similar to: Continuance on Polycom issue, not ringing

Hi, I have was testing some of the different ring types with my polycom 500, and the ALERT_INFO settings. Now when my phone receives a call it won't ring. All the other phones ring fine, and my phone wasn't the only one I rebooted with the changed sip.conf and impd.conf. I have reverted back to a standard sip.conf and impd.conf and I still can not get my phone to ring for any incoming

Andrei, Do you have X-Windows running on the linux box? I had a similar issue that was eliminated when I stopped this process and samba from running. Now samba is only allowed to come up during non-business hours, for changing BG music. Also, make sure your registration period in either (polycom) ipmd.cfg or sip.cfg is set to be at least the default 3600 time period. I also removed the

I'm still trying to work this out. I've got this in my sip.conf [1003polycom] type=peer secret=abc123 host=dynamic defaultip=192.168.1.215 context=default mailbox=1003 subscribecontext=phonestatus [1004polycom] type=peer secret=abc123 host=dynamic defaultip=192.168.1.214 context=default mailbox=1004 subscribecontext=phonestatus And this in my extensions.conf [phonestatus] exten =>

Has anyone seen an issue with SIP phone (polycom 500) dropping calls at irregular intervals with no errors in the asterisk log files? I am having this issue as described and it is a complete pain in my rear to trouble shoot because when I call my cell phone I can get a call to last over 30 minutes yet when I call another office that uses a standard pbx I can't get past 10 minutes. For some

asterisk-users-bounces@lists.digium.com wrote: > In the following setup: > call coming from a pstn line -> into FXO card -> asterisk -> SIP > phone > > i get an incredible loud echo in the SIP phone (about 0,5-1s) > (everything i speak into SIP phone microphone i hear in its > speaker). The person calling from PSTN is not getting any echo. Make sure you're not

Could someone please direct me (via personal email) to a provider with good prices on Polycom Soundpoint IP 500's with POE cables? I need 14 of them. Thanks, Adam ________________________________ Adam S. Robins Executive Vice President & CIO PHARMACENTRA, LLP 5901B Peachtree Dunwoody Road, Suite 380 Atlanta, GA 30328 Office: 770-395-0088 x34 Fax: 770-395-0989 Mobile:

Hi All - I'm running version SIP version 1.3.4 on various IP300, IP500, and IP600 Polycom phones. I'm having a tough time with MWI. I thought I remembered somebody on the list saying that they had it working, but I can't find it in the archives now. I have all the phones configured for MWI as specified in the WIKI: ipdmid.cfg: up.oneTouchVoiceMail="1"

Greetings, Recently there has been talk of the presence/buddy feature with asterisk and Polycom phones. I have it setup, and working as expected, however I can only get 7 buddies to appear on the screen at any given time. Has anyone gotten more than 7 buddies to appear? I'm just trying to find out if this is some polycom limitation, bug, or my error. Thanks, Matt -- Matt Gibson VOIP

I just set up a Polycom 500 on *. Every few calls I make, the call connects and the receiving party can hear me (thru Broadvoice), but I still get ringing on my end, as if they never picked up. * logs look just fine. Does any one have any suggestions? Thanks. ________________________________ Adam S. Robins Executive Vice President & CIO PHARMACENTRA, LLP 5901B Peachtree Dunwoody

Hello everyone. This is a message I've sent before on Sunday, no one replied so I'm reposting it (guess not everyone's at work 7/7) I've got this really annoying and beyond-my-knowledge-to-debug problem. The line connected to my FXO port gets marked "out of order" by my telco operator. I don't know how to explain this further. If I dial my own number from a

Hi all, I'm researching IP phones for a new office setup. We will need 30 phones. I have read the wiki and the polycom site for the phones, but I still had some questions about real world experience with these phones. -According to the documentation, the 500 series ( and 600, according to a polycom rep ) have built in hubs. Has anybody noticed performance issues in this setup, when

I have my * and polycom system setup to do Auto-Answer for internal SIP/Staff calls, and I am running into an issue with this and the polycom call transfer feature. * is seeing a new call come through from the polycom and is then transferring the call over. I need to know if there is some way I can grab a message from the SIP header or something to determine if I should not set the ALERT_INFO tag

Hi, We are in the process of moving from an Avaya Definity to Asterisk for our institution's phone system. I got one feature that the Avaya had, which I have not been able to reproduce with Polycom phones and Asterisk; since this feature seemed so small and useless to me when testing, I kind of ignored it. Now I am getting more "I miss that" requests than I expected. =) On the

Not sure if Polycom is changing their policy or if this is an accident, but you can actually download SIP 3.1.1 right from their web site. Anyone looking for firmware should get it now before it disappears. SIP app and release notes can be found here: http://www.polycom.com/usa/en/support/voice/soundpoint_ip/soundpoint_ip450.html -Dave

Let me add another variable into the mix...what about the Linksys SPA-962? Good, bad or otherwise? The 32 button sidecar seems like a deal at $80 street price. Michael On Fri, 22 Feb 2008 08:28:37 -0500, Matthew Brothers wrote: > >Michael Graves wrote: >> I need to add a few phones to an existing installation. They have a >> dozen IP430 at the moment. Does anyone feel that

Hey all, I've setup my asterisk install on a CentOS5 server, I've got a few IAX2 and SIP softphone clients connected on the same subnet and at least 1 external IAX2 softphone. However I'm having some difficulty getting the Polycom hardphone to function correctly. Watching the logs and debug trace it: - Registers correctly - Is able to make calls to other peers However it is not able

Hello, At one of our locations, we have started to see Polycom 501s (running 1.6.7 firmware) randomly reboot. We have taken packet traces of the phones to determine if there is something odd in the Layer 2 or 3 of the network that might cause it, and have not seen anything strange. There are no errors on the ports. This appears to be affecting POE powered as well as AC powered phones. The Polycom

Hello, I trying to configure asterisk to store sip and iax2 user in a mysql database. All goes well, but my problem is when i try to add a new user (sip or iax). I have look for an aplication with a web interface that lets us manage the user account in asterisk without success . How could I manage the users without making sql query throw the mysqlclient? Do you know any application that

Hi List, I've got * randomly hanging up on inbound or outbound calls on zap channels. I use a Digitnetworks X100P clone card. Any idea of what might be happening? Cheers, Jean-Michel.

Client - Not for Profit in the Middle of the Jungle/Rain Forrest Infrastructure - Datacenter is Non Climate Controlled, Prone to Flooding, and has Sketchy Power, LAN - NEW Cabling in main Office building, Hodge Podge of DYI wiring across remaining buildings. Phones - Total of about 50 extensions. Only about 25 - 30 phones will be IP phones, 20-30 more will have to be analog due to the distance.