all operating systems and browser.Plus the codec is great and only uses 10k - Original Message - From: Steven To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Best Click-to-call client Date: Thu, 24 Apr 2008 08:20:

[asterisk-users] Sip or IAX device with professional balanced audio out

2008-04-17 Thread Bob Pierce
might work nice here, or is there some other way of tackling this problem that I may have overlooked? Thank for your suggestions. Bob ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or

Re: [asterisk-users] Drag and Drop transfer application

2008-04-16 Thread Bob G
Sorry - Original Message - From: "Tzafrir Cohen" To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: Re: [asterisk-users] Drag and Drop transfer application Date: Thu, 17 Apr 2008 00:00:57 +0300 On Wed, Apr 16, 2008 at 03:24:15P

Re: [asterisk-users] Drag and Drop transfer application

2008-04-16 Thread Bob G
Why the guy asked a question? - Original Message - From: "Lee Jenkins" To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: Re: [asterisk-users] Drag and Drop transfer application Date: Wed, 16 Apr 2008 16:21:54 -0400 Bob G wrote: &g

Re: [asterisk-users] Best Click-to-call client

2008-04-16 Thread Bob G
Introducing Click-to-Call Posted: 16 Apr 2008 9:55 AM PDT The 1EZphone browser softphone has created so much buzz in the media that a lot of individual users and companies who have a web-presence; Websites, Online Advertising, Blogs, Customer support etc have asked for a Click-to-Call service. T

Re: [asterisk-users] PSTN to SIP

2008-04-16 Thread Bob G
Rhino or audiocode PSTN gateway - Original Message - From: "mark morreny" To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: [asterisk-users] PSTN to SIP Date: Thu, 17 Apr 2008 01:25:49 +0800 Dear all, A quick question on deploying Asterisk over E1. I am

Re: [asterisk-users] Drag and Drop transfer application

2008-04-16 Thread Bob G
Introducing Click-to-Call Posted: 16 Apr 2008 9:55 AM PDT The 1EZphone browser softphone has created so much buzz in the media that a lot of individual users and companies who have a web-presence; Websites, Online Advertising, Blogs, Customer support etc have asked for a Click-to-Call service. T

Re: [asterisk-users] multiple simultaneous access to single voice mail box

2008-04-10 Thread Bob Pierce
need. Have a great day. Bob ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] multiple simultaneous access to single voice mail box

2008-04-09 Thread Bob Pierce
ge. Now I can be taking off the messages & some one else can also be taking off the same messages. We should not be able to do this!!" Has anyone else seen this? Is there a way to setup the voice mail so that each box can only be accessed by one pers

Re: [asterisk-users] Polycom IP600 + PC share same switch port with VLAN

2008-02-28 Thread Bob G
You can paste and copy nterface FastEthernet2/0/1 switchport access vlan 20 switchport mode access switchport voice vlan 120 srr-queue bandwidth share 10 10 60 20 srr-queue bandwidth shape 10 0 0 0 mls qos trust device cisco-phone mls qos trust cos auto qos voip cisco-phone spanning-tree port

[asterisk-users] I would like to hire someone to automate my asterisk for hosted PBX service

2008-02-28 Thread Bob Gibson
I would like to hire someone to automate my asterisk for hosted PBX service for fetures like user signup, adding money and call bridging Please contact me offline at [EMAIL PROTECTED] - Original Message - From: "Philipp Kempgen" To: "Asterisk Users" Subject: Re: [asterisk-users] Ru

Re: [asterisk-users] 1EZphone is only two way browser softphone - SIP Softphones and Citrix ?

2008-02-28 Thread Bob Gibson
Yes, try http://1ezphone.com its a browser softphone. - Original Message - From: Zoa To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: Re: [asterisk-users] SIP Softphones and Citrix ? Date: Fri, 01 Feb 2008 23:09:56 +0200 I'm working for zoiper.com and i

Re: [asterisk-users] BRI card with PCI-E interface

2008-02-01 Thread Bob Pierce
gt;http://lists.digium.com/mailman/listinfo/asterisk-users > > The Sangoma A500BRX is a 2 to 6 bri pci-x interface, > although I've never tested it > (A500BRECX comes with hw echo cancellation). > We've been using the Sangoma A104DE in production for almost a yea

Re: [asterisk-users] X100P Woes

2008-01-04 Thread Bob Smither
On Wed, 2007-12-26 at 17:47 +0200, Tzafrir Cohen wrote: > On Wed, Dec 26, 2007 at 08:38:31AM -0600, Bob Smither wrote: > > On Wed, 2007-12-26 at 07:27 +0200, Tzafrir Cohen wrote: > > > > > Just to stress the point: you mention that you don't see the card on > >

Re: [asterisk-users] X100P Woes

2007-12-26 Thread Bob Smither
On Wed, 2007-12-26 at 10:22 -0500, dave cantera wrote: > bob, > > look on p20 of 'the book' edition 2, or p16 edition 1 > this explains the 3.3v vs 5.0v issue with motherboard slots > > http://www.oreilly.com/catalog/9780596510480/ > daveC The slots are 5V PCI o

Re: [asterisk-users] X100P Woes

2007-12-26 Thread Bob Smither
On Wed, 2007-12-26 at 15:46 +1000, Mattt wrote: > Sounds like a PCI bus version issue ;-) Thanks Matt, but could you elaborate? The card supposedly supports both 5 and 3.3 volt PCI slots, and the slots on the older system that works with the card appear the same as on the board giving me grief.

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