similar to: Best strategy to find and solve voice quality problems

In our experience, the total cost of softphones(money, reduced sound quality and lower reliability) in a large call center environment is actually greater over time than the cost of a channelbank and cheap analog headphones. We've tried 2 softphones, 2 kinds of SIP VOIP hardphones, 2 kinds of SIP analog adapters and we've tried channelbanks over the last 3 years. Right now we are half done

Hi All, In the midst of trying to pilot a deployment of Microsoft Lync (mainly for non-voice collaboration, specifically IM) and integrate it with our Asterisk (11.6.0 if it matters) deployment and a "everything in one place" tool when people are out of the office. I have everything on the voice side playing nice from the Lync side (Lync->Lync, Lync->Asterisk,

Hey all, Alright. So we decided to not go with Avaya for our next PBX and we are now full on into an Asterisk/Lync 2010 implementation. Asterisk/FreePBX is our SIP gateway and call center and Lync is our internal UC and IP-PBX server. I've already got Asterisk tied with our Nortel/Merridian Option 11 with QSig and all is beautiful (except for the Opt11 not receiving names from * but

Hi, It seems that asterisk doesn't send RTCP packets when a call is on hold. Is there any way to get asterisk to send these packets? I'm in the process of setting up a Lync (microsoft voice) server which will use an asterisk box as a gateway. The trunking between asterisk and lync is 'working' however when a call is put on hold asterisk stops sending RTCP packets to lync, and

I have a client that want to try the softphone with USB handsets route to see if hardphones will even be needed. I always push for hardphones (Polycom) so I am not sure about softphones or USB handsets. This is going to be for a 300+ seat call center onsite and many offsite, I plan on using OpenVPN for the offsite machines. Any advice on softphones, handsets, or practical experience with

Dear Sir, I use asterisk 1.8.11 (192.168.100.202)to connect lync server .I use tls port 5068 to connect to this lync server . The tls is ok to establish and I make call from softphone 3200 (register to Asterisk) and dial 9XXXXXXX (9+85225082162) , this prefix will dial to trunk lync_trunk and pass to lync server(192.168.100.14) using tls . But the lync client in opposite side ringing and they

I'd recommend any of the following, which are all in your price range Snom 300 Polycom IP430 Polycom IP501 Aastra 9112i Linksys SPA-922 Grandstream GXP-2000 Cory Andrews ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Zeeshan Zakaria Sent: Wednesday, November 01, 2006 11:17 AM To: Asterisk

snom 300 :"> CS -----Urspr?ngliche Nachricht----- Von: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Im Auftrag von Kristian Kielhofner Gesendet: Mittwoch, 1. November 2006 12:01 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: Re: [asterisk-users] Which IP phones have best voice quality,preferably under $150 Zeeshan Zakaria

Hello, Can anyone recommend a hardphone that has the following qualities... Both headset and handset ports Headset port has amplification built in Supports SIP using G729 We are switching from a Nortel switch to Asterisk. If anyone is familiar with Nortel phones, the Nortel 2216 phone has the features we're looking for in a SIP phone. Any help would be greatly appreciated. Thank you.

We have a client that currently has a Microsoft Lync setup. I must admit I know nothing about this setup. What we would like to be able to do is allow the phones on desks connected to this server the ability to dial something that would allow the phone to access an asterisk box to be able to do an agent login over their LAN. Is there any way to do this? Can the Lync server have a SIP trunk to

Hi guys We're using asterisk 1.8.23.1 on CentOS 5 and are trying to create a trunk to MS Lync server. If I create the peer in sip.conf the trunk connects with no problem. However, we prefer to use ARA. Whenever we define the peer in our peers table, the trunk does not work, even if we use sip show peer <peer-name> load. Has anyone got any experience of connecting to Lync using ARA?

sorry... typo.... the problematic phone has the 192.168.0.13 the asterisk has 192.168.1.211 when i connect a snom phone on the cable that was in the soundstation 6000 before and configure the phone to use 192.168.0.13 it does register on the asterisk via 5060/UDP... it would be helpful if someone, that has a running soundstation ip 6000 could send the configuration... :-/ regards, yves Am

Hi all, I have to buy some IP phones. Previously I have used Grandstream GXP-2000, Budgetone 101 and Linksys SPA-841. I always had problems with sound quality with all of them, and I was always of the opinion that it were the phones which were not good. In GXP-2000 deployment of about 50 phones, some work good, some have sound problems like words missing, clicking sounds when talking, and some

Hi all, I have looked through the wiki guides and also Siemens user guides but they haven't proven useful. Nor has the normally trusty googling. Also have upgraded to the latest Optipoint 400 Standard SIP firmware. Having read a few previous threads on the Optipoint it seems that there isn't much take up with Asterisk. Which seems a shame as my experience with testing it has been

Are they any gotchas to be aware of in getting Asterisk and Lync 2013 talking to each other using SIP? Or is Lync a pretty standard implementation of SIP? Cheers Tony -- Tony Mountifield Work: tony at softins.co.uk - http://www.softins.co.uk Play: tony at mountifield.org - http://tony.mountifield.org

Dear Friends, We have implemented "Asterisk" in our organization. There are 150 members in our organization. At present all are using softphones. Now, I want to buy hardphones for our staff. Can anybody suggest me that what is the best hardphone for Asterisk with low-cost? Thank you. Regards, Chandra. --------------------------------- Ring'em or ping'em. Make PC-to-phone

Hi Mark, yes, you are right... these are different VLANs I configured the other phone to use the same IP (192.168.1.13)... and it worked flawlessly... on the SAME Networkcable in the same plug... so it must have something to do with the polycom phone config... remember... when I use tcp the phone tries to register, but does not even try with udp... thank you, yves Am 21.12.2016 um 13:34

I have looked at several IAX and SIP soft phones but I have been disappointed with the sound quality on my Windows XP Pro PC. Also the GrandStream problem is that they don't yet support headsets. When I turn auto answer on and I dial in it instantly picks up with the speaker phone. But if I have the handset picked up when a call is coming in the line is busy. That means that the phone itself

Some SIP hardphones (Polycom) or softphones (Counterpath) embed a module that metter MOS. Regards 2015-03-25 14:21 GMT+01:00 Patrick Beaumont <p.beaumont at hatsoffsoftware.co.uk>: > Hi everyone. > > We regularly get customers complaining about call quality issues. Most of > the time it turns out to be their own broadband. Very occasionally server > load. Does anyone have

Hi Patrick, You are welcome to try our tools out for active and passive voice quality measurement tools. It's waveform analysis (like PESQ or POLQA) and VoIP metrics analysis (like G.107 E-model and other metrics). You can read more at http://www.sevana.biz or older site http://www.sevana.fi On Tue, Mar 31, 2015 at 1:16 PM, Patrick Beaumont < p.beaumont at hatsoffsoftware.co.uk>