similar to: Some questions about a potential usage scenario for asterisk

Dave: Of course I know nothing about your project so you can politely ignore this message if it does not apply. However having worked in call centers all my life and used CTI extensively for "screen pop", "data pass", what ever you call it, I want to give you a quick heads up about using CID to identify callers. If you are in a line of business where callers will be calling

Hi 2 all. I've a nice issue my * system. I have: asterisk latest cvs T100P + Zhone cb (3fxo+10fxs used) Diva Server 4BRI running Capi a bunch of pots and sip phones The whole system is running good (thanks to capigod and Bic. for the help with the Eicon Diva), but there's a very strange issue when calls arrive on the Diva and then bridged to the pots on the channel bank: the voice is not

Hi list, Hopefully, some of our Dutch members can help with this one. I'm also based in the Netherlands and am using a Sipura (Linksys) SPA-3000 (firmware v3.1.10(GWd)) as a PSTN to VoIP gateway for my Asterisk test system. It works fine, except that the Called ID (CID) is not working. I'm aware that KPN (our local telco) requires a separate subscription to activate CID on POTS

I am using asterisk version 1.2.6 with Zaptel version 1.2.5. I have a POTs line coming into a Digium TDM01B. It appears to not be getting CID at all. If I hook up a POTS phone to the line CID comes through fine. Inbound and outbound calls work fine but there is just no CID on inbound for this channel.The incoming route for the channel is Zaptel Channel 0. No DID or CID settings applied. My IP

Hi list, After successfully configuring Linksys SPA3000 and SPA3102 devices as Asterisk PSTN gateways, the only thing I can't get working is the PSTN Caller ID. The analog and SIP phones I've used can both display CIDs for internal calls, while the analog model also displays CIDs correctly when attached directly to the PSTN line. However, when PSTN calls come in via the SPA

Hello I use the Linksys 3102 to connect Asterisk to a POTS line, and XLite on XP as an SIP client: http://img694.imageshack.us/img694/1421/3102asteriskxlitecid.png The problem is that by default, Asterisk doesn't rewrite the CID name + number in incoming calls, so that XLite displays whatever name I used in the 3102 and the extension the 3102 uses to register with Asterisk. How can I tell

Mine goes to chan_bluetooth. Somewhat of a pain getting it going but I am totally floored with how cool it is! Thanks, Steve Totaro

Does anybody out there run * on an AT&T broadband phone line? I'm not seeing any callerid and I can't tell if its AT&T doing something funky or if its my setup. I do see CID on my normal phones.... Thanks, Chris -- The face of a child can say it all, especially the mouth part of the face. http://ccicolorado.org Exceptional Dogs for Exceptional People - Help Out Today!

Good Morning, I'm having an issue with callerid display when calles are placed _from_ an mgcp device (8x8 ata w/mgcp firmware). Internally, there are several different sip devices and one mgcp device. Calls from any of the sip devices to any other device (sip or mgcp) have name/number displayed properly by the called party's phone. Calls from the mgcp device to any other device display

Hi, we''re seeing a really weird problem in our production environment. "Connection: Keep-Alive\r\n" is being pre-pended to the RAW_POST_DATA, which makes requests error out. This bug is seen infrequently and *seems* to be browser-related, although I can''t reproduce it using the browser reported in HTTP_USER_AGENT. Set up is a apache 2.2 -> hardware load balancer

Hi, we''re seeing a really weird problem in our production environment. "Connection: Keep-Alive\r\n" is being pre-pended to the RAW_POST_DATA, which makes requests error out. This bug is seen infrequently and *seems* to be browser-related, although I can''t reproduce it using the browser reported in HTTP_USER_AGENT. Set up is a hardware load balancer (NetScaler)

Hello list, I know this has been brought up before but I dont think there was ever a final answer. Is it legal in the US to modify asterisk to show the CID information that was received as blocked ? Thanks. Dovid p.s. Sorry for the poor typing format, it was written from a mobile phone. __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam

Hi I've got a cheap card from x100p.com for my pots line. I haven't found a definate answer if it is possible to get danish(DTMF without signaling before it). I have had a look at bug #9 but that is written longtime ago. I am running zaptel 1.4.7 and asterisk 1.4.14 BRIstuffed 0.4.0-test4 both from the xorcom debian packages. I believe they should contain some kind functionality for DTMF

I have the following two mapping data frames (r) and (h). I want to fill teh value of r$seid with the value of r$seid where r$cid==h$cid. I can do it with sapply as such: > r$seid = sapply(r$cid, function(cid) h[h$cid==cid,]$seid) Is ther a better (faster) way to do this? > r <- data.frame(seid=NA, cid= c(2181,2221,2222)) > r seid cid 1 NA 2181 2 NA 2221 3 NA

Hello I am trying to compile asterisk-oh323-0.7.0 with pwlib-Janus_patch4 and openh323-Janus_patch4 downloaded from inaccessnetworks so I did this: tar -zxvf openh323-Janus_patch4-src-tar.gz cd openh323 patch -p1 < /root/asterisk-oh323-0.7.0/openh323_1.13.5-make.patch ./configure make opt cd asterisk-oh323-0.7.0 vi Makefile (to set the paths and options according to my system...) NOW I

Hi, I have a problem for install chan_capi My pc: Suse 9.1, with asterisk current stable en cvs I have download http://www.junghanns.net/asterisk/downloads/chan_capi.0.3.5.tar.gz And the path from http://www.levigo.de/VoIP/chan_capi-0.3.5-cvs-HEAD-patch.tar.bz2 And patch the chan_capi # cd chan_capi-0.3.5 # patch -p1 < chan_capi-0.3.5-patch.diff # make /for install My error

On Tue, Mar 15, 2016 at 06:12:55PM +0200, Michael S. Tsirkin wrote: > On Tue, Mar 15, 2016 at 03:15:29PM +0000, Stefan Hajnoczi wrote: > > On Mon, Mar 14, 2016 at 01:13:24PM +0200, Michael S. Tsirkin wrote: > > > On Thu, Mar 03, 2016 at 03:37:37PM +0000, Stefan Hajnoczi wrote: > > > > Michael pointed out that the virtio-vsock draft specification does not > >

On Tue, Mar 15, 2016 at 06:12:55PM +0200, Michael S. Tsirkin wrote: > On Tue, Mar 15, 2016 at 03:15:29PM +0000, Stefan Hajnoczi wrote: > > On Mon, Mar 14, 2016 at 01:13:24PM +0200, Michael S. Tsirkin wrote: > > > On Thu, Mar 03, 2016 at 03:37:37PM +0000, Stefan Hajnoczi wrote: > > > > Michael pointed out that the virtio-vsock draft specification does not > >

Hello Instead of using PrivacyManager, I'd rather use my own dialplan to prompt the user for a ten-digit number if they called while blocking CID. This code does prompt the user, but 1) hangs up if the user didn't type the ten digits before the timeout 2) if the user did type the right number of digits, it still hangs up instead of Returning and then jumping forth to the "cid"

Hi all, A working search using ldapsearch on some object containing parenthesis in attribute's value: ------------------------------------------------------------------------------------------------- ldapsearch -Y GSSAPI -h dc200 -b 'DC=ad,DC=domain,DC=tld' -s sub 'CN=CID 85 \(Join\)' dn SASL/GSSAPI authentication started SASL username: administrator at AD.DOMAIN.TLD SASL SSF: