similar to: polycom soundpoint ip 300

I have an extension setup in my extensions.conf for hold music. ext. 600. If I pick up a phone (polycom soundpoint 300 sip) and dial extension 600 I hear the hold music playing. If I call another extension and pick it up and put the call on hold with the hold button on the phone I hear nothing at all. Does anyone have any experience with these phones and getting the hold button to work?

I've been googling and talking with Libretel about my setup and the fact that incoming calls to my asterisk box through the Libretel number reach my box (I hear the greeting being played) but then don't accept DTMF. Here is a rough diagram of my setup: Asterisk | server | NAT <------------ Libretel | router | Note that there are NO SIP

I'm evaluating using Polycom phones for our call center and I've set up my first phone (a SoundPoint 560) to give it a try. The phone is working and can successfully place and receive calls. But every minute, there's an error in the log file: chan_sip.c: Registration from '<sip:6193644850 at jtsd05>' failed for '192.168.200.99' - Username/auth name

When my SIP phones try to register with my asterisk box, this is what I get my log file: Mar 26 14:46:41 NOTICE[3896] chan_sip.c: Registration from '<sip:201@192.168.2.13>' failed for '192.168.3.2' - Not a local SIP domain In sip.conf I have this for my global settings: [general] context=from-sip ; Default context for incoming calls

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 am new to asterisk and need to dig up some info on how to set it all up. It looks a bit daunting especially all the options available in the .conf files. I have 2 SIP phones, GXP2000 and a budgettone 100. phone1 - 192.168.0.160/24 extension 1000 phone2 - 192.168.0.161/24 extension 1001 Server - 192.168.0.57 I get the following all the time, but can make calls between the 2 extensions,

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

I have a PolyCom Soundpoint 500 sip phone. I'm tring to get the phone registered on an asterisk box but am having no luck. I get the following errors 192.168.22.196 being the phone and 22.254 being the asterisk box.. Apr 23 11:41:33 NOTICE[1133742896]: chan_sip.c:5623 handle_request: Registration from '"110" <sip:192.168.22.196@192.168.22.254>' failed for

I have Polycom ip-300 phones that worked yesterday but dont seem to work today (at least dtmf signalling once connected to the asterisk box) The current configuration is: [general] port = 5060 bindaddr = 0.0.0.0 context = test srvlookup = yes dtmf = inband allow = all dtmfmode=inband progressinband=no disallow=all allow=ulaw pedantic=no [202] type=user secret=xxxx context=test mailbox=202

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 just got a Polycom soundpoint and I set it up using the phone and web based admin. I cant seem to figure out the config files and they are confusing me greatly and I dont have time for it :) Some things are odd, like on every reboot it seems the volume I set is reset? is there any way to fix that. And the ringer seems low. - Even all the way up Anyone willing to point out a good asterisk

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

Hi, I just got my ip500 back after months of waiting. Is there an easy way to get it hooked up to asterisk without [t]ftp servers and all that or is there a quickstart page somewhere? tia r

Hi I am just getting started with asterisk and trying out using the sample files with a Polycom IP500 (latest sip.ld etc) My question: phone status says not registered: /system status/server status "Line 1 poly is not registered" console message says: handle_request: Registration from '<sip:poly@10.1.20.3>' failed for '10.1.20.50' I have tried dialing and

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

List, I am having trouble with one of our IP600. Every five days or so, the phone locks up. This is the third 600 I have put in place. I am running asterisk 1.0.9. Has anyone had this problem with the IP600? This electronic message transmission, including attachments, is for the exclusive use of the individuals to which this e-mail is addressed and is to be reviewed and used exclusively

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

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

Has anyone been able to get the minibrowser on the Polycom SoundPoint IP 500/600 phones working? If so could you share the relevant sections of your config with me?

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