similar to: Off Topic: Disable Polycom Soundpoint DoNotDisturb Feature

Well, yes and no. I tested that before and it causes a silent ring instead of a call rejection. I actually want to disable the entire feature. So the phone always rings unless you're actually on the phone. Thanks for the reply though! Regards, Jan ________________________________ Fr?n: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] F?r Jerry

Trying to install wanpipe 3.5.18. No errors during compile. But when I reach the point where wanpipe and dahdi_cfg is started, I encountered an error. Starting WAN Router... Loading WAN drivers: wanpipe done. Starting up device: wanpipe1 wanconfig: WAN device wanpipe1 driver load failed !! : ioctl(wanpipe1,ROUTER_SETUP) failed: : 22 - Invalid

Anyone here who has configured zaptel/dahdi for a singtel E1 line? What are the settings for coding, framing, line type and switchtype? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110210/4288ed84/attachment.htm>

I am currently testing a 57i unit. No problems configuring the phone's config via phone/web UI. We are trying to avoid using the web UI, the reason is it will take a long time typing the softkey xml applications URIs on each phone, so we chose TFTP. Tried configuring the phone via a TFTP config server, but no changes took effect. I wonder why it doesn't work with TFTP even if I was able

Also posting this question to people working on manager interface and dialers. I have a simple auto dialing script (using Originate) that forwards all incoming calls to a queue full of waiting agents instead of a meetme conference room. I use queues rather than meetme so I can leave the automatic call distribution to the queue itself. The problem is when the calls reach the agents, some of the

Hi, I'm looking to disable rejecting calls from my call center employees. They are using Polycom phones. Is there a way to either disable the reject/DND features on the Polycom phones (don`t think so) or have the Asterisk PBX ignore "Got SIP response 486 "Busy Here" back from 12.23.34.45" response from specific phones/SIP registrations and just keep on ringing?

Hi We have polycom 430,501 and 301 phones. Our customer does not need DND feature in any form. I can disable this feature from asterisk server but How can i disable this feature on phones. In the sip configuration file i found the parameter that change the phone behaviour during DND from busy to normal but still if the phone is in dnd mode the phone ringer would be off which is unacceptable.

Hello, Some of my users have phones lacking a DND button. I need to provide an extension they can dial that will put them in DND, i.e. tell the server not to send them any calls until they get off the DND. I've researched it for almost 3 days now and tried a range of configurations. I'm hoping somebody here has an answer. Currently, I have this in extensions.conf [app-dnd-on]

We're encountering no audio issues in ConfBridge. Only the moderator and the 1st invited have audio. When the 2nd invited number picks up the phone, only the announcement is heard, then followed by silence. This occurs frequently. Only 1 out of 5 test calls do not have this problem. We were hoping that switching from MeetMe to ConfBridge will solve the no audio issue, since there's no

Hi We've setup an Asterisk PBX recently and I encountered the following problem: When [mac address]-registration.cfg file includes the FQDN of the Asterisk PBX for the Polycom SoundPoint 501 phones it will not (even try to) register with the Asterisk PBX unless the DNS (it asks) successfully resolves the name: _sip._udp.[Asterisk FQDN]. Did this happen to anyone else? PS - The

I've got a couple Shoreline IP phones, their Shoreline model number is Shoreline IP 100. I believe this is actually a Polycom SoundPoint IP 300 phone. I believe the phone is using a MGCP stack. I want to use it for testing with Asterisk. 1. I suspect I need to re-image the phone to make it work with *. 2. How can I preserve the current image on the phone? 3. What is preferred image to use

Hi Folks; I just bought a Polycom SoundPoint 500 off of ebay after having spent way too much time trying to get updated sip images for our cisco phones. The phone I bought didn't have an AC power adapter; Could someone please tell me the volts & amps that the dc plug that comes with the phone puts out? Thanks! Mike

Hello all, I need the bootrom.ld file to set up some Polycoms I have Platform: Model=SoundPoint IP 330, Assembly=2345-12200-001 Rev=A I've publiched on my FTP files downloaded from http://support.polycom.com/PolycomService/support/us/support/voice/soundpoint_ip/soundpoint_ip330_320.html (3.2.3 combined and split zips) but my phones are still showing the message: "error, application is

I am looking for clues on how to configure distinctive ring for a PolyCom SoundPoint 300. Does ALERT_INFO apply? If so, how? Thanks, David Koski david.nospham@kosmosisland.com

I bought a couple Polycom Soundpoint 300's, and have them working nicely with SIP... but I'd like to be able to do automatic config via FTP, but it requires some XML config files. The docs discuss them in detail, but I can't seem to d/l them from Polycom. [No, it doesn't appear to be on the CD that comes with the phone.] I've created an account at Polycom's tech support

Hello everyone, I have just received 3 brand new Polycom SoundPoint IP 600 from "voipsupply.com" and I have the exact same problem on all of them. When I receive a call, the phone is ringing correctly but when I answer it, it takes exactly 10 seconds before I can hear the caller. I also have SoundPoint 300 and 301 and I don't have that problem with those. I'm using Asterisk

Chris Mason (Lists) wrote: > Cory Andrews wrote: >> >> >> >> >> IP430, will sit between the IP301 and IP501, IP430 will have dual >> Ethernet, PoE, and full duplex speakerphone. List price (MSRP) $239 >> street price should fall likely between IP301 and IP501. >> > That looks great, the 301 is almost useless due to the lack of speaker

Hello! I am selling hardware I have been using in various Asterisk test installations. If you are interested in buying all or any of the following items please reply off list. I got a few * X100P clones * Sirrix PCI4S0 ISDN BRI cards (+ 1 Sirrix S0 voltage/power supply) * 1 Polycom Soundpoint IP 500 (+ Plantronics headset) * Conrad HFC-S cards to give away. Some of the pieces are unused,

Does anyone have the Admin password for the phone in order to change configuration Roman

We recently took a few Polycom Soundpoint IP 500 to test out in Asterisk environment. So far it has been good. Call Hold, Transfer, DMTF etc. However, I do notice every now and then the Polycom fails to register with Asterisk. Asterisk console outputs the following: Feb 3 13:02:32 WARNING[278546]: chan_sip.c:2365 __transmit_response: Unable to determine sequence number from '' Feb 3