similar to: NetworkWorld article on Open Source Telephon y

An interesting article for those needing ammunition to sell Asterisk within their organisation or to others: "Is open source IP telephony ready for prime time? Yes" by Zenas Hutcheson, St. Paul Venture Capital Network World, 06/07/04 http://www.nwfusion.com/columnists/2004/0607faceoffyes.html On a related note, they also have an article arguing the contrary position (see link within

Hi, I'm trying to detect and or log the duration a a conference (Meetme). I need it in order to do some billing for theses services. Any ideas on how to do it? I googled around but found nothing. Thanks in advance epablo -- Pablo Endres <epablo@comvoz.com> ComVoz Communications USA: +1 954 343-2085 Ext 199 Venezuela: +58 212 7713195 Ext 199 Colombia: +57 1 3256840 Ext 199

Hi, Has anyone made a successfull instalation of cdr_odbc?? I've install unixODBC-2.2.8 (made my own RPM) and then built the module. I'm trying to send the cdrs to a M$ SQL server. The sql connection works because I can do any query via isql. When I do the calls I get the following output on the asterisk console: -- Executing Hangup("SIP/test1-a5e1", "") in new

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

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'm having another problem I can't work out - make for x in wrapper asterisk-driver; do make -C $x all || exit 1 ; done make: *** No rule to make target `ccflags'. Stop. make: *** No rule to make target `ccflags'. Stop. make[1]: Entering directory `/usr/src/asterisk-oh323-0.6.1/wrapper' ./check_ver /usr/src/pwlib pwlib ./check_ver /usr/src/openh323 openh323 g++ -Wall

All, I have access to a Cisco AS5300 w/ 4 T-1's and a Cisco 3600 with no boards. I was wondering if it would be possible some how to Have one of these Ciscos in-between our sip phones and the asterisk server so that we could use G729 Codec. Sip Phones (7960's & ATA's) via G729 -->Cisco Gateway-->Asterisk via G711. Any ideas? Has anybody done such an implementation or know

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

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

for inbound calls, i can register context = from-sipphone register => 1747xxxxxxx:passwd@proxy01.sipphone.com but how do i configure to make outbound calls to them? exten => _1747XXXXXXX,1,GoTo(dial-sipphone,${EXTEN},1) .... [dial-sipphone] ; ; SIP to sipphone.com ; exten => _X.,1,Dial(SIP/${EXTEN}@??????) ^^^^^^

We have integrated an Asterisk (TE110P) and a Mitel SX200. We usually get over 500 frame errors and over a 500 slip errors per hour. When the errors reach 1000 per hour the Mitel will take it's T1 card offline. At that point no calls can be routed from the Asterisk server to the Mitel and the TE110P reports a Yellow alarm. What can be causing all these Frame and Slip errors? We have been