similar to: Echo on SIP Side?

asterisk-users-bounces@lists.digium.com wrote: > In the following setup: > call coming from a pstn line -> into FXO card -> asterisk -> SIP > phone > > i get an incredible loud echo in the SIP phone (about 0,5-1s) > (everything i speak into SIP phone microphone i hear in its > speaker). The person calling from PSTN is not getting any echo. Make sure you're not

Hi, I have was testing some of the different ring types with my polycom 500, and the ALERT_INFO settings. Now when my phone receives a call it won't ring. All the other phones ring fine, and my phone wasn't the only one I rebooted with the changed sip.conf and impd.conf. I have reverted back to a standard sip.conf and impd.conf and I still can not get my phone to ring for any incoming

I'm still trying to work this out. I've got this in my sip.conf [1003polycom] type=peer secret=abc123 host=dynamic defaultip=192.168.1.215 context=default mailbox=1003 subscribecontext=phonestatus [1004polycom] type=peer secret=abc123 host=dynamic defaultip=192.168.1.214 context=default mailbox=1004 subscribecontext=phonestatus And this in my extensions.conf [phonestatus] exten =>

Has anyone seen an issue with SIP phone (polycom 500) dropping calls at irregular intervals with no errors in the asterisk log files? I am having this issue as described and it is a complete pain in my rear to trouble shoot because when I call my cell phone I can get a call to last over 30 minutes yet when I call another office that uses a standard pbx I can't get past 10 minutes. For some

Ok, so I emailed the list earlier about my polycom phone not ringing when anyone called in. Well, polycom support said that is impossible that this could happen because of a change in a configuration file. However the new phone arrived today (a refurb.) and it also would not ring. So I obviously got rather frustrated and blasted away all of my configuration files from the FTP server. I then copied

Hi everyone, This one has me baffled.... We installed 16 Polycom IP 500 Phones in January on an Asterisk Server: 3.0Ghz 768MB of Ram 30GB Harddrives 10/100/1000 NIC Our Lan setup is like this: 3 Netgear FS526T Switches 2 10/100/1000 Mbps copper gigabit ports 24 10/100 Mbps ports 100Mbps out to phones the 1Gig ports connect the switches together, and one gig port goes to the

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

Good news! FYI, worked with Mark this afternoon to test changes needed to reduce or eliminate echo involving pstn calls on the new tdm fxo card (bug #1902). Three simple source code changes (testing-only at this point) resulted in zero detectable echo on all incoming and outgoing tdm pstn calls, even in the first second or two. I've not noticed any unusual side effects at all. Since the

> > First, ensure the TDM card has been defined to match the telephony > standards for each country where the card is installed. Bad echo on these outside North America is likely to be because of impedance mismatch. Echo cancel isn't likely to properly fix it. I was surprised at the lack of info about setting up the TDM cards for various countries. I eventually worked it out from a

On Wed, 13 Dec 2006, Wireless wrote: > Hi Gordon > > I too have this problem with one of my two BT lines, very very annoying. I > was using a TDM400P and am now using a Sangoma A200 (lightning got the TDM), > I think the sangoma is very slightly better (less echo) but I might just be > kidding myself. Another think I've found it that using a faster CPU (now on > a 450

I'm having a bit of an issue with one of the bargain x100p clones, and I'm not sure what the right approach is. My symptom started as way loud offset delayed echo from voip hardphones -> PSTN through the clone card. I played with and then learned everything I could about echo cancelling, and have managed to bring the echo down to acceptible levels, where it's not too intrusive. In

Hello again, I am using mix monitor and the majority of the sound records perfectly. I then get a "Internal Data Stream Error" near the end of the sound file. Has anyone ever seen this? I am allowing the ULAW amd ALAW codecs and an example dialplan entry is ; ; phone line phone1 exten => phone1,1,Answer() exten => phone1,2,MixMonitor(test.wav|av(0)V(0)) exten =>

Okay...here's one for all of you.... 3 party meet-me conference: Call 1: Comes in to MyAsterisk on an E1 PRI into the system. All TDM, no VoIP at all involved. No echo at all. Call 2: Call comes in via IAX....(TDM -> Asterisk_1 -> IAX/GSM -> MyAsterisk. Caller immediately hears his own echo Call 3: Call comes in via IAX....(TDM -> Asterisk_1 -> IAX/GSM -> MyAsterisk.

I was wondering if anyone has had a problem with an initial chirp from asterisk after dialing an outside line through a TDM400? I am using 3 polycom 500's and it happens regardless of what phone I am at. So I was thinking maybe it was an issue with asterisk or the uLaw codec? Thanks. Jared Armstrong -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have my * and polycom system setup to do Auto-Answer for internal SIP/Staff calls, and I am running into an issue with this and the polycom call transfer feature. * is seeing a new call come through from the polycom and is then transferring the call over. I need to know if there is some way I can grab a message from the SIP header or something to determine if I should not set the ALERT_INFO tag

Hi, I have an TDM400 4 FXO module setup on my dual Celeron server running asterisk 1.0.2 and I have had to restart the asterisk process multiple times lately. I was wondering if anyone else has to restart the asterisk process after storms roll through their area. Before I restart I usually have sound quality issues and weird pickup problems. But, once I restart the asterisk process everything

Hi, everyone: I am developing a system using Asterisk, TDM-400 analog cards, analog lines, and Polycom SIP phones for internal extensions. Initially there was bad echo but after a series of efforts, I've managed to reduce it to a negligible level (it only happens when both parties speak simultaneously, and even there, only for a few hundred milliseconds). From an echo standpoint, things are

OK... so I'm not sure what I'm looking at. I've had the good old echo problems on my Rev C FXO again this morning, so I thought I'd attempt some debugging, though I'm not sure what I'm looking at. This call has echo. Channel: 2 File Descriptor: 20 Span: 1I> Extension: Dialing: no Context: incoming Caller ID string: "External Call" <99999999> Destroy:

Let me start by saying no, this is not the normal stupid echo question :) I currently have a SIP device and an X101P, and have had the usual echo issues and have played around with the various solutions, none of which are quite perfect (understandably).. however, my girlfriend is a little more picky than I am when she uses the phone, so I'm considering a Digium TDMXXB card for the house. My

We have occasional echo, mostly on lines that come across as quiet.. I am trying to troubleshoot how to disolve the echo. i have tried the steps on voip wiki, but have not recently been able to make any difference. the echo isnt horrible most of the time, and seems extremely random in that i can call a number once without echo, then dial the same number a second time and get echo. things i am