similar to: Static problems with Asterisk + Polycom phones

Hi, i can't compile the latest svn update from zaptel: /lib/modules/2.6.14-1.1653_FC4smp/build make -C /lib/modules/2.6.14-1.1653_FC4smp/build SUBDIRS=/usr/src/zaptel modules make[1]: Entering directory `/usr/src/kernels/2.6.14-1.1653_FC4-smp-i686' CC [M] /usr/src/zaptel/zaptel.o /usr/src/zaptel/zaptel.c:6193:5: warning: "CONFIG_ZAPATA_DEBUG" is not defined

Hi-- 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

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 the problem getting the Polycom SP 501 to send *any*

i have ordered 500s from tritechcoa.com several times over the past 4 months. great service and delivery, and the prices are lowish, only problem is, they add a $20 handling fee per phone, on top of phone price, and shipping, making the lower price not as good -----Original Message----- From: Mojo with Horan & Company, LLC [mailto:mojo@horanappraisals.com] Sent: Friday, July 15, 2005 12:01

Hi, I'm using Asterisk@home and am having trouble using the conference bridge that comes built in. We're using Polycom phones. When we transfer the first person into the conference room (e.g. 8101) , they get into the room fine. When we try to transfer a second person into the conference room, they get dropped as soon as we finish the transfer. This is using Polycom SoundPoint 301

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

Hi list! I have been experiencing a lot of mystical hangs on my box. At first I suspected bad memory or motherboard but I found out that the mystical hangs disappear when I do not load the drivers for my x100p. The box will only hang under high load (copy operations with 30Mb/sec to a raid array in the box for example). With little or no load the box will stay up forever. (The imposed load

Hi, i just want to know whether there are any alternative file systems for HP SFS. I heard that there is Cluster Gateway from Polyserve. Can anybody plz help me in finding more abt this Cluster Gateway. Thanks and Regards, Ashok Bharat -----Original Message----- From: lustre-discuss-bounces at lists.lustre.org on behalf of lustre-discuss-request at lists.lustre.org Sent: Tue 2/12/2008 3:18 AM

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

This page http://www.freecall.com/en/index.html is advertising free calls to: Argentina, Australia, Austria, Belgium, Canada, Czech Republic, Denmark, France, Germany, Hong Kong (+mobile), Hungary, Ireland, Italy, Luxembourg, Malaysia, Monaco, Netherlands, New Zealand, Norway, Panama, Poland, Portugal, Puerto Rico (+mobile), Russian Federation, Singapore, Slovenia, South Korea, Spain, Sweden,

Recently I started getting weird errors. All of a sudden I see the harddrive of my box glowing steadily and syslog is getting filled (and the cpu hogged) with these errors: It seems to start with this line: Jan 1 03:28:04 kernel: FXO PCI Master abort Jan 1 03:28:04 last message repeated 12 times Jan 1 03:28:06 kernel: CI Master abort Jan 1 03:28:06 kernel: FXO PCI Master abort and seems to

Hi there, I'm experiencing an echo problem and dammed If I can sort it out. We're running Asterisk on Fedora Core 3 64bit, installed as per http://www.voip-info.org/wiki-Asterisk+Fedora+Core+3. These are the specs of the Machine ? 1 x AMD A64/3500+ CPU: Desktop Athlon64? Retail w/fan SKT 1 x Asus A8N-SLI Deluxe Athlon? 64 S939 NVIDIA nForce(r)4 SLI? PCI Express Req: 24pin ATX 1 x

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

Hi! I don;t know what i am doing wrong because i have still Low ID on aMule. I have action.AllowaMule and accept tcp 4662:4771 and udp 4672. Thanks, Mitja

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, --

[This email is either empty or too large to be displayed at this time]

For those newbies who seem to not know google exists, I've setup a searchable forum located at http://asterisk.keystreams.com/ . Yes some of the threads are doubled up, but that sort of conversion is not perfect so just use the search feature. It currently has 2002,2003, 2004, and Jan/Feb of 2005. It will *not* be updated in real time (at least not for now) Please direct