similar to: Zaptel 1.2.11 released

Displaying 20 results from an estimated 2000 matches similar to: "Zaptel 1.2.11 released"

2006 Oct 27
3
[OT] wi-fi ip phone scenario
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
2007 Nov 23
2
How to bridge two connected calls
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
2008 Jan 23
2
Modem bridging on Asterisk (no VoIP involved)
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
2006 Oct 16
1
ZapHFC & quadBRI D-Channel going down randomly
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,
2006 Nov 22
1
gotoiftime and blocking calls
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)
2008 Feb 01
1
BRI card with PCI-E interface
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
2006 Oct 28
1
Diva server 4bri and Portuguese BRI
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
2006 Nov 15
2
some questions about atxfer usage
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
2006 Nov 16
3
Nokia E70
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:
2006 Nov 10
2
config template for Grandstreams
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
2006 Oct 18
3
identifying Eicon Diva Server V-4BRI-8M vs 4BRI-8M
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
2006 Oct 30
3
Cisco 7960 Skinny calling SIP phone
Before I got down the path of converting a Cisco 7960 I have over to SIP I wanted to try and set it up using Skinny. The phone registers ok with Asterisk. When I call a SIP softphone extension on my network the call is made and I can answering it. However no voice is heard over the call. When I debug Skinny on the console after the call has connected I see the following messag: Recieved Alarm
2006 Oct 23
2
T.38 faxing with spandsp and Grandstream HT.486
Hello ! I 'm trying T.38 faxig with spandsp using rxfax/txfax as fax terminal. As another endpoint I 'm using Grandstream HT 486 ATA and a fax machine. Has anybody success with the HT486 as T.38 terminal ? ATA as originator: I managed only onetimes a successfull T.38 fax session. The other times the HT486 did not initiate a re-invite with T.38 parameters. Or shall the Terminator
2006 Oct 23
4
Problems with chan-capi and Eicon Diva 4BRI
Hi! This weekend we had a problem with our Asterisk Box which ran flawlessly for nearly 4 weeks. The Asterisk server sits between the PSTN and a Siemens PBX and bridges 2 BRI lines. No calls, not incoming, not outgoing. The admin rebooted the Dell Box and then everything worked fine again. Now, I'm analyzing log files to find the cause. During the Asterisk outage the logfiles only show
2006 Oct 19
3
Bristuff qozap drivers problem
Hi, For a significant time now (since about 0.2.0-rc8n) the qozap driver has become very verbose if an ISDN line is not connected... I get the messages below every couple of seconds in the asterisk logs. The "flaw" in the messages is the "Alarm cleared" message - The alarm cannot possibly be cleared because there is no physical media connected into that port!!! (BTW - All
2008 Feb 02
2
ATA with pulse dialing support over FXS
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
2008 Jan 26
0
Extension Mobility with Asterisk and Cisco 79x1 phones
Hi. I'm trying to develop a module that emulates the Cisco Extension Mobility feature from CallManager (the ability to log in to a phone and temporarily acquire the extension, soft key programming, and all other settings for that user profile) with Asterisk 1.4 and Cisco 79xx phones (some with SIP and some with SCCP, as the 7914 extension module does not support SIP). I've almost
2006 Oct 30
0
Asterisk and Siemens C450IP
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
2006 Dec 18
1
Thomson ST2030S and BLF
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
2006 Nov 22
0
iax2 - wildiax phone & myself puzzled
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!!