similar to: Polycom 501 Multiple Line Instances

Hi folks, New to the list. Just updated the bootrom and app firmware on a Soundpoint IP 501 as per: http://www.voip-info.org/wiki-Polycom+Phones Updated from: to: APP 1.4.1.0040 1.5.2.0054 BootROM 2.6.1.0003 2.6.2.0032 After I did this, it appears that the Web interface for the phone won't change the settings, nor will it actually reboot the phone now. What do I

I'm testing out some IP501 phones and I ran into an issue. WHen I try to add a new contact into the directory, I am not able to. A window blinks really fast but the entry isn't saved. When you exit the Contact Directory system you get a 'Busy! Please try again' window. What the heck could be going on? -- Jesse Keating GameHouse -- Systems Engineer

I've got a problem w/ zap calls being dropped right after they are answered. I have a log file: http://pastebin.com/368526 Everything looks OK except for the DEBUG[25563] chan_zap.c: Exception on 9, channel 1 that seems to come up quite often. As soon as the other end of the Zap answers (my cell phone), and I can even hear a half second of noise, the line goes dead and gets hungup. In

I've got an issue w/ 1.2.0beta1, where if I call VoiceMailMain from a sip phone, and then either put in incorrect passwords or just hang up, I never get a Spawn Extension that hangs up the call, and my sip phone is not capable of making any more calls until I restart the daemon. Can anybody help me fix this? -- Jesse Keating GameHouse -- Systems Engineer

When I hang up in voicemail, Asterisk seems to stop responding. (hangup vs pressing # to disconnect). After that, no calls can be made until I restart Asterisk. In IRC, a developer seemed to think it had to do with me using switch => in my dial plan. Basically I never see the calling extension get the -1 signal. Can somebody help me figure out why this is happening and how I can fix it

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'm attempting to set up my SoundPoint 501 with my Asterisk server. I've configured DHCP and TFTP and successfully updated both the BootRom and SIP application. I've also created a custom cfg file for this phone's MAC address and the settings seem to be taking just fine. I can see that the phone registers with my Asterisk server but 'sip show peers' reports that the phone

Hi I am looking for a good wirless headset to use with the Polycom Soundpoint 501 phone. I would greatly appreciate hearing from anyone with good experiences with a specific device. Thanks -- Jim Freeze

Hello, Sorry, I know I read this somewhere but now I can't find it when I need it. I'd like to force a call to go out one line if we dial '9' first and then the number. Same for '8' only I will force it out a different line. There is a parameter or a method to allow the dialtone to come back after pressing the first 9... but I can't remember how to do it. Anyone know?

Hello, Can anyone point me in the direction of software to monitor channel usage on voice T1s? Using a TE410. The wiki documentation seems geared to SIP channel usage.... Thanks Charles -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051004/8450d0c2/attachment.htm

Here is what I had to change on the phone1.cfg file: I had this value in my 1.6.7 file, put in there following suggestions from the Wiki (http://www.voip-info.org/wiki/index.php?page=Polycom+Soundpoint+IP+501) : reg.1.server.1.expires="30" Now, this worked flawlessly with 1.6.7. But with 2.x, this seizes up the phone with a huge CPU load (approaching 100% at times) and makes it

I have installed Asterisk and i have configured with two SJPhones; i am able to make calls between these two phones.I am planning to develop a application basically web based application from which the administrator able to trace the call logs or call summary, i mean from which user agent to user agent call is going , and what is the staus, if second user tranfers the call to the third

I like the Plantronics CS55/HL10, it's a DECT Wireless boom headset with a lifter kit for the phone, works like a charm, great range. -Cory -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Jim Freeze Sent: Wednesday, October 25, 2006 2:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion

Hi, I'm trying to configure a Polycom IP Phone SoundPoint 500 to connect it to my Asterisk PBX but with no success. First of all, I downloaded the SoundPoint IP SIP Administration guide I found on internet and then I tried to make a boot server creating an FTP account on my Mandrake 9.1 Linux box but I needed the following files: 000000000000.cfg sip.cfg phone1.cfg ipmid.cfg sip.ld so I

I have a polycom soundpoint ip 501 phone I was having problems getting some of the functionality working like the **number to pick up calls to other phones then it just started working (when it couldn't update itself for a while) Unfortunately then I got that issue sorted and it updated to the latest firware and now it doesn't go again. Is there something in the config files themselves

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

Is there a way I can have "voice mail check" calls coming from my internal users automatically get to the right extension, without having the user enter their extension? I'm thinking that I could have the local SPA boxes translate, or have each user live in a context where the extension in question exists uniquely per user, but both of these seem kludgey. Thanks in advance for

Problem: Polycom SoundPoint IP phones (running SIP) ceases to send DTMF that Asterisk can detect and use. It worked in 1.0.5, but has not worked since. This has been verified on SoundPoint IP 300's and SoundPoint IP 600's. Workaround: It used to be that for DTMF to work, I had to set the mode in sip.conf to "inband". Without making any configuration changes on the

I'm replacing a SoundPoint IP 600 with a SoundPoint IP 650. I attempted to simply reuse the existing config files for the old phone on the new phone, but the new phone would lock up on the 4th digit when attempted to dial out certain numbers. So, I downloaded the newest firmware and config templates from Polycom, and attempted to migrate the settings. Seems I'm missing something from

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