Re: [asterisk-users] opening Doors with Asterisk!?
My existing setup is based on ITS Pantel device (http://its-tel.com/upload/Door_Release_5.6(1).pdf) that is connected to Linksys PAP2 device that is connected to Asterisk (SIP, of course). Works well, very flexible, quite cheap straight-forward setup. The only thing you need to put attention to is to set up the dial plan of the pap2 properly (to accept two digits dial, if I recall right). Carlos Chavez wrote: You need a door phone or a switch to control the buzzer. I have used the 2N Entrycom and Helios line and they work very well. If you have an Astribank (Xorcom) you can use the output ports as switches. There are many brands of door phones you can choose from. You can connect them as regular extensions on the FXS port of the 3102 and control the door from there. On Mon, 2008-08-18 at 22:59 +0300, RoLaNd RoLaNd wrote: Hello all, i read a few articles online about the possibility to setup a buzzer door system to PBX using asterisk! currently my setup contains asterisk of course, and a sipura 3102.. what do i need to get such a feature done?! or should i ask if its possible?! __ Connect to the next generation of MSN Messenger Get it now! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP response 480 Do Not Disturb
The only solution that I found for this is to use Asterisk 1.4 with devstate backport (http://svncommunity.digium.com/svn/russell/asterisk-1.4/func_devstate-1.4/) and use the hints and to determine if it's inuse (or any other status) before the dialing - in order to generate a proper reply. I didn't find a way to handle the SIP 480 reply using Asterisk 1.2 properly. Note that it's an idea I was about to run but I didn't get to it yet. devstate on test machine compiled fine seems to be working from first sight. Tomer. Stefan Guenther wrote: Hello, I have switched on DND on a SNOM 360. When I call this phone, I get the following output: -- Executing [EMAIL PROTECTED]:1] Dial(SIP/user4-0821b0e8, SIP/user3|20|tr) in new stack -- Called user3 -- Got SIP response 480 Do Not Disturb back from 192.168.0.34 -- SIP/user3-081f8d20 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Executing [EMAIL PROTECTED]:2] Goto(SIP/user4-0821b0e8, fehler|s-CONGESTION|1) in new stack -- Goto (fehler,s-CONGESTION,1) -- Executing [EMAIL PROTECTED]:1] NoOp(SIP/user4-0821b0e8, xCONGESTION) in new stack -- Executing [EMAIL PROTECTED]:2] Hangup(SIP/user4-0821b0e8, ) in new stack I'm using a separate context to catch the dialstatus [fehler] exten = s-NOANSWER,1,NoOp(xNOANSWER) exten = s-NOANSWER,2,Hangup exten = s-CHANUNAVAIL,1,NoOp(xCHANUNAVAIL) exten = s-CHANUNAVAIL,2,Hangup exten = s-BUSY,1,NoOp(xBUSY) exten = s-BUSY,2,Hangup exten = s-CONGESTION,1,NoOp(xCONGESTION) exten = s-CONGESTION,2,Hangup exten = _s-.,1,NoOp() exten = _s-.,2,Hangup Now my question is: Is it possible to tell asterisk that SIP 480 shouldn't result in dialstatus CONGESTION, but in dialstatus BUSY? Thanks for your help, Stefan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT: Quad-band cellphones with wifi stable sip support
Hello, I am looking to purchase a new quad-band cellphone and I'm looking for one with WiFi and enough CPU power for stable SIP calls. I was wondering if anyone here can share his experience and recommend on a good cellphone. Any chance there is such a phone with even good WiFi profiles management or am I asking for too much now? :-) Thanks, Tomer. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP phones at multiple locations
Over my experience with 1.0 and 1.2 branch, if you register both phones the same SIP account and you will call it then both phones will ring, however, from reading here and there I heard mixed feedback about it so I just dedicated an account for each phone and I dial both of them at the same time until one answers ( in Dial()). Tomer. Michael Welter wrote: Each employee has a Polycom phone at his desk at the real office as well as a Polycom at his home office. I'd like a call to the employees extension to ring both phones. I'd also like one entry in the buddy list for each employee, and the buddy list to indicate he was on a call no matter which phone was being used. Should I do this in the dial plan, or should I register each phone on the same SIP userid? Has anyone done this? Thanks, Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXO USB that works with Asterisk?
Does anyone know if Xorcom's Astribank can work within a Xen VM ? Guillermo Salas M. wrote: On Thu, 2006-12-07 at 10:16 -0600, Nathan E. Pralle wrote: Hi all. Done some research, Googled a lot, but can't find out if there is a USB FXO adapter that works well with Asterisk. If someone knows of one or has used one, I'd be very interested to hear about it. Take a look: http://www.xorcom.com/astribank/features.html Many thanks, Nathan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTP Media Path
This is correct, if no NAT is involved anywhere and reinvites are allowed then Asterisk will stay out of the media path and be used only as Signaling server. So as for your answer yes, it will be able to handle more calls than expected because there is no CPU overhead of the media path. It is common strategy to have a single signaling server and have RTP servers all around the globe for latency and etc, media gateways. Vicky wrote: Asterisk wont sit in media path if both callee and caller agrees on common codec, both have canreinvite=yes in sip.conf, no t,T are used in dialplan ( please correct me if i am wrong ) , no call recording is enabled . I think asterisk does native bridging even if one is behind nat ( i tested with atleast one party behind nat not sure if it works when both are behind nat ) and devices should support reinvites .. On 03/12/06, *Dovid B* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I know this has been asked before and I went over the wiki but I have not been able to come to a clear answer. 1) If I have SIP Provider Asterisk - ATA and vice versa (ATA - Asterisk SIP Provider) from what I understand if NO NAT is being used then asterisk just starts and stops the session however the RTP media stream will be passed directly from the SIP provider and vice versa. (This is of course if there is no NAT involved). Now say I had such a set up will the server be able to handle more calls than average if the only responsibility if the server is to authenticated and pass along the calls ? (There will be an AGI running in the begining to determine what route to used based on how many minutes each route has used). Now if the ATA's are behind VOIP and asterisk is on a public IP then does asterisk have to sit in the media path ? Also can some one explain exaclty when the RTP session is started and stopped. Also another set up we are woroking on is SIP Provider (Incoming DID) Asterisk (for authentication based on PIN) - Back to SIP Provider. The asterisk server will be on a public IP. Can I have asterisk stay out fo the media path (here I asume yes. Just wana be 100% sure). Thanks a lot. Dovid ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk + Avaya S8700
Michel R Vaillancourt wrote: Tomer Horn wrote: Hello list, I am curious here if anybody here got an experience connecting Avaya to Asterisk using H323 / T1. I am completely lack of experience with Avaya and I wanna know if anybody here has connected Avaya to Asterisk using H323 and managed to stabilize it. Google provides mixed comments regarding the matter. The purpose of Asterisk on this matter is to provide outgoing calls from the Avaya through Asterisk, so features such as MWI and stuff are not necessary for me. Thanks, Tomer. I have done it with a Definity G3. It was actually pretty straight forward. Have you done it with H323 or T1/E1 ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk + Avaya S8700
Hello list, I am curious here if anybody here got an experience connecting Avaya to Asterisk using H323 / T1. I am completely lack of experience with Avaya and I wanna know if anybody here has connected Avaya to Asterisk using H323 and managed to stabilize it. Google provides mixed comments regarding the matter. The purpose of Asterisk on this matter is to provide outgoing calls from the Avaya through Asterisk, so features such as MWI and stuff are not necessary for me. Thanks, Tomer. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] several behind NAT
I'm not claiming to be an expert on the matter but I'm running here in my small office 2 softphones and 3 hardphones (I'll replace the softphones soon as well) and all of them are being NAT. The asterisk is located remotely and all the clients connect to it with the help of public STUN servers. My softphones are X-Lite and my hardphones are Snom. I also worked with Grandstream ATA (Handytone 286), Motorola ATA and Welltech phone - they all worked just fine. Tomer. Todd- Asterisk wrote: I've got my asterisk server in the DMZ of my local LAN - I've used my Budgetone and GXP2000's from the Internet- on direct IP connections with no problems. However, I'm about to deploy about 5 phones (either budgetone or GXP2000's) all on a LAN behind a NAT- on a different network than the Asterisk server. Should I look into using STUN servers? Will this setup be a problem? I've read about NAT and STUN on voip-info but am looking for more information.. btw- I'm not set on Grandstream. If you think Polycom or something can handle NAT better, then I'll use that instead. I guess there's no IAX phones yet... Thanks in advance. Todd ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Binding a peer context to a specific IP address
Hello, I was wondering if it's possible to set a specific sip/iax peer context with a certain IP addresses, so Asterisk will bind itself and connect to the peer only using a specific IP addresses. That way, Asterisk will use different IP addresses for different contexts (and making it easier for me to take advantage of gre tunnels/multi-homing). If not, does anyone have any idea how can I have certain contexts binded to certain IP addresses? Proxies? anything? Thanks, Tomer. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk in Xen 3.0
Hello! Are there any known (bad) issues / experience running Asterisk inside Xen VM? Has anyone experienced running Asterisk inside a Xen VM with PCI access to PRI adapter? Regards, Tomer. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and SATA Raid 1
I believe that depends what you do with your Asterisk. If your hard drives I/O is not intensive then go for it. I got a SOHO Asterisk installed on Athlon XP 2000+ and ATA 100 Software RAID-1. The server is mostly idle but there are times it does some work and it's also serving as extremely low I/O file server and a pptp gateway. The asterisk's job, if matters, is to take incoming calls, record them into a wav, encode to mp3 and mail away. It also being used as a SIP host to peer with other VoIP suppliers. It does some work, but not so much I must admit; works tho. No complains here. mustardman29 wrote: I was just wondering if there are any problems using the latest FreePBX with SATA Raid 1 using hardware assisted software Raid like most modern chipsets support? I know that Digium and FreePBX were not recommending it awhile back but I think that was based on 2.4 Kernel and Digium hardware issues. I am assuming it is not a problem with the latest FreePBX using 2.6 Kernel and Sangoma cards? Any info would be greatly appreciated. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
