similar to: Polycom Echo using IAX2

Lately I have been experiencing a lot of echo from my Polycom phones. Only I hear the echo and it's not on every call. I've researched it via google and the forums and every echo problem usually relates when it's using a Zap card and not an IAX provider. Can anyone give me some advice or where to look to help solve this echo problem? This never occurs on any of our other phones,

I?ve been tearing my hair out trying to get a Polycom phone (behind a NAT) to register to an * box behind a Cisco SIP ALG. With known good credentials configured on the phone and in *, I get 403 Bad Auth when trying to register. If I put the phone onto the same LAN as * it works fine without changing any authentication parameters whatsoever. If I make the secret blank (null) on the phone and *,

For some time (since pre 1.0), I've been seeing the following messages fairly regularly from some, but not all, of my SIP devices: Nov 23 06:37:59 NOTICE[2568]: chan_sip.c:7645 handle_request: Registration from 'John Doe <sip:9005552368@100.200.200.100>' failed for '200.100.50.25' I have a mix of Sipuras, Grandstreams, ZIPs and Ciscos, but the message seems to come

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

A Polycom 550 and a IAX client (Mozphone and ZoIPer were used) are having a conversation. Call quality is reported as good except for an echo with a 3 second delay. Most of my searches are saying echo happens only on the PSTN piece, but there isn't one here. Can someone point me in the right direction? Asterisk 1.4.21.2 Under 40 users Quad-Core AMD Opteron(tm) Processor 2352 and 4G RAM

I'll do that, but I'm sure that it's the Polycom :) Caller ID shows up fine on my 7960... W/ the name on top and the number below that. -- Executing NoOp("SIP/614-3ede", "Caller*ID is Matthew Marlowe <6092521155>") in new stack When the phone rings, only 'Matthew Marlowe' would display. When I answer, both the Name & Number will show.

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

Victor Hoodicoff wrote: > > > I think your impressions of Aastra are outdated. Install the latest > firmware, download the latest documentation and test and THEN give an > opinion! Did you miss the part when I wrote I have Asstras sitting on my desk collecting dust. I program on average about 5 per month, deal with about 40+ per day. They're as impressive as that Hyundai in

We are using MFC/R2 driver successfully in at least three places in Brazil. I have problem with an Asterisk integrated with MFC/R2 with a Siemens Hicom 300. I can get a good audio quality with Grandstream, Polycom, and X-Lite softfones, but SIPURAS and Linksys get a garbled audio, something like a "Darth Vader" voice. We have tried everything in Sipura. The SIPURA 2000 and the Linksys

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

Good evening all. I'm having this strange behavior when dialing two or more simultaneus calls via IAX to other * boxes. Sound starts to have more latency, wich increments until it's almost impossible to talk (6 or more seconds), I try this calling with two grandstreams, one grandstream one tdm410p, one tdm410p and sjphone, sjphone and one grandstream, the result are similar.

I recently dug into this, from what I've seen, the best bang for the buck out there is going to be Polycom's. A local vendor has Polycom IP500 phones for $174 shipped to me. IP500 would be comparable to a 7940G I'm assuming. I ran into the same problem with pricing, don't want grandstreams, but can't afford the nice Ciscos. Check out the Polycom's. -Tim -----Original

A while back i had asked about possible replacements for snom 360 phones that were breaking and causing issues and we all discussed the problems we had with the 360s and some suggestions were made but the new polycom phones had just hit the market and not many people were able to comment on them. Basically i am looking to get some new phones and in the process get rid of the countless number of

Andrei, Do you have X-Windows running on the linux box? I had a similar issue that was eliminated when I stopped this process and samba from running. Now samba is only allowed to come up during non-business hours, for changing BG music. Also, make sure your registration period in either (polycom) ipmd.cfg or sip.cfg is set to be at least the default 3600 time period. I also removed the

I'm hearing an echo when using a headset with my IP 501 / 601. The phones are using BR 3.1.2 and SIP 1.6.3. I use tftp to configure the phones. The sip.cfg is the default from polycom except for the parameters required to connect the phones to asterisk. I have absolutly no echo with the handset, but do have a slight echo on the speaker phone. I haven't ruled out room acoustics as the

Hi, We are in the process of moving from an Avaya Definity to Asterisk for our institution's phone system. I got one feature that the Avaya had, which I have not been able to reproduce with Polycom phones and Asterisk; since this feature seemed so small and useless to me when testing, I kind of ignored it. Now I am getting more "I miss that" requests than I expected. =) On the

Contact me off list, I have a sample extensions.conf file that I can share. It has Paging (one to one and One to Many) Ivr includes, time of da routing and it is geared towards Polycoms. > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Thomas Johnson > Sent: Friday, January 20, 2006

Wiley, There are a couple of issues that we saw while not using this option. 1) sip authentication failures as Asterisk is not able to reach Polycom phones. A typical problem description is here: http://lists.digium.com/pipermail/asterisk-users/2004-December/079251.ht ml 2) DTMF issues for Transfers, Hold or simply to dial extensions. This problem is more pronounced when you are using

Hi, all-- What do you make of this? Here's my call log--looks like there are a lot of calls going in and out of the server that are not real incoming or outgoing calls. Does anybody have any clue what is happening? 2006-11-14 16:41:00 Local/8183... 8183461773 "8183461773" <8183461773> 8183461773 NO ANSWER 1 47. 2006-11-14 16:40:59 IAX2/Voice... 8183461773

I am in the process of setting up an * system using Polycom IP 500's. I don't want to spend time setting a ftp server for application and configuration files at the moment, just want to use the web interface to the Polycoms. DCHP works OK and IP is obtained correctly. Polycom fails to load .cfg file and holts. I have read the 143 page admin user guide a couple of times...and I missing