search for: 501s

...hought with QoS and an MPLS, I could eliminate the remote PBXs (and not have to buy another for the new locations) by simply using the MPLS to tie everything together. See below. /--------------------\ | CORP HEADQUARTERS |----> PRI (23 Channels, 100 DIDs) | Asterisk PBX | | 30 Polycom 501s | \--------------------/ | | < 3Mbps MPLS (2 T1s) | /--------------------\ | MPLS "Cloud" | \--------------------/ | | | | | \---SIP over MPLS 1.5Mbps T1---Branch Office 1 (5 Polycom 501s) | | | \-------...

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 a...

...QDN 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 application version running on the SoundPoint 501s is 1.6.7.0098 TIA Paolo

When I want to pick up a ringing line, I dial *8 and hit New Call softkey on my Poly 501. For some reason, if I pick up the hand set and dial *8, it seems to ignore or drop the 8 digit. I've confirmed that this happens with all of my 12 Polycom 501s. Does anyone know what would cause this or how to fix it? Craig Bruenderman Network Advocates, Inc. 300 Envoy Circle Suite 300 Louisville, KY 40299 Main: 502-412-1050 DID: 502-992-5929 Fax: 502-412-1058 Mobile: 502-548-1100

Hello - I have 8 polycom 501s all setup great using ulaw. We have put them through a pretty rigorous torture over the last 4 months, and they've performed famously. No dropped calls ever. We invested in some g729 licenses. changed my ipmid.cfg so that g729 is priority 1 and ulaw is priority 2. I added allow=g729 to...

Asterisk 1.2.13 - Evolution PBX from Intuitive Voice Technologies We have an installation of 35 SIP phones (Polycom 501) and one receptionist phone (Polycom 601). I have 15 of the 501s set up to accept a "Page". From what I understand, the "Page" is done using the asterisk page application that throws the extensions into a conference room and then set the originating caller to the only one who can talk. The problem I am having is about 1 out of 25 pages will...

...-- many of the stock Linux FTP servers will give the exact problem you discussed, below. I should know -- took me almost a week before trying proftpd, and WHAMMO, worked like a champ. -Ken On Tue, March 7, 2006 12:37 pm, William M Conlon wrote: > I spent a weekend battling similar issues with 501s, using FC4/ > proftpd. I finally switched from FTP to HTTP. > > > On Mar 7, 2006, at 9:53 AM, Kristian Kielhofner wrote: > > >> Hello everyone, >> >> >> Please forgive the exclamation points but I have been battling >> this one off and on for about f...

I have an Asterisk 1.2 install running on RedHat 9. I have a bunch of Polycom 501s co-locacted in the same building as *, and some more 501s in satellite offices (also registered to my * server) . Finally I have some road warriors running XLites. Ideally when a road warrior (XLite) calls a satellite office (Polycom 501), I'd like to avoid having Asterisk in the media path....

...by name and escape. I had set ext 500 as exten => 500,1,VoiceMailMain(${CALLERID(number)}@default|s) but now that the contexts are different. this does not work #1 how do I have everyone use an ext to get the voicemail regardless of context. #2 can I get the mail buttons to work on my polycom 501s and swissphones #3 where do I put the "i" ext to allow the caller to go from the voicemail back to a ext in the dialplan Thanks Jason

I have two asterisk boxes (1.2.14) connected via SIP with Polycom 501s registered to each. I set callerid name and num before sending the call from one box to another but the phone registered to the receiving server only properly shows the caller name, not the number. The number on the phone always shows as the name of the sip registration of the "calling"...

Howdy, Working on some niche requests from one of my hotel clients. asterisk 1.4.20-1 on CentOS, Polycom 501s. The first request is for the Polycom's screen to show the CID of the inbound caller when a call pick is executed, so the picker knows if the call is internal or external. I have already "worked around" this issue by using ALERT info to give seperate ring tones for outside and in...

Has anyone been able to get the IP501 to discover the FTP server IP address (via dhcp or dns) and download 100% of the config from a provisioning server? We are still having to touch each unit to enter the ftp server address and password, as well as set many of the options that will not take from the config file. Have a sample config file you are willing to share? What is required in

...Is memory leak still as much an issue with 1.2.7 versus 1.2.5? In other words, is it worth it to upgrade a working, memory-leaking 1.2.5 to 1.2.7 or 1.2.8 just to potentially encounter other bugs in the new versions? Have other people been satisfied with the new versions so far? I have Polycom 501s and 301s. Call transfers are prone to crashing the system, getting sent to the wrong phone, etc. Is there some sort of rollback function? I'm considering having a second PBX box for the upgraded version, then keeping the working production system as a backup. My PSTN providers are voipjet (...

Hello everyone, Please forgive the exclamation points but I have been battling this one off and on for about four days now. Sorry for the cross post. It all started with a box of IP 501s. I contacted my reseller and obtained the latest BootRom and SIP firmware. Unzipped, configured, copied over to my FTP server (running AstLinux, of course). The phone booted, so far so good. Updated bootrom, nice. Rebooted again. Updated sip firmware. Also nice. Upon the next reboot, th...

I've just got a batch of 301s and 501s in and am trying to get them to work. I'd like to manually configure everything via FTP rather than the web or phone interfaces, but I can't seem to find a good source of definitions for all the options in the sip.cfg or phoneX.cfg files. Anyone know of any? Also, I'm having quite...

...To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] seems-to-be-inexpensive source of polycom 301 and501 Hello, sorry if this kind of a price really isn't news to you all, but it seemed really good to me. www.tritechcoa.com has the 301s for $114.20 and the 501s for $171.29. I ordered them on the 11th and they arrived today, on the 15th, but I'm in Alaska so I call that _good_ order response. Mojo _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/aste...

...nightly and update the timestamp so asterisk knows to update its values. Has anyone tried anything like this? I would like the phones to regrab their spot once the softphone is logged out. We have a Asterisk box (gentoo linux) which is running realtime (mysql 5). Our phones are Polycom SoundPoint 501s and the softphone is xlite (windows). Rob

I am using Polycom 501s with asterisk 1.2.4. When transfering to call parking wih "#1" -> 700 the user is able to hear asterisk tell him what extension the call was parked on. However, when I press "transfer" -> blind -> 700 . The user is not able to hear what extension the call was p...

...e just gone live with asterisk in a new office with approx 40 Polycom 501 handsets. I have a few questions: 1) Call Parking: I am able to park calls using the standard Asterisk call parking system (transfer to ext *70 etc...) I would like to make this a little easier for my users The Polycom 501s w/ SIP 1.5.2 seem to support some type of standard call parking, however I don't think it works with Asterisk. Is this true? Is there a way to integrate the to call parking system etc? 1a) If I can't use the Polycom built-in call park feature, is there a way to remap one of the butto...

Hi all, I have to buy some IP phones. Previously I have used Grandstream GXP-2000, Budgetone 101 and Linksys SPA-841. I always had problems with sound quality with all of them, and I was always of the opinion that it were the phones which were not good. In GXP-2000 deployment of about 50 phones, some work good, some have sound problems like words missing, clicking sounds when talking, and some