similar to: How to get a beep on a Auto Answer Intercom - Cisco 7960

I have been able to upgrade my Cisco 7905G phones to the SIP Image without any problems, but I just got a 7960, and I can't seem to get to the settings so I can upgrade to a SIP Image. When the phone boots up, it says Configuring VLAN, Configuring IP, TFTP ????..., then Protocol Application Invalid. I noticed on the wiki page "Firmware issues on 7940 - 7960"

I've been digging around with some cisco engineers for about a week & I finally got an encouraging response to the Auto-Answer issue with the SIP Phones. Here is their reply: =============== == FROM CISCO == =============== Auto-Answer feature is introduced in SIP IP Phone 6.0 version. This software version is expected to be available for customers shortly. Please let me know if you

I've just received a couple of the Grandstream GXP-2000 enterprise phones for evaluation. When a line on the phone is configured for auto answer, it connects silently. Has anyone been successful in havein a "beep" sound played to alert the user that he has an autoanswer call? Thanks Bill

OK, I got my Cisco 7960's to auto-answer on the second line but I can't get the Asterisk to call all the lines at one time. I have 4 phones I would like all of then to answer when I dial x300. Any help would be great Thanks Tuska extensions.conf [conference] exten => 300,1,AGI(callall) exten => 300,2,MeetMe(300,dtqp) ; press # to exit the conference exten =>

How would I go about setting this up. I have a few 7960's with an extension set to autoanswer. How do I let all extensions answer and be active? Thanks, Will -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031110/f17a2971/attachment.htm

I have a Cisco 7960 phone. I cannot seem to use the settings button to get into the phone to change the TFTP server. I've set up a DHCP Server, TFTP Server with the same address, and the phone requests the address of 0.0.0.0, the server offers the address of 192.168.2.2, but the phone seems not to take it. I have no action on the TFTP side. So, since I can't seem to server the phone

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

A friend pointed out this url http://www.cisco.com/univercd/cc/td/doc/pcat/clmn32.htm where it lists intercom/auto-answer as being a feature in Cisco Call Manager (which as I understand it, uses SIP predominately for handsets). I've come across comment somewhere that intercom isn't supported in the SIP spec. Does anyone know if the apparent capability of Intercom being available in SIP

I would like to know if it is possible to slow down the dialing process in asterisk. I have 4 of my 8 phone lines that are VoDSL. When we try and dial out these 4 VoDSL Lines, the number is often miss dialed, or incomplete. I added a wait before Asterisk tries to dial the whole number, but that has not solved my problem. If I use a regular phone and dial out these lines, they work fine. My

I have a few Cisco 7905G phones and I having a little trouble configuring them. They are working with Asterisk. I'm able to get the sip image loaded, but I can't get the phones to blind transfer. Does the Cisco 7905G Phone use XML Services? If you are using the 7905G phone, would you post any of your configuration files so I can try and figure out where I'm going wrong? Thanks for

There are no cordless devices in this situation. :) Jerry Geis wrote: >/ I have a normal setup of calls coming in on analog lines (4 of them) />/ coming into />/ an old KX1232 pbx. I have those lines forwarded to a T1 card in the />/ KX1232 />/ going over to my T1 card, it then uses IAX going over to my REAL />/ asterisk pbx. />/ />/ (these steps are there for testing

I have a normal setup of calls coming in on analog lines (4 of them) coming into an old KX1232 pbx. I have those lines forwarded to a T1 card in the KX1232 going over to my T1 card, it then uses IAX going over to my REAL asterisk pbx. (these steps are there for testing and other items) Anyway after a 10 minute call asterisk gives a faint beep, beep, beep and then hangs up the line. Is there

I read about this intercom stuff on page 62 & 63 of the book "Developing Cisco IP Phone Services" isbn 1-58705-060-9. Primary calls take place on streaming channel 0. When streaming channel 0 is not in use, streaming channel 1 can be used for asynchronously streaming (in and out) stuff like voicemail, email, and, yep the one we want, intercom. Page 87-88 of the book talks about

Guys. Anybody using sipuras 2002 knows if there is a way to make the phones connected to it to autodial an extension when the phone is picked up? For example, if the phone is on a police booth (building entrance) and you want the guys to just pick up the phone and make the phone auto dial the receptionist extension without the guys having to dial anything (ala batphone). Is this possible with

Hello There, I would like to play a busy tone (ie BEEP BEEP BEEP) when the underline carrier sends back 486 Busy Here. Looking at Dial parameters ( http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial), it mentioned something about the r parameter as not being very professional or something like that... Then there was: U(x): Executes, via gosub, routine x on the called channel. This is similar

This is really doing my head in! For some reason, my asterisk box can't playback beep.wav. I have this extension defined in my internal context: '10001' => 1. Answer() [pbx_config] 2. Wait(2) [pbx_config] 3. Record(/tmp/asterisk/10001:gsm) [pbx_config]

Hi, [I'm Using redhat linux 7.1 with R-1.4.1 rpms.] Last summer, Henrik Bengtsson wrote to R-help: > Is there a way to turn of the (annoying) beep that occurs when one calls the > locator() command and clicks the mouse. Try > > plot(0) > locator(n=10) > > I am running Rterm on WindowsMe/Cygwin bash and I don't know if this is the > case on other platforms.

Does anyone have a audio file they would be willing to share for on hold music? I am looking for something like the old norstar beep every few seconds. I tried 3 seconds silence, beep.wav, beep.wav. But it just didn't sound right. I need one that has a "softer" beep. Thanks! -- *** Forrest Beck IAXTEL: 17002871718 jonforrest.beck at gmail.com http://www.shift8.biz

Hi all. When a user transfers a call by pressing the "transfer" soft button on their phone, I'd like it to "beep" at them when the transfer is complete. I've got it turned on in features.conf: xfersound = beep ; to indicate an attended transfer is complete xferfailsound = beeperr ; to indicate a failed transfer However, it seems that

Hello; I am using vicidial which is using asterisk 1.8, mean while when the extension has voicemail, I always see the red light on the Polycom and hear the beep sound (toot toot) in period time. Also, I can see at the LCD an option to select it for accessing the voicemail ?but I am facing the following problems: 1) The red light and the beep: How I can let the Phone only have the red light