similar to: Extension Mobility with Asterisk and Cisco 79x1 phones

Hi everybody. I know maybe this question has been posted some time ago, but I need your updated opinion on the subject. I'm replacing our old pbx with asterisk. I have two TE207 dual pri (e1) cards on a clustered system (one on each node). I absolutely need to connect 4/5 analog extensions with modems, they're being used for remote assistance on very old systems which cannot be upgraded

Hi. Again one big mysterious problem I hope some good guy can help me solve. I'm trying to connect some Siemens C450 SIP IP Dect phones to asterisk (1.2.13) (I have actually 3 handsets + 3 ip base). After configuring them and rebooting, all of them register properly on asterisk, then, after the first call, they appear no more registered as registered in asterisk, and on the handset the

Hello. Once again, I came up with a problem for which I can't seem to find a solution. I'm not able to make BLF work with Thomson ST2030 phones and Asterisk (1.2.13). I've set up hints in dialplan, as well as Subscibe keys on the phone. The LED status gets updated according to the associated line status. However, when a phone is ringing, If I try to pickup the call by pressing the

Hi. I'm running some asterisk boxes on different sites, some equipped with a couple of ZapHFC cards, others with Junghanns quadBRI cards. All boxes were compiled with Asterisk 1.2.10 (libpri 1.2.3 zaptel 1.2.6) and bristuff 0.3.0 pre 1s, distribution is Fedora Core 4 with kernel 2.6.17.3 The cards are connected to Telecom Italia's NT1/NT1+ S/T lines; some of them are point-to-point,

Hi everybody. I am in the following scenario: 1 Customer "A" calls an asterisk box over a Zap channel on a toll free number during night time 2 The incoming call enters an AGI script on the dialplan 3 The AGI script plays back a welcome message, then starts the music-on-hold stream 4 The AGI script originates a calls to a stand-by operator's cell phone (operator

Hi. Does anyone know about a simple one-fxs ATA with pulse dialing support that can work with Asterisk? A SIP one would be ok. I've been told that the Digium S101i IAXy does support pulse dialing; although it's a iax2-only ata it could be enough. I need a bunch of them to convert some old fashioned rotary phones into VoIP ones (I'd like to disassemble the ATAs to remove the boards

Hello everyone. I know it's a little bit off-topic, but I was just wondering... Has anyone ever had any experience with asterisk, a wi-fi meshed lan (with more than one access point) and wi-fi sip phones? I made some tests but I'm not really satisfied.... Wi-fi phones are a curse (as far as I know even Nokia eSeries -I personally own an e70 model- have their flaws): - random sip

I know in advance maybe I'm overlooking something stupid, however I'm really lost and cannot find the solution... Situation: - asterisk-1.2.13 on a linux box with no iptables active. - one iax2 peer defined - one wildiax phone running on my laptop the soft phone is configured to connect & register on asterisk, however, I cannot get it working. What am I missing? Please help!!

Hi. I'm stuck into an odd situation. Here's what happens: 4 Thomson ST2030S 2 Cisco 7912 3 Cisco 7940 2 AAstra 480i Asterisk 1.2.17 Diva 4BRI + chan_capi I've just upgraded yesterday from Asterisk 1.2.13 to 1.2.17. Until yesterday, everything was just fine with 1.2.13. Immediately after the upgrade, *all* the 7940 are no more able to make calls, just receive them, while 7912

Dear all, I'm using the ellipse package and I would like to verify if the confidence region that I build with this package can be equivalent to an Union Intersection Test. I used different value for the t-statistic but I can not find the right equivalence. Does someone know how to choose the right value? Thanks a lot. Antonio -- Antonio Lucadamo, Dipartimento di Scienze Economiche e

The Asterisk Development Team is pleased to announce the release of version 1.2.11 of Zaptel. This release includes a small number of fixes, primarily to support recently updated hardware products from Digium. It also contains a very large XPP driver update from Xorcom for their Zaptel-compatible products. Thanks for supporting Asterisk and Zaptel!

The Asterisk Development Team is pleased to announce the release of version 1.2.11 of Zaptel. This release includes a small number of fixes, primarily to support recently updated hardware products from Digium. It also contains a very large XPP driver update from Xorcom for their Zaptel-compatible products. Thanks for supporting Asterisk and Zaptel!

I am trying to use the Gotoiftime CMD to not allow calls to be placed between the hours of 12am-5am, except if you know the PIN number to dial out and if the call is for 911. What is the best way to implement this solutions? I have the gotoiftime like so: exten => s,1,GotoIfTime(5:00-11:59|mon-fri|*|*?custom-blacklist,s,1) and using Read for the PIN like so: exten => s,3,Read(Secret,,3)

Hi, Does such card exist ? It seems all existing models are designed for PCI buses. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080201/b449b98e/attachment.htm

Hello, I need to connect one diva server 4bri to a portuguese BRI interface. The operator (PT) said that this bri is in point-to-multipoint mode (S0). Previously one PBX has connected to that interface. The asterisk and diva drivers are working ok but i cannot communicate to outside via this bri. Xlite gives me the message: "call failed: declined". Anyone have experience with this

Hi all. I have enabled the attended transfer feature in features.conf. I'm using it and I want to resolve some questions, I hope someone can help me :) When I transfer a call to an extension: - The extension rings during 15 seconds and the call returns to the "transferer". Is there any possibility to recover the call before the timeout of 15 seconds expires? I mean, I would like

Hi, Anyone here has any experience with the Nokia E70 and asterisk ? I read on the nokia website this phone is capable of talking SIP and do Presence based on SIP/SIMPLE. Please share your experience, I'm thinking of getting one but want to be sure I can use it with * before I do. Thnx. -- Michiel van Baak michiel@vanbaak.eu http://michiel.vanbaak.eu GnuPG key:

I'm preparing to deploy a small number of Grandstream BT101's and GXP2000's to a remote location (which I won't have access to). I'd like to have them pull a config file from my server - I'm almost there... The phones are looking for the config file on my webserver which is good. I need to generate that file however. I see a tool on the GS website to generate

Dear all, I would like to ask information about the package mvtnorm in R. It is very useful for me the "qmvnorm" comand, but I see that it can compute only quantile for equicoordinate. Is it possible to have a curve (or a set of values) if we don't want equicoordinates? Thank you Best regards Antonio. -- Antonio Lucadamo, Dipartimento di Scienze Economiche e Metodi Quantitativi

Hi (Armin)! Does someone knows how to identify the type of the card? The delivery note says it is a V-4BRI-8M, whereas lspci reports a 4BRI-8M. What is it really? Are there any Eicon tools to identify the card type? thanks klaus 0000:0a:03.0 Network controller: Eicon Networks Corporation Diva Server 4BRI-8M Rev 2 (rev 01) Subsystem: Eicon Networks Corporation Diva Server 4BRI-8M Rev