search for: 301s

...ry 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 (out) and...

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 hav...

...y, July 15, 2005 12:01 PM 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...

Hey all, I'm having problems where there is significant static when making SIP -> PSTN calls. SIP -> SIP and SIP -> VM calls are totally clear and fine. Here's the setup: Polycom 601,501, and ten 301s. Digum 2400 TDM card w/echo cancelling, 12 FXO ports. The TDM card is on IRQ 5 with nothing else on it. Server Specs: Asus P4P800E Deluxe P4 3.0 Ghz 1 GB Ram 80 GB SATA HD - There is no static when using a normal phone direct to the 66 block. - The sound is also a bit low, and bumping the volume...

...phones. Now when we use a Polycom SoundPoint 501 phone, we have an option to do a blind transfer when we transfer someone (appears the first time you hit the transfer button). When we do this, people stay transferred and everything works great. However ... we only have one 501 and the rest are 301s. So my questions are: * What's the difference between a blind transfer and a regular transfer? * Is it possible to do a blind transfer from a Polycom SoundPoint 301? * Is there another way to get this to work? Thanks! Jen

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

...about every 10 minutes which is really bad for a production machine. The only way to fix the hang is to reboot the server. My users are getting extremely impatient :-/ I see this on the clients- LustreError: 2814:0:(client.c:975:ptlrpc_expire_one_request()) @@@ timeout (sent at 1202756629, 301s ago) req at ffff8100af233600 x1796079/ t0 o6->data-OST0000_UUID at 192.168.64.71@o2ib:28 lens 336/336 ref 1 fl Rpc:/0/0 rc 0/-22 Lustre: data-OST0000-osc-ffff810139ce4800: Connection to service data- OST0000 via nid 192.168.64.71 at o2ib was lost; in progress operations using this service...

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

Where can I investigate the origin of brief silences during calls from/to my SIP phone? Only rare pauses of 0.5-1.0 sec when I'm not able to hear anything. Thanks Mimmus

I am having a problem with some Polycom 601 phones. If I dial without picking up the handset or selecting the speaker I can dial numbers that are any lenght. But if I pick up the handset or are using the speaker I can only dial numbers that are 8 digits. When I dial the 8th digit it dials immediately. Obviously this creates problems when I am dialing long distance numbers or anything that

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I have a customer that has two buildings that are connected with a fiber link. We have a single Asterisk server to cover both buildings. Now the customer went and bought an overhead paging system for the remote building and they want to integrate it with Asterisk. Is there a device that can connect over IP or an ATA that has an audio output port? The buildings are about 500 meters apart so we

on our cluster that has been running lustre for about 1 month. I have 1 MDT/MGS and 1 OSS with 2 OST''s. Our cluster uses all Gige and has about 608 nodes 1854 cores. We have allot of jobs that die, and/or go into high IO wait, strace shows processes stuck in fstat(). The big problem is (i think) I would like some feedback on it that of these 608 nodes 209 of them have in dmesg

I have a couple of Polycom phones, bootrom 2.5.0, SIP 1.3.1.0056. Works great with Asterisk when I power on the phone. However, after some time, say an hour, I cannot receive calls on this phone. On Asterisk, when I do "database show", it does show the phone in there, but it cannot reach the phone. Is there a way to keep alive the connection with Polycom? Any suggestions? Thanks, --

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