similar to: Polycom 301 questions

Hi all, I'm having an odd problem with my polycom 301. I am initiating a call to it with AMI Originate() function: Action: Originate Channel: local/112 at Management Context: to_meetme Exten: s Priority: 1 Variable: dropped_conf=111 The "to_meetme" context is very simple: [to_meetme] exten=>s,1,MeetMe(${dropped_conf},id) If I specify every other device I have to test: *

Are all of the sip phones in the same context? > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Robert Jenkins > Sent: Tuesday, January 16, 2007 1:44 PM > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: [asterisk-users] Polycom IP601 - some hints working,

Could not find your post for 4 months ago. -------------- Original message -------------- From: "Anton Krall" <akrall-lists@intruder.com.mx> > Yes, check a post that I made about 4 months ago, I posted the cofig for > setting the speaker, handset and ring volumes .. > > |-----Original Message----- > |From: asterisk-users-bounces@lists.digium.com >

I have the following situation: Asterisk 1.2.1 25+ Polycom 501 telephones. Bootrom 2.6.2.0032 Application 1.6.2.0041 Some 501's local to my network, some across the great INTERNET divide. PRI connected to Sangoma card. Issue: horrible echo (and squeals, and "underwater-like" sound) on speaker phone when calling from extension to extension. echo not present when calling outbound

Hi all, thanks to the all of you. This list is very interesting also for a newby like me. My problem: I just setup my first full working asterisk installation with this config: 1. n.1 GXP-2000 2. n.4 Budgetone 102 3. n.1 TDM400p (3 FXS, 1 FXO) Everything seems to work fine, but the sidetone... it's really annoying! We can hear the sidetone only when we call to the outside (PSTN), it

I'm having a problem with polycom ip601. If I Dial() directly eg Dial(SIP/4000) it works perfectly. The polycom rings, and stops ringing as soon as I hang up. But if the phone is called via a queue, the polycom continues to ring long after I've hung up. Other phones in the queue (grandstream, cisco) don't have this problem, and stop ringing properly when I hang up. polycom bug

I have quite a number of users complaining that when they are using handsfree to talk over a landline, the other end can't hear them. It's like the person is speaking 5 feet away and can barely hear their voice. However between internal SIP calls, it's fine. What could be the problem? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello list- I'm having an extremely odd issue with an installation of mine. The system is running * 1.2.12.1 and currently handles around 100 handsets. With the exception of a few Grandstream DTA's, all devices are Polycom 320, 430, or 601's. After a recent power outage, I'm having an extremely odd issue with one of the handsets. One of the Polycom 601 units simply reboots every

has anyone done it? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080528/ff622d28/attachment-0001.htm

Hi everyone, I have been dealing with a certain issue with a particular customer site for months now. The problem occurs when there is an error with caller id as shown in the following: WARNING[16036]: chan_zap.c:6309 ss_thread: CallerID returned with error on channel 'Zap/3-1' When this happens, it appears that the call still goes through as I can see the caller still navigating

Is the main difference between the two the full duplex speaker on the 330? -- Warm Regards, Lee

Hi all, My company has pretty much standardized on Polycom phones and I am in the beginning phase of writing a GUI for administering/managing polycom provisioning at multiple sites which we intend to release as OS. I've started studying the docs and I'm having trouble understanding the following xml attribute: voIpProt.SIP.requestValidation.x.request.y.event I understand what it

Skipped content of type multipart/alternative-------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 3097 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20040712/2a1bda93/smime.bin

Hello, I'm trying to setup the revised Polycom remote reboot script as found on: http://www.voip-info.org/tiki-index.php?page=Polycom+reboot+hardphone+script I'm not sure how to use this script, it's just a perl script, so I tried creating an executable perl script and running it, but I get the following: [root@asterisk1 agi-bin]# ./polycom_reboot.pl 192.168.3.205 Checking ARP

Hello, I got 3 Polycom 300 phones, and upgraded to the latest firmware provided by the reseller. This is my first experience with Polycom and I cannot make them register in my Asterisk Box. I followed every advice I found (including separating [user] and [peer] in sip.conf. Using ethereal, I found that it tries to SUBSCRIBE to the asterisk box and it receives a 403 FORBIDDEN message. I

I've been going nuts lately trying to get rid of an annoying echo problem that makes my asterisk server unusable when clients try to call me. Here's the breakdown of the issue - Hoping that someone can throw me a clue: My setup is as such: Single AMD Athon machine with X100P clone card and voip through multiple providers . * Inbound calls through the X100P that do not bridge to

Hey everyone, I have been trying to figure this out and I am just getting no where with it. The office is using Polycom IP 601 phones. Everything sounds great in terms of quality on both heads. However, users of the phone are having trouble with their headsets and handsets. Some users are hearing their voices back come back on the phone. If they adjust their volume lower, then it goes away,

Greetings, I am trying to get either of the above features to work with *, but can't seem to get it quite right. If anyone has them working, I'd sure appreciate an extract from the relevant config files. Or, maybe I'm tilting at windmills, and * doesn't support them - in which case, the underlying business need is to provide the one incoming call on more than one

For people who are using GS 101, what do you think the sidetone generated by the phone. I find mind a bit annoying. It has a delay and you notice it as an echo. The volume of the sidetone is also quite hight. I am distracted when both caller and called party talking over each other occasssionally. The volume of the sidetone can be turned down using the volume button but it also control the

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 -