similar to: IAX crackilng

I have a cisco VPN from router to router over a Data T-1. The ping times are consistently 32ms with random ping responses of 295ms -408ms about every 30 secs to a minute, I have jitter buffer enabled. The connection goes like this... Mitel SIP phone to Asterisk A, IAX trunked to Asterisk B and then T-1 to a PBX, the calls are internal so they are terminating on Toshiba digital phones. Loud

I have two IAX trunked *, there are loud crackles and pops, they are dialing out a T-1 and are sip devices, it also drops words, any help or Ideas? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060516/dc68274f/attachment.htm

On 23/11/17 19:49, Sorin Srbu wrote: > The Skype flatpak at home worked beautifully, that is to say for about ten > minutes, then the sound (using a Koss SB/45 headset) it started crackling to > the point of not being usable anymore. I hung up and reconnected, then sound > was fine, for another ten minutes at which point the crackling returned. > Checking the sound prefs, I saw

Hi all, I have setup a Dell PowerEdge 2550 with a Sangoma A200 card with 2xFSO and 1XFS modules. The PowerEdge specs are 1 x P3 1133MHz, 512MB RAM. Sangoma A200 has 3 analogue PSTN lines connected. This server is based in Office 1, with 5 users all with a Linksys SPA942 VoIP Handset. There is another Office (Office 2) connected to here using VPN. There are two users in Office 2 with the

On 11/19/2017 12:53 PM, vychytraly . wrote: > Maybe try flatpak version? there should be no problems with dependencies > there... :) Thanks, I didn't know about flatpak... even tho it's in the base repo!? I have to admit though, after reading through the flatpak website about it, I still don't understand how it could overcome the dependency problem with the skypeforlinux

Is anybody using Mitel phones? It appears that when you page a Mitel phone using asterisk's MeetMe, the paged phone will hang up the call its on to take the page. Thanks in advance. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110621/764a6fa9/attachment.htm>

Does anyone have any experience getting a Mitel SX-200 EL/ML PBX system to work with *? I'm not getting inbound or outbound calls to work. (Inbound caller gets line busy tone.) SETTINGS FROM MITEL: I built a Crossover cable and connected it like this: PSTN--T1--ASTRISK--T1--OLD MITEL -Crossover Cable Pin-out: 1 - 4 2 - 5

Hello all. I just received a TE110P and am trying to hook it to my Mitel 200SX has anyone successfully done this? My configuration is as follows. Asterisk -> TE110P ->Kentrox (csu/dsu) -> Mitel T1 Card. All I get is a blinking yellow on my TE110P card and an alarm on my Mitel. T1 card. Any advice would be great. Zaptel.conf span=1,0,1,d4,ami e&m=1-23 dchan=24

I have a small government department that wants me to implement a Asterisk installation, however, they connect to the Government PBX, a Mitel SX200, and want to keep the ability to do that. I know there is no chance to connect the digital extension lines, but would it be possible to have the pbx admins send analogue extensions over and have those lines interface through an FXO interface? Or

I have an Asterisk box, 1.4.30 with a PRI. A Mitel 3300 is connected to the Asterisk box via SIP trunking. When a user calls from the Mitel through the Asterisk box the user can speak but can not hear the far end. But - when I route the Mitel user to echo() it works, send and receive. The Mitel user also can record and playback greetings. One thing I have noticed is that when the Mitel user

I know this is not strictly an asterisk issue but it is related I guess. Just to let you know that after many calls to Mitel the consensus is that they will be releasing a new version of the 5220 that is dual boot (minet and SIP) next week or the week after. This firmware will only appear on NEW phones manufactured after the release date (no one could confirm but the 23rd of sept was

We have a S100U which we were using as our sole internal phone extension. Today it died. Picking up the phone now results in a brief dialtone followed by bursts of random static and crackling noises and then a second or two of fast-busy followed by silence and more crackling noises. I've tried the hardware attached to a couple of different machines as well as a couple of different phones.

Hi all, Our company has deployed a Mitel 3300 system (only about 2.5 years ago) and we are experimenting with setting up Asterisk in our head office (for business continuity, ie we have a bird flu epidemic and no-one can come in, therefore use SIP softphones at home to co-ordinate activity) and at a remote site in the Isle of Man (connected via 2Mbps SDSL) Ideally we'd like anyone on either

Hello, I have seen the Mitel 5055 SIP phone mentioned a few times on the list, does anyone have any wonderful or horrible things to say about it? We are thinking about using them because they have many more programmable buttons than the Snom200 phones and are about $70 cheaper. Thanks, MATT---

Hi, Some Mitel products, which are CentOS 6.x or 7.x servers based with some telephony services added, are only supported on VMware virtualisation platform, even all if theses CentOS guests are certified on last CentOS/Redhat Virtualization or Microsoft Hyper V platforms. So, at he moment we have three bad choises : - Migrate part of our virtualization services from KVM/libvirt to VMware ; -

Hi, I have a client using Mitel 5212 phones in SIP mode with a 1.4.21.1 Asterisk server (and a couple of previous 1.4 versions). They're mostly happy with the combination except for this one issue. For incoming calls only, either originating from other local SIP phones or from a PRI, calls won't get bridged (remote party get's hung up) if the call is answer too quickly on the

I have a problem which I need to resolve. We are trying to put an asterisk between a Mitel PBX and the world. We are adding Voip service via Asterisk. Here is are config files for the settings but our problem is the following. We are able to send calls to the Mitel pbx and it's the T1 connections is green saying it's ok. The support department from Mitel said that they use e&M and

I guess my CentOS is screwed up because that function doesn't work. Only way to resize it was to go into the menu areas and set it there. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.centos.org/pipermail/centos/attachments/20060412/4f37d0af/attachment-0001.html>

Does anyone know if this phone can or does work with Asterisk ? I have a potential client who wants to throw away their current mitel ip system (keeps crashing on a regular basis - not good for a phone-based company), but are reluctant to purchase new handsets. Thanks. Julian.

So I reached the point where my PRI is accepting incoming calls, but I cannot dialout. I must be doing something stupid, but I can't figure it out. The Asterisk box is sitting between the Mitel and the phone company, and has PRI lines to each. Asterisk was built from CVS r1-0 Log for a call from mitel heading outbound: ------------------------- -- Accepting call from '' to