similar to: Short burst of static then disconnect

Hey all, I'll give my reseller a call for support in the morning, but I usually have better/faster luck on the list. I've got a SoundPoint IP500 that I upgraded to BootROM 2.6.2 and SIP image 1.5.2 on someone elses advice, I forgot to change out the old config for the new when I loaded the image up (I guess the config changed a bunch between 1.5.2 and 1.3.1) I was prompted with an error

Does anyone know where I can get replacement handsets for the Polycom SoundPoint IP phones? Or does anyone have any they want to sell? From the looks of it you have to buy a whole new phone to get a new handset. My vendor, TriaTechCOA, told me I had to buy a whole new phone to get a handset, which is pretty ridiculous. Maybe there is a more sane vendor I should be buying from? Thanks, -Ryan

I have a problem where if an agent's extension is busy and has voicemail the queue app will follow the dialplan and send the caller to an agents voicemail. This is really bad, because it takes the caller out of the queue when it hits that agent. But we also would like to have voicemail for some extensions like the shift managers etc. Is there s solution/workaround/patch? Thanks, -Ryan

Hi, my situation is: incoming call goes into the queue and is picked up by callback agent. The agent then wants to transfer the call to another device (another SIP phone). But 'transfer' button doesn't work and '#' button attempts to start channel monitor. Tried with both Queue(testq) and Queue(testq,tT). Is it meant as a feature that agents won't transfer calls at

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

Hello, I am new to Asterisk and new to this list. I got Asterisk setup and running using Asterisk@home, and purchased a PolyCom SoundPoint IP500 phone to test out. I cannot get the phone to talk to the Asterisk box. On bootup of the phone, it tells me that it cannot contact boot server. Why is that? It gets an IP fine, and I have also tried manually setting the IP of the phone and the Asterisk

For the past week or two, our customers who have Polycom phones have been experiencing a problem... but our customers with Cisco phones do not have this problem. The phones in question are: Polycom SoundPoint IP300 (firmware 1.3.1 or 1.3.4) Polycom SoundPoint IP500 (firmware 1.3.1 or 1.3.4) Cisco 7960 (firmware 7.2 or 7.3) The problem is this: when our Polycom users dial _some_ PSTN numbers,

Hi folks, I've got a very bizarre problem recurring when making calls with Polycom SoundPoint IP500 SIP phones and Asterisk. Sometimes when a call comes in to an IP500, one of the sides of the conversation is cut off (i.e. the caller can't hear the callee, or vice-versa). This isn't easily repeated, and rebooting the phone, or restarting Asterisk, doesn't seem to have an

We have Polycom IP500's, and just starting recently (after the broadvoice patch I might add) after about 1-2 days these phones ring, and answer, but we get no audio on the phones. The caller can hear us, but we cannot hear the caller. Its happened 4-5 times and is only intermittent. No errors on the console, using g.711u. Any ideas? Tim Jackson Network Engineer Angelina County, Texas

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I was wondering, most of the p2p programs are bandwith wasters, because they open lots of parallel connections. I have 5 queues to prioritize traffic, but these p2p open thousands of connections and my systems gets REALLY HIGH latence. Does anybody of you know by any means, for a DSL connections, the ammount of parallel connections for a good rate

Hello all, I've got myself into a bizzare situation that I can't seem to get myself out of... Was wondering if anyone had some advice that might get me 'over the hill' on this... Some background: PBX consists of an Asterisk box (running TrixBox), 4 Cisco 7960's, 2 Polycom IP500's, and now an additional Cisco 7960. The phones are all on a separate LAN. There is

I am trying to use a Polycom Soundpoint IP400 with my asterisk setup, and have been unable to find the proper firmware and application files to make it work. The phone can access the FTP server, and downloads the IP500/600 configuration files, but claims that the sip.ld file is larger than its file system. I can think of two possible ways to make this work. One is to find the IP400

>> Actually, computing the "power spectrum" for each frame of W shows >> how large an ammount of the original signal at time offset j the >> echo canceller thinks should be removed from the current input frame. > > Careful when looking at W because of how the real and imaginary parts > are packed in the array. Err. Ok, as I got it, 'bin 0' has it's

I just noticed something when I 'sip show peers' from the CLI, I get the following: 6113/6113 x.x.x.x D N 255.255.255.255 62927 OK (66 ms) 6112/6112 x.x.x.x D N 255.255.255.255 50079 OK (160 ms) 6111/6111 x.x.x.x D N 255.255.255.255 60810 OK (141 ms) 6109/6109 x.x.x.x D N 255.255.255.255 51331 OK (151 ms) All of those

Hi all, I just tagged acts_as_ferret 0.2.2 as the current stable version, so get it while it''s hot ;-) new features: - added support for the multiple models/single index approach. - find out the total number of search results by calling total_hits on the array returned by find_by_contents. fixes: - trac tickets #20 (find_by_contents breaks ferret sorting) and #24

Hi, I am new to asterisk , i am getting the following error,& the /etc/zaptel.conf file entry is as follows defaultzone=us loadzone=us span=1,1,0,esf,b8zs,yellow bchan=1-23 dchan=24 Parsing '/etc/asterisk/zapata.conf': Found Jul 1 18:33:35 WARNING[16384]: chan_zap.c:664 zt_open: Unable to specify channel 1: No such device or address Jul 1 18:33:35 ERROR[16384]: chan_zap.c:5296

Hi, I've had a small case with noise suppression and AGC. I have a fairly noisy environment here, and with the default parameters, noise suppression works fairly well while I talk. However, when I shut up, AGC starts slowly increasing the gain until it has amplified whatever noise is left to levels about equal to having no filtering at all. As soon as I talk, AGC backs down fairly quick

Hi, In this patch, I unify segment_32.h and segment_64.h into segment.h They have some common parts, but a considerable ammount of code still has to be around ifdefs. The only patch that is really important to paravirt is the first one, that moves a paravirt definition into the common header. The other two are just normal integration, and pretty much independent

Hi, In this patch, I unify segment_32.h and segment_64.h into segment.h They have some common parts, but a considerable ammount of code still has to be around ifdefs. The only patch that is really important to paravirt is the first one, that moves a paravirt definition into the common header. The other two are just normal integration, and pretty much independent

Hi, anyone out there with the same problems and a possible solution to the following? The functions callparking and hold use the same transferdigittimeout in features.conf. While I think 3 to 5 seconds are enough to let the user "find" their keys on the phone, the double ammount of time ( 2 x 5 secs) you have to wait before a call is parked and the parkposition is announced, is