similar to: Auto Answer BEEP

I have the auto answer working fine on my 7960 phones on line 2, but when line 2 answers, it does not make any sound until the call begins speaking. I would like to have the phone beep when it answers so the person being called knows the speaker phone is in use. This would be much like a overhead paging systems that plays a tone before an announcement is made. How can I have asterisk answer the

Hi , I'm having some problems with functions enable auto answer in some Grandstream GXP 1405 , I have enabled this feature in the snom 821 phone and work gr8 , in the gandstream not work, I enable the function on the phone "Allow Auto Answer by Call-Info: yes Dialplan: exten => 501,1,SIPAddHeader(Call-Info: answer-after=2) exten => 501,n,Page(SIP/140&SIP/110,d) exten

Hi, When implementing click2dial feature, I can trigger an Aastra phone to auto-answer using statement like : SIPAddHeader(Alert-Info: info=alert-autoanswer); This is very convenient when trying to reach a distant party (ie through PSTN) The trouble is when 2 Aastra are calling each other over the LAN, this single statement is memorized somehow and both phones (caller and callee) auto-answer.

I'm using a mix of Cisco 7960, Linksys SPA-942, Cisco 7961, Cisco 7970 phones in a paging group. I have all the phones set up with an extra line that auto answers the dial from my paging extension when the primary line is not in use. All of these are operating correctly however the 7961/7970s all ring once and then auto answer so the person paging all the phones has the first part of his

Does anyone know of a way to make an originate request coming over the management interface (e.g. AstTapi click-to-dial) include the relevant Alert-Info SIP headers to enable the originating phone to auto-answer? I've tried setting up a custom context (see below), but the dial plan is only entered AFTER the originating call is answered, so the SIP header is added to the terminating call leg,

Is there anyway to turn off the call waiting beep in the grandstream and/or cisco ata186? I have a dial statement in my extensions.conf that rings 5 phones at the same time by combining them with the & in the dial statement. i.e.) exten => blah,blah,Dial(SIP/GS1&SIP/GS2&SIP/GS3&SIP/ata186a&SIP/ata186b,25,t) If one of those lines is being used, then the user gets a really

Hello, I have some trouble with the BLF indicator. I have two phones that use the same hint: 13 => hint,1,SIP/phone13&SIP/phone13-wlan This works great from the asterisk side, but it seems the status change is too quick for the attached Grandstream-phones. When I ring the extension the hint changes to "Ringing". The Grandstream blinks. Great. Now, when someone picks up one of

>> So I guess the problem is in my config for the phone? Or maybe >> asterisk >> has to send "alert-info" more than just once? Does anybody have this >> auto-answer config working reliably on a Polycom phone? >> >> Thanks! >> Noah > > Noah, > > Please see my Polycom config files at > http://www.kriscompanies.com/modules.php?

Greetings all I was wondering if any of you have found this problem. I have a setup with 80 GXP-2000s which work great until I decide to stop now my asterisk server. When it comes back up half of the phones (not always the same ones) fail to register with Asterisk. I set up the phones to register and send keep alives (register every 4 minutes, send keep alives every 20 seconds) to no avail.

Having scoured the web, I still am no better off .. I have 2 Aastra480i's , and 120+ cisco 7940's :) . I am trying to decide which model to use going forward when we purchase more kit. They both seem much on a par regarding features. Q1: Is there anyway of making the cisco auto-answer _without_ having to manually edit the configuration on each phone ? I've been able to get the

I have my Asterisk server all setup. But have an odd problem and hope someone here can help. I have a Polycom IP 300, a Grandstream GXP-2000, and an installation of X-Lite. They can each call each other just fine (extension-to-extension). I can also dial-in from the outside (via Broadvoice) and can leave and retrieve voicemails. When I set ANY of the extensions (clients mentioned above) to

Hi People, We use the Grandstream GXP-2000 phones, firmware 1.1.1.14, Asterisk PBX, Slackware Linux 10.2, loaded on a Intel(R) Pentium(R) 4 CPU 2.66GHz Box... The issues that we are experiencing involves our Telephone Operator's/Receptionist whom answer multiple incoming calls... As an example.., when they answer line 1 and Line 2 starts to ring they would ask the person on line 1 to

Check your FW-1 tracker and see if any sip packets are dropped during call initiation. I had this problem and it went away when I upgraded the BT's firmware to the latest (16). Beware, though, that people on the list claim that this firmware breaks functionality of the message button and autoanswer. I haven't checked this yet, cause I can't afford to go back a version. I prefer a

I started out a few years ago with some SPA-841 sets, because the Grandstream 2000 I thought I wanted was perpetually delayed. The GS had more call appearances, and I didn't want just the 4 max that the SPA offered. As it turns out, with the greater flexibility of VOIP, I don't need 'dedicated' CAs the way I needed them on ISDN previously, so 4 is actually adequate. Along the line,

Hi When I make a call with the Grandstream GXP-2000 through Asterisk (and SER) to landline using VSP, after I hang up the call the other party are still connected for another 30-40 seconds. I've notice that the SIP BYE is sent to Asterisk, but Asterisk sends no SIP BYE on to VSP. When I use the SPA-941 the call terminates on the other right away soon as I hang up. I have updated the

Hi list, I'm having problems transferring certain calls made by the attendant between the PSTN and to an internal extension. Although, transfers between the majority of the calls ends successfully. Debugin this, I've found that calls made to certain "numbers" (Telephony Providers), aren't detected as ANSWERED in the CDR, so they are not properly accounted (for billing),

Hello All, I've asked this on the asterisk-dev list, so sorry for cross-posting. So far I'm not sure how to accomplish this without looking at the source code or looking at some other way to get around this issue. I'm trying to have an automated call to an Aastra SIP phone and have the call auto-answeredby the phone. I know that a SIP call placed to the phone can be auto-answered if

Hi I have a grandstream gxp-2000 and trying it on an asterisk 1.6. When I call internally between extensions I can hear the other person in the gxp2000, but when I call externally from the gxp I can't hear the person on the other end, but he can hear me. How do you configure the grandstream 2000 to work on asterisk 1.6? Regards /ralf ________________________________________________ Ralf

As far as I can remember I only opened sip and tftp ports for the phone. For some reason (didn't look into it too much) the call stays with sip and doesn't use RTP. The problem you describe (the call doesn't even ring on the other side) is something I had and was solved by upgrading the firmware. Checkpoint's tracker explicitly said what connection attempts were blocked and why.

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I've got Asterisk SVN-trunk-r9059 currently running on Fedora Core 4 w/ 2 eyebeam softphones and 2 Grandstream GXP-2000. At my desk I've got the grandstream and the GXP-2000 I would like to ring both. Using macro-exten-vm and dialparties.agi Macro(exten-vm,200,200-202) the caller is sent to the unavailable voicemail but if I use