similar to: Notification when on the phone

On 2020-05-28 10:15, Doug Lytle wrote: > Everybody, > > I've had a request from my manager that I figure out how to get our Asterisk 13.x system using chan_sip to be able to display on the Polycom VVX series phone display (firmware 5.9.5), when an extension is called and the person on the other end is on the phone. > > He said, "Our old Analog phone system could do it,

>>> But if you've already got the caller on the phone, then you might consider the CONNECTEDLINE function in Asterisk... And that we don't. It's the third party that would like the notification the the destination phone is currently busy with another call. CONNECTEDLINE only functions after a channel has been answered. I was successful with using CONNECTEDLINE when issuing

Hi - Just wondering if anyone has gotten a Polycom VVX phone to successfully do an Auto Answer with asterisk. I have an older generation of Polycom phones that do this just fine, but I can't seem to make the VVX phones work. I tried the guide here: http://community.polycom.com/t5/VoIP/FAQ-How-can-I-change-my-Ringtone-or-Ring-in-a-special-manner-for/td-p/5167 And I have this in my diaplan:

    I am having a strange problem.  We have an Asterisk 16.12.0 server (we have upgraded at least two versions since we found the problem) where users complain that confbridge calls end after about 30 minutes or so.  The problem is that according to Asterisk the calls are still active.  All users are cut off at the same time but a "core show channels verbose" still shows channels as

On Mon, Mar 9, 2015 at 9:40 AM, David Wessell <david at ringfree.biz> wrote: > Welcome to our hell. > > We ran into this on VVX 300 and 400 phones running UCS 5.2.x. We finally > got Polycom to issue a hotfix firmware version. I'll be happy to share it > with you offlist, just email me. > > Officially Polycom will fix the issue in 5.3 in a few months.. > >

Hi all, When moving from a self compiled Asterisk 13.23.1 to Asterisk 13.24.0, has resulted in a MWI clearing delay of around 5 minutes. After listening to a voicemail and deleting it, the Polycom VVX 601's MWI light is left on for around five minutes, before clearing. Installing Asterisk 13.24.1 did not fix this. Moving back to 13.23.1 allows the MWI to clear immediately. I see a note in

Hi, Am running a freepbx install and created trunks, extensions and groups. Now I'd like to hand out the Asterisk phonebook to the phones (all VVX 310's). Is there an easy way to do this? Best, Stanley

We use Asterisk 14 to proxy calls between two servers, 10.0.0.192 to 10.0.0.228. But sometimes another of our servers becomes listed as a SIP agent, even though the server's IP address isn't part of our sip.conf, extensions.conf, nor any other config I know of. For example in the log snippet below, the source server experienced an SDP renegotiation in the middle of a call, and seemingly as

>> So which option is preferred? >> >> A) Have a softphone aor/auth_user/password for a particular human, and >> expect them to configure it on multiple devices. Do not worry that 1) >> multiple are registered at once (because that's normal in SIP) and 2) >> asterisk has no idea which is which (because the intent is to place a >> call to

When I set CONNECTEDLINE() info for an incoming ISDN call, the calling party sees only CONNECTEDLINE(num) and the name does not get displayed. Some time ago I called a number, where I did get back a name and a number and everything was displayed correctly. So I think the calling site should basically be able to handle all connected line info. Looking at a pcap trace of the D-channel data, I

Hello, Is there any equivalent of the CONNECTEDLINE function which can be called from an application using the AMI? Thanks for any ideas. -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 Australia: +61 (0) 2 8063 9019 -------------- next part -------------- An HTML attachment was scrubbed... URL:

>>> Nobody has any information or opinions on any of this? Personally, I don't think MACROS are going anywhere any time soon, so I have not bothered looking into a substitution. As for ael; I've never used it. Doug

Make all the static constant strings as char arrays, so they can be fully stored in read-only memory. --- align/scan.c | 2 +- builder/index-validate.c | 2 +- cat/cat.c | 2 +- cat/filesystems.c | 2 +- cat/log.c | 2 +- cat/ls.c | 2 +-

Hello, I recently gave CONNECTEDLINE function a try on an Asterisk 11 setup with a couple of SIP phones. When a SIP phone dials an other one, with a CONNECTEDLINE statement in its dialplan, I noticed that Asterisk update caller's information using a Remote-Party-ID header in 180 Ringing message. For instance: Alice ----------------> Asterisk ------------------->Bob ------- INVITE

Le 15/01/2020 à 19:50, C.Maj a écrit : > On 2020-01-15 11:24, Administrator wrote: > > 8<'s > >> One of the provider took a pcap and told us that expiration was set to 0 >> that's why they don't accept the registration. We took a pcap on our >> side when SIP packet goes out of our server and we see that the >> expiration parameter is setted to

Gentlemen, Did tryout "CONNECTEDLINE" function, was exactly what I have been looking for. But there are at least one thing I cant figure out. Did a very simple and "stupid" extension 0317998955 and ran a test. My phone (0317998975) dials 955, the display on my phone changes from "955" to "Connected Line 955" when my call is answered, shouldn't the

Hi Rich: I just send a patch to request for your comments, then will do further work about this. things like this: [root at Allen ~]# virt-sysprep --ipconfig="eth0:192.168.1.2,255.255.255.0,192.168.1.1" --enable=ipconfig -d clone-6u1 OR [root at Allen ~]# virt-sysprep -d clone-6u1 Please comments. Thanks -Wanlong Gao Signed-off-by: Wanlong Gao <gaowanlong at cn.fujitsu.com>

Hello! Just read http://www.voip-info.org/wiki/view/Asterisk+func+CONNECTEDLINE tried on dahdi, it works, i.e. if I call asterisk user from my pbx connected phone I see what I set in Set(CONNECTEDLINE(name)= But if I call the same user over h323 ( no matter is it asterisk with ooh323 or cisco gateway) I don't see this. Could you tell me is it possible? Thank you!

I have been racking my brain attempting to get the remote callerid information for calls made to extensions on another Asterisk system connected via IAX2 but nothing has worked. To clarify, I would like to display the number AND name on the calling phone when calling extensions on another Asterisk system. I seem to be able to 'send' all the information I want to the system I am calling but

Hi all, we face a strange behavior while connecting an Asterisk16 instance with PJSIP to 2 providers: we receive error 401 Unauthorized, both of them having Kamailio as front-end. With other providers -we don't know if they run kamailio- registration is just fine. One of the provider took a pcap and told us that expiration was set to 0 that's why they don't accept the