similar to: Sip phone displaying caller name while on call

Hi, 2013/4/19 Olivier <oza_4h07 at yahoo.fr> > Hello, > I've just realized that several phones display both caller name and number > while ringing but when on call, caller name is not displayed anymore. > Could you recommend a sip phone that still displays caller name during > phone call ? > Regards > I've been testing Aastra 6757i SIP phone and it appears

Hello list: I'm not sure what is going on, but I am using: asterisk -stable cvs (november cvs download) SIP channel Cisco ATA186 Zaptel 4-port PRI for PSTN Caller ID is enabled on the cisco ATA and seems to work fine. We do not get caller name at this time over the PRI ... Caller name works fine VoIP to VoIP and for PSTN calls, it displays the calling number like caller ID. But for some

I got most of the features of my phone working. Polycom TEch support refuses to help or even talk to me. So I'll have to ask here again. On incoming calls, only the NAME is displayed. I am trying to figure out how to get the NAME & NUMBER displayed. If anyone can help me do this it would be GREATLY appreciated. Thank you in advance

Stefan Gofferje wrote: > Hi folks, > > I have a rather nasty problem. I have set up an asterisk test system > with a Cisco phone, a X-Lite client and so on and did some testing. > To the developers: great work! Hell of great! > However, it seems to me like asterisk puts the Caller ID Number into the > SIP Display Name and Called ID into the Caller ID Number. That is kinda

Hi Setup : Asterisk 1.6 on Fedora Core 9 with TE410P.. 1. I;ve noticed that whenever during "background(menu-filename)" method - i try to press any key for selection like 1 for some prompt, 2 for another prompt etc...Asterisk takes a while before it takes me to the respective option..Is that normal behaviour ? by the time the caller waits to listen to the appropriate prompt on selecting

An additional follow-up question, if I need to set the P-Asserted-Identity on the create (originate), is there a way to do this with ARI? From: asterisk-users <asterisk-users-bounces at lists.digium.com> On Behalf Of Dan Cropp Sent: Friday, August 7, 2020 11:51 AM To: 'asterisk-users at lists.digium.com' <asterisk-users at lists.digium.com> Subject: [asterisk-users] With ARI,

Hi Jöran, Would it be possible to see an example using curl of how you are passing the PAI Header through ARI create? Dan From: asterisk-users <asterisk-users-bounces at lists.digium.com> On Behalf Of Jöran Vinzens Sent: Friday, August 7, 2020 12:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Subject: Re: [asterisk-users] With

Hello Folks, I am setting up a call center but we have few agents so one agent is able to handle calls of different languages and different queues. For the agent to identify the caller, I want a popup to appear as the phone starts to ring with the caller's number, language (selected in the IVR), Queue (sales, support etc) and any other information (e.g a URL with parameters) I can send this

Hi Dan, i did it wrong, sorry: curl -v -H "Content-Type: application/json" -u asterisk:asterisk -X POST " http://localhost:8088/ari/channels/newChannelId" <http://localhost:8088/ari/channels/1400609726.3/play?media=sound:hello-world> --data '{ "endpoint": "SIP/Alice", "variables": { "CALLERID(name)": "Alice" ,

I'm trying to transition from AMI to ARI. Running into a small hiccup when I try to create (originate a call) with the caller id name and number I can pass the Name and Number if the name has no spaces in it and it shows up in my PhonerLite application. curl -v -u asterisk:asterisk -X POST http://asterisk:astersk at localhost:8088/ari/channels/mycallerid.1?endpoint=PJSIP/1003 at

----- Original Message ----- > Message: 3 > Date: Thu, 16 Sep 2004 07:57:15 -0400 (EDT) > From: David Troy <dave@popvox.com> > Subject: Re: [Asterisk-Users] No Caller Name sent from Asterisk over > National or DMS100 PRI to a Norstar MICS? > snip> > > I have a PRI link up and running between Asterisk and a Nortel Norstar MICS > > v4.1 . I'm having a

I'll do that, but I'm sure that it's the Polycom :) Caller ID shows up fine on my 7960... W/ the name on top and the number below that. -- Executing NoOp("SIP/614-3ede", "Caller*ID is Matthew Marlowe <6092521155>") in new stack When the phone rings, only 'Matthew Marlowe' would display. When I answer, both the Name & Number will show.

Does anyone know how to change the display format of caller id on the screen of a polycom 300/500/600? When I call FROM my 'shop phone 203' TO my 'office phone 201', a Polycom 500, it only says 'Shop' as the calling party. More specifically, the two lines look like this: Incoming call from: Shop I'm looking for a way to make it use both lines for caller id,

hi, how to set caller ID for internal SIP users when dialing out on telco ISDN p2p (hfc card) line? I need to setup numbers from 0 - > 9 (10 sip users).internal caller id is working correctly .. from 0 to 9 ,but when I dial on isdn telco line -> gsm show only our prefix number ( xxxxxxY ) , only 6 (x) numbers of 7 (Y). here is extension 1 dialing on isdn line but on gsm is only 6

2010/10/24 Olivier <oza_4h07 at yahoo.fr> > > > 2010/10/14 Danny Nicholas <danny at debsinc.com> > >> ------------------------------ >> >> *From:* asterisk-users-bounces at lists.digium.com [mailto: >> asterisk-users-bounces at lists.digium.com] *On Behalf Of *Olivier >> >> *Sent:* Thursday, October 14, 2010 3:34 PM >> *To:*

All good information, thanks. However this is private network between Asterisk and a Norstar MICS about six feet away. So I'm holding both ends of the link. :-) > -----Original Message----- > From: David Troy [mailto:dave@popvox.com] > Sent: September 16, 2004 4:57 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] No Caller Name

Hi, I'm using Asterisk 1.2.1 on a Debian Sarge with a TDM400P and a monoBRI using chan-mISDN from beronet site. It seems to work all right except for autodial calls, monoBRI ISDN channel behaves differently waiting for the caller to answer and then continue. Asterisk console says: analog: -- Attempting call on Zap/2/3391818250 for 104@inbound_originate:1 (Retry 1) > Channel

Now that we have voicemail working, people have asked to be able to dial in externally and be able to access their voicemail. My dial plan is simple, after ringing a few extensions for some time, it goes to voicemail. What needs to happen to allow for someone to switch out of this into Voicemailmain in such a fashion that an external inbound caller wouldn't at least hear the option? Can the

>On Sat, 29 Jan 2005 12:53:11 -0600 > -----Original Message----- >From: <asterisk@draughon.org> >Subject: [Asterisk-Users] RE: Q: Can I over-ride the value of > ${CALLERIDNAME} ? >To: <asterisk-users@lists.digium.com> >Message-ID: <001a01c50633$d9e10a30$6701a8c0@calhoun> >Content-Type: text/plain; charset="us-ascii" > >Folks, > > Many

We recently switched my wife's business over to an Asterisk setup using Cisco IP phones (7940s and 7960s) with chan_sccp. They didn't use any kind of "office-style" phone system before, they had one phone in the office with a built in answering machine that would display the Caller ID of the person who left the message while playing the message. I know in the Asterisk