similar to: half channel audio after upgrade to 1.4.18

Hi all, I upgraded from asterisk 1.2.23 and zaptel 1.2.19 to asterisk 1.4.18 and zaptel 1.4.12.1 I use polycom 501 phones internally. Everything seems fine. I can pick up the phone and call out, calls coming in work just fine. The issue I see is when the system first calls me, then calls someone else. This works if its polycom to polycom. I hear audio full channel. If I do polycom to external

Hi, I am getting half channel audio on cisco 7960? Any idea why? Details below. Jerry This same phone has been working for MONTHS using a TDM2400p with no issues. Today we got a T1 installed coming into Box A with all incoming calls going to Box B (the TDM2400P box). The TDM2400 is no longer in use. All incoming and outgoing calls are done with the T1 and Box A and Box B use SIP between them

Wondering if anyone has experienced an intermittent one way audio (called party can not hear) problem in these conditions; Several IP501 phones local, same subnet. Remote asterisk No NAT anywhere Polycom IP501 ulaw only, canreinvite=yes Asterisk Call termination path is to a sonus GSX operated by the upstream carrier, ulaw only, canreinvite=no The idea is that if the Polycoms are

When I bought two Polycom 501 SIP phones, I naively thought they were Power-over-Ethernet (IEEE 802.3af) because they were "powered over ethernet." Silly me. Polycom must have some odd voltage or funny way of injecting the power, because the POE switch I bought for them (Netgear F@510P) won't power them, though if I use the Polycom-supplied AC adapter and ethernet power

I have a new Polycom Spectralink 8002 and am having trouble with the configuration or the unit but I can't see what's wrong. The unit does not seem to even attempt to register with the Asterisk proxy but I can make calls to it. I have viewed the syslog from the device which it will actually write to the asterisk server so I know it can be reached. I have also run a sip debug and

I am using asterisk 13.33.0 and POlycom phone with the latest firmware. The polycom phone is behind a firewall, the server is in the cloud. If the polycom has just booted - it receives a call, after some time (couple minutes) it no longer receives a ring. I see no errors in the CLI - looks just like the previous call as far as I can tell. Then reboot the phone and as soon as its ready call it

Hi all, I am working on asterisk 1.2.18 zaptel 1.2.17 Polycom 650 polycom 430 SIP version 2.0.3.0131 for IP 650 SIP version for IP430 2.0.3.0127 freepbx 2.2.1 I am trying to configure BLF using asterisk but failed. I would be thankfull if somebody help me. Regards FArooq ********************************************** 1 ********************************************** in my

I am using asterisk 1.4.18 with a polycom phone. sip.conf has: [532] type=friend username=532 secret=XXX dtmfmode=RFC2833 host=dynamic context=smvoice-sip callerid=532 qualify=no nat=no disallow=all allow=ulaw allow=alaw allow=gsm canreinvite=no I call into the dialplan and try to play demo-congrats and I hear nothing. Firewall is disabled. Everything is on the 192.168.1.X network for this

Giorgio, I'll answer in reverse order: I've not had reports of "noise" from my users. However, when I went down to get the s/w version from the phone that seems to be acting up the most, the user reported that earlier they were actually on a call that was ok then spontaneously dropped the audio. Per my instructions (based on another similar report I read on Digium's site),

I'm guessing that I need more info entered into the 'message centre' section. What did you key in? Paul Hales IT Support Adairs -----Original Message----- From: Jeff Pyle [mailto:jpyle490@gmail.com] Sent: Friday, 17 September 2004 11:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 I have two IP 500's on my Asterisk

Does anyone have a recommendation for a good SIP phone with a busy lamp field? I need my operator to be able to see extension status for about 20 extensions and transfer via HOLD + extension button. I've got a pair of SNOM 360s with the sidecar, but I'm very disappointed with them. The buttons are cheap and rubbery like a Sipura 841, the handset cord is short and cheap, the audio quality

I'm having the same "random" problem. I have "canreinvite=no" on all extensions. I have "qualify => yes" on all non-NAT extensions. I do have several NAT extensions, but I've not had reports of problems from those. 95% of my extensions (all polycom 501/601) are on a brand-new network comprised of 2 48-port Cisco 3560 1GB switches. In all cases, the

I searched and read all the relevant posts, but I still don't have a solution to my problem.. I've got a small queue for tech support calls using AddQueueMember. The agents are using IP300's from polycom. In my example, only one agent is logged int. When a call comes into the queue, asterisk sends the call to the one agent logged in. The agent answers and is talking to the

Hello , I learn handbook-draft.but I think I don't understand asterisk. where I can find some learning book about asterisk? thank u. B.R. John. -----????----- ???: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]?? asterisk-users-request@lists.digium.com ????: 2004?12?24? 7:51 ???: asterisk-users@lists.digium.com ??: Asterisk-Users Digest, Vol 5,

I have a small system, server, client and 2 phones. Phones are polycom 501's. In general all is working fine. I can call the two phones, speak etc... I can have the server call each phone and play a wave file. However, when trying to setup a direct dial number of 1044 that calls another machine running asterisk - something ODD is happening. ; This is not working.... [smvoice-sip] exten

I'm setting up a new Asterisk system on a Dell server and I'm getting "fuzzy" voice between the Polycom IP SoundStation 550 and the Asterisk server. I've checked all of my codec settings and both the Asterisk and the Polycom agree on u-Law encoding. I'm using the latest release of the Asterisk code (1.4.17) and other software. If I call between phones (i.e. two

I just tried this in my extensions.conf exten => **,1,Noop(Testing) exten => **,n,Playback(demo-congrats) Did a reload... and the above does not happen. I created as 12 instead of the ** and that works fine. Is there anyway to get the ** to work? I also am using a polycom phone if that affects things. I'm using asterisk 13.15.0 Thanks Jerry -------------- next part --------------

Converted a strictly VOIP system in NYC to NEC IPK TDM system... will have 25 Polycom 501's for sale. Best offer, offlist only please. R

On the landing page of the Polycom web site there's a "We're listening" nanosurvey, asking what is the one thing Polycom can do to improve their products. The link points here: http://polycom.zuberance.com/survey.htm I wrote a sentence about tweaking the user interface on the IP Soundpoint series phones, so that one can escape any level of any menu with repeated pressing of the

What gives with the stupid stupid power system on the Polycom IP500 (I hope they changed it on the 501's). Anyway, does anyone know if you can get some type of joiner to connect two Cat5 cables together, and will this affect the call/signal quality because my Ethernet cable is longer than the dumb 11ft polycom supply (oh and their manual is the wrong about which end goes into the handset