[Asterisk-Users] voip phone reviews
Hi all, I keep coming across the same question over and over again - which phone is best ? So I've setup a website (www.voip-reviews.net) with the intention of bring phone reviews together into one place. If anyone would like to add a review of their phone, please go ahead. In good faith, I can only personally review phones that I've actually used. So the more people that add reviews of phone's they use, the better. I think this could be a good resource for newcommers to the VOIP arena - feel free to shout me down if you don't agree. Regards, Jon Lawrence ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk@home silly problem, please help!
On Wednesday 09 March 2005 23:12, Junk Mail wrote: What's making me desperate is that the lines go by, capiinit is, in fact, runned, and Asterisk still fails in the end. I login and type my very first command asterisk -vvvc and it then starts with no trouble. Is this strange or what ? Thanks in advance for your help. I'm willing to bet that the modules haven't actually loaded when asterisk tries to start. I'm pretty sure I had the same problem on my box. You need to slow down the boot (I know sounds weird everyone nowadays wants their machines to boot asap), instead of inserting text grabage, find where the modules are loaded and insert a couple of seconds of sleep after the modules are loaded - I'm away from my box atm and can't for the life of me actually remember the syntax. Run a quick google for 'bash sleep'. HTH Jon Jon ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Comparison Charts
On Thursday 10 March 2005 13:50, Anton Krall wrote: I couldnt agree with you more Jim. Im realdy using Asterisk and agree 100% with what you say... I was asking for a comparison list with other PBX's because for example, for a customer, they have heard of Avaya and Cisco and they all are selling IP now... So In order to get your customer to trust Asterisk over those guys, you need to show him the diff. Between the two and some lists of the features on the others compared to Asterisk.. For a list of asterisk features see http://www.asterisk.org/index.php?menu=features I've never seen a direct comparison chart, but yes I can imagine situations where one would be useful - if you create one, perhaps you'll post a link to it on the list :) Jon ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk behind IX66
On Sunday 26 December 2004 16:21, Steve Beaumont wrote: Marc, I'm not sure what you mean. Are you suggesting that I enter a sip entry, E.g. [0870xyzabc] for the telappliant provided PSTN number ? Yes. I had to do something similar for my PSTN numbers over IAX - not near my * box at present, but I'll post what I needed when I get chance. Jon ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Incoming Calls
On Tuesday 28 December 2004 04:32, C F wrote: Just a note on this. I tried using an external device with the TDM400 configured as 4 FXO to ring even with asterisk. But no matter how I configured it, asterisk always picked up. and the external device didn't ring (just the first ring for CallerID to come in). Asterisk should only pick up in 1 of 3 conditions: 1) you have an answer() statement in your dial plan 2) your dial plan dials the extensions and one of the IP phones picks up the call. 3) Asterisk drops the call into voicemail. I'm thinking that you probably have an answer() statement in your dial plan before the dial() statement. Jon ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] switch statement.
Hi all, I'm a bit confused about how to use the switch statement. I've got an IAX2 link between 2 servers (SA SB). I have use the switch statement to include extensions from SA onto SB which is happening perfectly. I've read that I can't use the switch statement the other way (ie SB-SA) at the same time. What is the correct way for both servers to know about each others dialplans ? TIA Jon ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream and CallerID
On Monday 20 December 2004 20:24, David Ishmael wrote: Jon, I went in and setup my extensions.conf just like you stated, and I can see in the incoming call number in the log file but the phone does not display the number (it's just showing the phone's extension number). Can you post your sip.conf context for the GS so I can compare? sip.conf [2002] type=friend host=dynamic username=2002 dtmfmode=inband mailbox=2002 context=remote callerid=Name 2002 canreinvite=no secret=xx Jon ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoIP Termination
On Friday 17 December 2004 09:10, Shoval Tomer wrote: snip Fortunately, asterisk will do that for you to the second, looking at the cdr records and totalling up the duration column for a specific period will tell you what your bill would be at the few cents a minute you'll be charged. Actually not true, although a common mistake. You do not pay to the provider by the second, so never, ever, ever total the duration column. Usually you don't pay by the minute either, but by some intermediate, like in units of 12 seconds, but you need to check it with the provider. This is a serious mistake, and it gets worse the more calls you total. Let's say, that you pay by the minute. Having made 6 calls of under ten seconds costs you as if you've made 6 calls of 59 seconds. If you just total the duration of the calls, you'd think you're paying for one minute and you're actually paying for six. That's a big mistake. Now imagine you had 12000 calls to total. Always total the price for each call, never the duration. OK, yes there's more to it than simply adding the up the duration of the calls. But, you shouldn't have any problem finding out from your provider how they charge for calls. For example I believe that BT in the UK charge a minimum of 0.05 for any call, but so long as the call is longer than that you pay x amount per minute -= rounded to the next minute no doubt. Once you know how your calls are charged, calculating your bill from the cdr records should be no big deal. Either load the records into a spreadsheet and do the calculations or use something like php or perl and pull the records into a webpage. Jon ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream and CallerID
On Monday 20 December 2004 14:59, David Ishmael wrote:
I'm having similar problems with my Grandstream BT-100 SIP phone. I've
removed the fromuser=1234 from the sip.conf file but the phone still shows
1234 in the display when getting a call. I can see the incoming PSTN CID
in the log file but for some reason its not passing this to the phone. The
CID looks something like:
Joe Somebody 7035551212
Others have stated that the BT-100 can't take characters, only numbers so I
would assume there's a function like SetCIDNum(${CALLERID}) to extract the
number and send it to the BT-100. Can anyone that has the CallerID working
post their setup/configs so I can see what I'm doing wrong?
I've used SetCallerID(${CALLERIDNUM}) with /gS phones and they display CID
correctly.
ie:
exten = 2000,1,SetCallerID(${CALLERIDNUM})
exten = 2000,2,Dial(SIP/2000,30,Tt)
Jon
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk, Capi, Controller
On Thursday 16 December 2004 06:59, SIN - Robert Siedl wrote:
Hi List,
I have asterisk 1.0 on a SuSe Linux 9.0 with one AVM C4 ISDN card an one
AVM Fritz card running for outgoing an incomming calls.
From austria telecommunications company I have two isdn nt, the connectet
on
avm c4 card and I have one gsm-gateway for mobile handy.
How can I asterisk instill, wenn a outgoing call beginn with 0664, take the
controller 3 (=avm fritz card) else take controller 1 or 2 (avm c4 card on
telekom nt)
Have somebody a idea?
Hi,
you control which controller a call goes out by specifying the relevant msn.
configs something like (of top of my head):
capi.conf
[interfaces]
msn =12345
controller=1
msn = 54321
controller=3
extensions.conf
exten = _0664.,1,dial(capi/54321:${EXTEN})
exten = _.,1,dial(capi/12345:${EXTEN})
first line would send call out of controller 3, 2nd line sends it out
controller1 .
HTH
Jon
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] least sucky FXO interface?
On Tuesday 14 December 2004 18:44, Jean-Michel Hiver wrote: Dorn Hetzel wrote: My FXO card doesn't seem to work so well. Never tried my SIPURA as an FXO device though... I use it as an FXS and for that usage it gives me no echo. To the list: Am I right understanding that Fritz + BRI line = no echo issues? From what I can gather, most of the echo problems seem to be down to motherboard problems. I have seen echo problems on only one machine here at home, changed machines and echo went away. However, although I get little or zero echo with my fxo cards I have had isdn installed and get absolutely no echo using a AVM fritz card. Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk to sip client behind Firewall/NAT-cancall but cannot receive calls ?
On Tuesday 14 December 2004 15:19, Shoval Tomer wrote: As far as I can remember I only opened sip and tftp ports for the phone. For some reason (didn't look into it too much) the call stays with sip and doesn't use RTP. SIP is what sets up the session (ie it does session handling) RTP is the transport protocol that the audio uses. If you're using SIP then you're using RTP eos. Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi question
On Friday 10 December 2004 09:50, Peer Oliver Schmidt wrote:
Jon Lawrence wrote:
I can receive incoming calls. However, I can't call out.
When ever i initiate an outgoing call, I get the following on the
console:
Executing Dial(SIP/2014-8817, CAPI/*msn|bdialednumber) in new stack
Dec 9 23:10:24 WARNING[1390]: chan_capi.c:653 capi_call: Destination
*msn* requres a real destination
Does any one know what this means and/or what I can do to fix it.
relevant conf's:
capi.conf:
[interfaces]
msn=mymsnnumber
incomingmsn=*
controller=1
softdtmf=1
accountcode=
context=isdnin
devices=2
extensions.conf
[ISDN1]
exten = _0.,1,dial(CAPI/${msn1},b${EXTEN})
Do you have msn1 set to your real MSN within extensions.conf? It has to be.
Yes.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi question
On Friday 10 December 2004 10:41, Peer Oliver Schmidt wrote:
My msn is 1234, the called number is 0123-45678. This is my log entry
Executing Dial(SIP/26-dd65, CAPI/1234:b012345678|60|T) in new stack
In extenstions.conf I have
exten = _.,1,dial(CAPI/1234,b${EXTEN},60,T)
in capi.conf
msn=1234
I asked this on the irc, Jas_Williams said he had
dial(CAPI/msn:b${EXTEN})
note the differenne to mine ie msn:ext not msn,ext
I changed the , to : and I can make outgoing calls :)
Now I need to find out how to set the outgoing clid - never had to do that
with pots ;) but I'm sure it's on the wiki.
Jon
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi Fritz! - FXO or FXS?
On Friday 10 December 2004 11:02, Ron Norton wrote: Hi There, I got chan_capi working OK with a Fritz! PCI card, so I can make and receive ISDN calls into Asterisk. Great. Besides the Fritz! card I have a few ISDN phones connected to the same ISDN S-bus, each answering to a different MSN. -- to telephone carrier -- incoming ISDN LINE -|NT-1 box| --- Sbus\ | --/ x--x- ISDN SBus x-x - -- ---- | Fritz | | ISDN PHONE | | ISDN PHONE || ISDN PHONE | | card | | MSN X2 | | MSN X3 || MSN X4 | | MSN X1| -- ---- - || PCI || ASTERISK NT-1 : ISDN network termination box installed by the phone company. However whenever I dial using these ISDN phones ... Asterisk is bypassed, and the call gets routed only by the telephone switching system connected to the ISDN/NT box. What I would like to do is to use the ISDN phones connected to the same ISDN/Sbus as the Fritz! card to make (also) calls into Asterisk. How can I call Asterisk with any of these ISDN phones, without making an outgoing (paid) call through the telephone switch ? Should I need to install two Fritz! card, one connected to the ISDN/NT line, the other to the ISDN phones ? :-)) Firstly don't hijack threads - many people won't see this message as it's way back in the past. Start a new thread instead. To do what you want, you'll have to install another card - not a fritz either. If I'm right on this then the card needs to work in NT mode. I think only a hfs (hfc ?) based card can do this. Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BT100 how to pickup a parked call
On Friday 10 December 2004 13:23, Greg - Cirelle Enterprises wrote: Does anyone know why the bt100 cannot park and pickup a parked call? attendant announces the call is parked at extension 701 but the call cannot be retrieved by any other phone. also, the bt100 constantly rings when the phone is hung up after parking. anyone experienced this? using the basic features.conf [general] parkext = 700 ; What ext. to dial to park parkpos = 701-709 ; What extensions to park calls on context = parkedcalls ; Which context parked calls are in parkingtime = 60 ; Number of seconds a call can be parked for pickupexten = *8; Configure the pickup extension. Default is *8 Have you got 'include = parkedcalls' in the bt100's context ? Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Training Needed in SouthEast U.S
On Friday 10 December 2004 21:31, Paul Rodan wrote: here and no longer willing to assist. I found an interesting 3-day course offered by this one company, but they turned out to be in London, and it'll be tough enough to get them to transport and pay for this course in a local area, trying to get them to fly me out of the U.S is infeasible. Can you give me any details about the company in London TIA Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Guide to Cisco 79xx
On Wednesday 08 December 2004 21:41, Paul Rodan wrote: Any idea on an ETA? Like within a day or two? If you need help, 4 hands are faster than 2. It's simply a case of when I get chance to sit down and do it. I hope to have something ready before Monday. Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_capi question
Hi all,
I've finally taken the plunge and had isdn installed.
It's a BRI from Telewest in the UK.
I'm told by the provider that it's in point to multipoint mode with 6 digit
presentation.
I'm using an AVM fritz (bt speedway) with the fcpci-suse9.1-3.11-02 driver.
Chan_capi seems to be installed fine.
I can receive incoming calls. However, I can't call out.
When ever i initiate an outgoing call, I get the following on the console:
Executing Dial(SIP/2014-8817, CAPI/*msn|bdialednumber) in new stack
Dec 9 23:10:24 WARNING[1390]: chan_capi.c:653 capi_call: Destination *msn*
requres a real destination
Does any one know what this means and/or what I can do to fix it.
relevant conf's:
capi.conf:
[interfaces]
msn=mymsnnumber
incomingmsn=*
controller=1
softdtmf=1
accountcode=
context=isdnin
devices=2
extensions.conf
[ISDN1]
exten = _0.,1,dial(CAPI/${msn1},b${EXTEN})
I have a button on my 7960 that uses the context ISDN1 for outgoing calls.
I've tried to get mISDN working, but chan_capi refuses to admit that capi's
loaded - so I abandoned that and went back to the avm module.
TIA
Jon Lawrence
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP endpoints ---- RTP stream
On Wednesday 08 December 2004 04:44, Gonzalo Gasca Meza wrote: Hi all, I have just setup Asterisk, but the problem is that all RTP stream pass through Asterisk, I mean all call setup and voice stream pass trough Asterisk once the call is established. Is there a way that call setup is established, the RTP stream pass just between the SIP endpoints. Example: Works like this SIP IP phones ---Asterisk RTP stream-- SIP IP phone Asterisk SIP IP phones --RTP SIP IP phone yes, unless you have canreinvite=no in your sip.conf, assuming that the phones negotiate the same codecs then they should be able to initiate a re-invite so the the stream goes peer to peer taking * out of the loop. Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Guide to Cisco 79xx
On Wednesday 08 December 2004 19:47, Paul Rodan wrote: Anybody have a guide to the Cisco 79xx phones? One that I can give the 7 or 8 ppl in my office so that they can stop asking me questions. I was going to type up a basic guide but then decided I don't want to reinvent the wheel, one of you may already have one. I tried to use Cisco's guide but it's for their own protocol, a lot of options are different or rearranged. I need a basic user guide that instructs on placing calls, answering calls, putting calls on holds, warm transfer, blind transfer, activing/deactiviating do not disturb, conference calls and how to access multiple calls on hold. Like how 2 can be on hold on line 1 and if another call comes in, it goes to line 2. The only way to get back to the 2 on hold on line 1 is to hit the line 1 button. Anything anybody has would help, it'd at least be a start and I can addon/enhance or even simplify/dummy down some of it. I get so many questions from ppl forgetting how to do something I think throwing a manual at them would be far superior. Any help would be greatly appreciated. We use a mix of 7940's and 7960's in SIP mode, with firmware 7-3 I'm actually in the process of writing something like this -will post a link when it's completed. Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: dont write me again
On Wednesday 01 December 2004 19:44, Stephen R. Besch wrote: Exactly. Would those people who respond from the mailing list digest -PLEASE-PLEASE-PLEASE- do the following simple things: 1)Strip out the digest messages that have nothing to do with your reply. 2)Copy the appropriate subject line into your message subject before you send the message so that we can actually tell what you are writing about without having to search through your reply. you missed out the most simple read the damn email. It clearly tells you howe to unsubscribe Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Interrupt latency problems
On Wednesday 01 December 2004 20:31, Steven Critchfield wrote: I am glad it solved the problem. Now if only someone knew what it was about the stock RH or FC kernel that makes it happen you could get RH or FC to stop using that patch. That or maybe more people will be like me and always cast a weary eye upon a prepackaged kernel no matter what distro it came from. First thing when installing any distro is to bin the kernel and install a vanilla one - how else can you be sure of the state of possibly the most important part of your system. Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Kind of off-topic: VoIP services and multiple callers
On Monday 06 December 2004 22:59, Rich Adamson wrote: Inline... I know that VoIP providers can supply their customers with a local number and/or virtual numbers, and then that number/account can be used with Asterisk (well, it depends on the provider and whether or not their service is compatible with Asterisk). However, I have a question: can more than one person make/receive a call at the same using one VoIP line? Some providers support multiple calls, others don't. If five people in the office all need to use their phones at the same time, would I need five VoIP lines, or would I only need one VoIP line? Am I over-thinking this? Same answer. Its up to the provider, and when they do support multiple calls, they typically charge a fee/minute/call so its no skin off their back. voiptalk.org allow 2 calls over a standard (free) account. You can get more calls allowed for a fee. I'm sure there are others that do similar. Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] two questions
On Tuesday 07 December 2004 04:36, Erick Perez wrote: Hi people, question one i see that asterisk is now in 1.x release. having tried it in the past i want to know if i can use a voice modem as an outgoing line. i know in the past that was not possible/supported so im just asking in case the option is now available. yes, if that voice modem is a x100p or clone (same chipset). question two im planing to use asterisk as a pure voip solution with sip phones and h323 phones no need for digium/dialogic hardware at this moment (but i will in the near future). however i have not been able to find a documentation (not so complicated for a newbie) that help me to setup asterisk in this mode. suggestion/comments/flames welcomed. see www.voip-info.org Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] High(er) availability
On Tuesday 07 December 2004 14:39, E. Versaevel wrote: Which app do you use for monitoring the primary box and if it fails taking over the IP address by the backup one? I haven't found a suitable (active-active) app so far. Thinking of using heartbeat or something. Take a look at keepalived I've used it (along with it's implementation of VRRP) to provide failover for routers. I see no reason why it couldn't do the same for an asterisk server. You might have to write a module to monitor the actual asterisk process. Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] full duplex sound card
On Monday 06 December 2004 18:29, Guido Rebert wrote: The ac97 is a realtek chipset.. bull. lspci |grep 97 00:1f.5 Multimedia audio controller: Intel Corp. 82801CA/CAM AC'97 Audio Controller (rev 02) 00:1f.6 Modem: Intel Corp. 82801CA/CAM AC'97 Modem Controller (rev 02) perhap realtek have been bought out :) if you want to know more about AC97 take a look here (http://www.intel.com/design/chipsets/audio/) - I believe the original AC97 standard was an Intel/Microsoft driven thing. Jon- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call parking/transfer not working on IAX2 connections
On Friday 03 December 2004 23:58, Dominik Strehlke wrote: Hi there, Maybe this has been cared about before but I could not find any solution to this problem either in the wikis or in the list archived. If someone has found a solution before please just tell me where I can find that info..thanks :) I am using iaxComm to call other people who are either using SIP or also IAX clients, like me. All of us are connected to the same asterisk server. When I talk to someone who is connected via SIP, I have no problems transfering him or parking him, it all works very well. But if the other party is connected via IAX like I am, then the pound-key will not work, I do not hear any transfer message. Please note that I am always using the same client, iaxComm, while just some of the others have SIP or IAX clients. So...why is my pound key not working when I am talking to someone who is connected via IAX? I did not find and clue about this anywhere :( Ah yes, the Dial entry is probably correct as it is always the same because it's included in a macro which is used by all normal extensions and includes both the t and T parameters. Any help is very much appreciated! Okay, this is a bit of a guess. iaxcomm - sip. call stream has to always go through * therefore # will work. iaxcomm - iax. call stream will re-invite (perhaps not correct phrase for use with iax) unless canreinvite=no. Once clients are talking directly * is out of the stream therefore # cannot work. Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] non blind call transfers
On Friday 29 October 2004 21:17, lenz wrote: Hello list, I was looking for a way to implement non-blind call transfers with *, i.e. the usual behaviour of most PBXs when pressing the flash button: - A and B are talking - A pushes flash - A is free to compose a new number - B hears music on hold - A's call is answered by C - A hangs up - B and C are in conversation As much as I can understand, * only supports blind transfers, where if C does not answer the phone there is no way for A to get back to B. Is there a way to have a standard flash behaviour? The above is exactly what happens with my system - I've not done anything special (ie patches) to make this happen. I can do attended transfers by simply doing 'flash' while in a call, dial the new number and talk, press flash again and hang up. It works perfectly for me :) pressing # while in a call allows blind transfers. Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] fallthrough extension.
On Friday 03 December 2004 09:50, Jason Williams wrote: The right way to do it is two have two contexts see below [internal] exten =_3XXX,1,Dial(. etc include=catchall [catchall] exten = _.,1,Dial(Zap. etc Excellent - works perfectly. I knew there'd be a way to do it :) OK, so I need to include a catchall for each user, but that's no great big deal. Thanks, Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk crashes my router!?
On Thursday 02 December 2004 20:25, Mike Dent wrote: Or can anybody suggest a good ADSL modem/router which works well on PPPoA in the UK. And does not crash with SIP! :) I'll need it to do NAT and also bridge several global IP's. depending on your money :) intertex ix66 (www.intertex.se) - sold by voiptalk.org cisco 827 cisco 837 cisco moular (ie 1721,2610,3620 etc) with wic-1adsl jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] fallthrough extension.
Hi all, I'm trying to sort out my dial plan. What I'm wanting is something like the following - a bit simplified but hopefully you'll get the idea. 1) match internal extensions: dial them 2) anything else: send out zap 1 is easy :) it's 2 that's giving me problems. I had hoped that the 'i' extension would act as a catchall extension but it seems to only do that from a menu. I've tried matching _. (hoping that * would parse the dial plan from top to bottom) but that just took over the entire dial plan and everything went out of the dial with the _. match. I'm sure that there must be a way of doing this, can anyone point me in the right direction. TIA Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GSM modems
On Thursday 02 December 2004 17:22, Stuart L. Morris wrote: Has anyone tried connecting * to a GSM mobile phone with internal voice modem? I'm trying to route calls to mobile phones out over a mobile phone connected to my * server via a serial connection. In this way these calls will be much less expensive than if they are routed via a landline. I'm guessing that I start in the modem.conf file but the only references on the web are a couple of years old and generally to people trying to use HCF modems as PSTN interfaces. The only way I know of is to use something like a 'cell socket' Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] threeway calling
On Thursday 02 December 2004 16:20, M. Smadi wrote: any idea on how we can setup threeway calling in * I was going just link to wiki but I couldn't find it - it must be there or on the mailing list, there's no way I learnt how to do this myself. But I couldn't find it so here you are: 1) ensure that your extension is allowed to transfer calls. 2) ring callee A 3) place callee A on hold (ie flash) 4) call callee B 5) when callee B answers press flash and you're in a 3 way call. It's the same as an attended transfer but without you hanging up HTH Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NEED HELP!!
On Tuesday 23 November 2004 13:17, WipeOut wrote: Please can someone look at my last two posts and try and shed some light onto why my system is dropping calls.. If I don't get it right we will be forced to drop Asterisk which I really don't want to do.. I'm willing to bet that your problem is related to NAT. What are you using as a firewall ? Can you simply set th firewall to foward the relevant iax port to * - this should create a static entry in the nat table. Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] fxo connection in the UK
On Tuesday 30 November 2004 13:05, Edward Eastman wrote: Most people get echo issues with x100p's in the UK due to mismatched impedance, the newer TDM400P is much better, and you could get this with 3 FXO modules (otherwise known as a TDM03B I believe). The TDM fxo modules aren't approved for use in the UK yet - use at your own risk. Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] fxo connection in the UK
On Tuesday 30 November 2004 13:23, Ian D. Wlloughby wrote: The TDM fxo modules aren't approved for use in the UK yet - use at your own risk. Jon But they do work great in the UK :-) Hopefully they'll get BABT approval soon. I'd imagine that using them at home isn't likely to cause problems - a business setting is a different kettle of fish (the stakes are much higher). Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Comparision of IAX2, FWD, iaxtel etc etc.
On Tuesday 30 November 2004 08:41, el Flynn wrote: Wilson Pickett wrote: How would the IAX (host=dynamic) client first register? Am I missing something? the host=dynamic directive in iax.conf simply tells * that the IAX client could be registering from any particular IP address. it's more of a control mechanism so you can limit an IAX account to only connect/register from a specific IP. as for how the client registers: you'd configure that in the UA itself, telling it where the * server is. I think what he meant was how can an external client initiate a connection. A. yes you have to forward the IAX port to your * box Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Fwd: Re: [Asterisk-Users] Adit 600 channel bank in UK setting]
On Thursday 18 November 2004 22:16, Tim Robinson wrote: Channel banks are a peculiar US thing. Be careful! You will almost certainly be better off using voip handsets (SNOMs are cool, avoid Grandstream for anything other than domestic environment) and a few Sipura-type ATA's for the analogue fax machines etc. or some Digium analogue cards. So what would you advise using in the UK to interface with standard 2 wire phones - I'm trying to avoid having to use ata type adapters. Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best SIP phone for high quality telemarketing
On Friday 19 November 2004 04:42, Luke Connolly wrote: I'm really happy with my Polycom IP 600 http://www.polycom.com About $200 cheaper than cisco and no difference in qual or features. If only :). In the UK, I found 7960G's for £200 (ish) if I could lose $200 (about £105) I'd definitely be tempted with the polycoms. Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] UK available SIP phone?
On Sunday 21 November 2004 12:03, Clive Carter wrote: Hi, Anybody here from the UK using Asterisk at home? I'm looking for a SIP phone which will work with Asterisk and not leave me broke! I got one of the Tecom ones from Solwise but it refuses to login to Asterisk server for some reason. May have to send it back. What are the other options please? Thanks Mike I use Grandstream Budge Tones. They are cheap, and some people say they look it, but they work ! I have also got ipDialogs SipTone II. They are twice the price, and although I have got the basic functions working, for some reason they just will not connect to VoiceMail I have used both sipura 2k's and ata286 - both worked perfectly with my dect phones. Currently 2K is connected to DECT, ata286 to fax and a 7960G for main use. I have in the past had budgetones and yes, they do look cheap - but so what if it's only for home use. I work from home, hence the 7960G (I simply needed more lines). But imho you'll struggle to beat a sipura 2K with a good quality DECT phone - although that works out a similar price as my 7960G. HTH Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Fwd: Re: [Asterisk-Users] Adit 600 channel bank in UK setting]
On Monday 29 November 2004 19:12, Peter Hoppe wrote: Jon, I actually had some more discussions with Tim on this issue, and it seems that the channel bank would still be a good option to choose for internal purposes. I would not see any other solution than a channel bank to connect many 2wire phones into one asterisk box. I had a talk today with Carrier Access, and it seems that the adit would do us fine. The fxs cards of the adit 600 are actually reprogrammable for uk phones (dip switches). We have requested a test platform from CAC which I hope would arrive here shortly, and we would test how the fxs ports work with different uk phones. I'd be interested if you could report back to the list with your findings. I would really be unhappy to scratch our existing phone cable network and to lay an entire new LAN and to buy many IP phones. First of all - new installations always have teething problems. Then the admin headache with the many IP phones. Also - the solution doesn't scale very easily. For each new phones we need a new network socket... or a hub. Then one mains connection per phone (with power supply - more fire risk). And on and on... Using 2wire phones eliminates all that - cables are there already, users can buy any phone they like, we can put in additional sockets without admin effort and so on. Really - 2wire rocks! Wiring isn't my problem - all our connections are over cat5e. In our building we rent out most of the offices, the clients provide their own phones and we simply provide the lines. Some of their phones look pretty expensive, so I'd rather not tell them that they can't use them anymore. As with any office scenario power sockets are an issue - people never put enough in when they design the rooms. The pstn connectivity is an entirely different matter. Our connection is via pri so this isn't a great issue. Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Top posting - are we there yet?
On Tuesday 16 November 2004 17:12, Jay Milk wrote: I'm a fairly reasonable person, and I have yet to see one good argument (and quoting netiquette is not on argument, that's opinion) for bottom-posting. To me, it is terribly inefficient and wastes time, especially when you hide your post between the original message and some ludicrously elaborate signature. Top-posting, to me, is more logical, as it presents the answer in a prominent position. And inline-posting makes sense when you're responding to multiple questions or points in an email... Whether you top post or not is irrelevant really. Top posting - you have to scroll around to find out what question they are answering. bottom posting - you have to scroll to find the answer. I'll reply to both top and bottom postings - if I think I've got anything to add. What's more annoying is people who just click reply instead of starting a new tread. Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail Mailbox Configuration
On Friday 05 November 2004 02:55, Darly Coupet wrote: Hi pamela, I have tried using option 2, by recording the files and placing them in mailbox directory and referencing the files with Voicemail() command. But it is not working... The busy and unavailable messages play fine. Please advise On 4 Nov 2004 at 5:23, Pamela Weis wrote: hello, there are two options to do this: 1. if you retrieve your voicemails via your phone you will played back some option like changing your busy and unavailable message (after dialing 0 - just follow the instructions). or 2. you just record your soundfiles with your favourite recorder and change them into *.gsm with sox (http://www.voip-info.org/wiki-Asterisk+sound+files) and place them into the directory of your voicemail. hope this helps pamela Use option 1. It simply just works :) Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] remote hold.
On Thursday 04 November 2004 21:45, Jon Lawrence wrote: Hi all, I'm pretty sure that there would be something (or at least some ideas) about this on the wiki - but the wiki seems to have disappeared. I'm wanting to implement a facility were by if a user dials a certain extension they can input the actual extension (ext A) they need to talk to, the call will be marked as an emergency. If a call is marked as emergency (or urgent if you like) then call other calls to 'ext A' are placed on hold and the new call is given priority. Has anyone any ideas how to implement something like this ? Jon Thought I'd add that all calls go through the asterisk box, as such * should know about all calls happening at anyone time. I'm sure I could script a connection through the manager API to hang up any calls ( a bit drastic) or transfer to a special extension. But, Is there a way to put a call on hold ? I suppose one way would be to park the call - assuming that I can make it play a specific file when the call gets parked. Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] remote hold.
Hi all, I'm pretty sure that there would be something (or at least some ideas) about this on the wiki - but the wiki seems to have disappeared. I'm wanting to implement a facility were by if a user dials a certain extension they can input the actual extension (ext A) they need to talk to, the call will be marked as an emergency. If a call is marked as emergency (or urgent if you like) then call other calls to 'ext A' are placed on hold and the new call is given priority. Has anyone any ideas how to implement something like this ? Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: How far is IAX to be a Standard
On Tuesday 02 November 2004 02:47, Steve Totaro wrote: This thread was started by Randy Bush, thought that name rang a bell. Good conversation nonetheless. http://lists.digium.com/pipermail/asterisk-users/2004-July/053278.html No, Randy didn't start this thread. He simply answered a post in the thread. I know that Randy uses * and if anyone on this list knows what it takes to get RFC's published and things started down the standards track then it's him. I'd have thought that it would be a good idea to 'use' his knowledge rather than try and hint towards him being something he's not. Just thought I'd point that out. Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T100P Caller ID UK
On Monday 01 November 2004 14:11, Alex Barnes wrote: I have to agree take that is a big slap in the face to the UK based customers/users. All I can assume is Digium don't need our money / support. Indeed. I couldn't careless about bloat in the driver - at the end of the day, the config process could be altered so that the bloat is only there for those that need/want it. I need it to work end of story. If a rolling buffer is the only way then so be it. There isn't a digium solution to connect to POTS lines in the UK other than X100P's, and I for one can't live without callerID - I'm even considering going across to ISDN so that callerID continues to work with future * versions. I have the patches against version 1 for X100P callerID, will upload them to a server at some point - when I find where I've put them. Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HandyTone 486 vs. Iaxy
On Monday 18 October 2004 17:43, Your Own ISP .com wrote: I can't believe how excited I am about a friggin piece of telecom hardware but this is getting to be adictive. What a geek ;) I am with you here, it's been a long time since I stayed up days on end without sleep just to mess with geeky stuff. It's like discovering computers for the first time again.. He He, Asterisk is basically a disease for me now that I can't get away from. Somebody help me :) A! Yep, I too am starting to have fun with this stuff. 2 x fxo moved both dects at home on to a sipura2K without swmbo noticing. I only got found out when someone left a VM :) Now to play with fax - I've got a spare ht286 to play with. After that, move the whole phone system to isdn. Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cisco router *
Hi all, Just checking that what I want to do is possible. I've got a cisco 3620 in my lab (IP plus ios). I'm thinking of moving my current pots lines to isdn BRI at home. What I'm thinking of is putting a wic-1b(s/t) into the 3620 and using that to pass incoming calls to * via sip. As I understand it this should be doable. I suppose my question is will the 3620 pay all caller ID information to * so that it can be logged. I don't yet have the BRI installed, so I can't post configs etc. TIA. Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cisco router *
On Tuesday 26 October 2004 14:13, [EMAIL PROTECTED] wrote: Won't work. The WIC-1B is a data only. You need an NM-1V (or NM-2V) and a VIC-2BRI to terminate voice and pass it via IP. Should have known Cisco would want more money from me to get this working :) It'll probably be cheaper to get a 1751V and a vic-2bri rather than a nm-2v + vic-2bri for the 3620. Going to have to do some more saving up. Plan 2. What's the best bri card to put directly in a * box in the UK. If anyone would like to quote for a NM-2V + vic-2BRI please send off list. I'm assuming that this is about the only way to get a 3620 connected to a BRI for voice ? Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hardware
On Sunday 24 October 2004 05:15, Steve Underwood wrote: Stay away from boards with Intel chipsets. Those are problematic in my experience. The FX, LX, 820, 840 and various others have been extremely flaky, and caused no end of problems. :-) VIA used to be bad, but seem to get steadily better. Intel are just erratic. I think most makers have made good and bad chipsets. Go with known good chips, not specific makers. The same goes with motherboards. So what are known good chips ?? Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Chaining more than one zap echo canceller?
On Tuesday 12 October 2004 22:58, Rich Adamson wrote: Adding resistance to one side of the line only begs for problems as it creates a tip-ring imbalance that will cause echo, etc, when other imperfections exist. If that approach works at all for anyone, its addressing a symptom and not the root cause. Try this one: Each customer loop is made up of copper and the longer the copper, the more resistance. Yet the impedance (in the US) is consistently 600 ohms. A short loop might be a 100 ohms while a long loop might be well over 1500 ohms; still both are 600 ohm impedance. That's how it should work. The resistance of a loop will change with distance, but the impendence of that loop should remain roughly constant regardless of distance. Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] One Question
On Thursday 16 September 2004 11:18, Evert Meulie wrote: Is it possible to search the archives somewhere online? Downloading all those monthly files in mbox format would be a bit too time-consuming for me... you can read a newgroup feed from www.gmane.org works pretty well. Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] internal external SIP
On Thursday 08 July 2004 22:49, Ian D. Wlloughby wrote: I am guessing the problem is that your internal clients can see the external SIP clients but not the other way round. The clients have to be able to make a physical connection to each other. You are not using any NAT capabilities I guess as your internal clients have their own network to access the server on. If you set nat on in sip.conf for one of your internal clients and get it to register on the public network, does this work? Yes, the internal clients can see the external but not the other way round. I thought that canreinvite=no meant that the clients didn't need to be able to talk directly - just be registered on the same * box. Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] internal external SIP
On Thursday 08 July 2004 23:04, Soren Rathje wrote: bindaddr = 0.0.0.0 ; Local interface externip = xxx.xxx.xxx.xxx ; Public IP address localnet = 192.168.0.0/255.255.0.0 ; All RFC 1918 addresses are local networks localnet = 10.0.0.0/255.0.0.0; Also RFC1918 localnet = 172.16.0.0/12 ; Another RFC1918 with CIDR notation localnet = 169.254.0.0/255.255.0.0 ; Zero conf local network Also, I saw some fixes to RTP address binding in CVS today. Hard to tell really without a trace.. Okay, I've made some changes. I've moved the local phones to public IP's. So now everything is connecting effectively from the internet to the * box. Things are still the same as before - I can initiate calls from local phones to remote ones. If a remote phone tries to initiate the call, the internal phone rings. When I pickup the internal phone, the call isn't completed. I've included a trace below of an incomming call. I don't know which bits are relevant so I've pasted it all. Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 82.145.37.29;branch=z9hG4bKd27e2ee695c179d0 From: Pete Murphy sip:[EMAIL PROTECTED];tag=2f9ab5f04daa9e4e To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 7711 INVITE User-Agent: Grandstream SIP UA 1.0.4.26 Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp Content-Length: 270 v=0 o=2003 8000 8000 IN IP4 82.145.37.29 s=SIP Call c=IN IP4 82.145.37.29 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 2 15 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000 a=ptime:20 12 headers, 13 lines Using latest request as basis request Sending to 82.145.37.29 : 5060 (non-NAT) Found audio format UNKN Found audio format ALAW Found audio format ULAW Found audio format UNKN Found audio format GSM Found audio format UNKN Found description format PCMU Found description format PCMA Found description format G723 Found description format G729 Found description format G726-32 Found description format G728 Capabilities: us - 524302, them - 285/0, combined - 12 Non-codec capabilities: us - 1, them - 0, combined - 0 Reliably Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 82.145.37.29;branch=z9hG4bKd27e2ee695c179d0 From: Pete Murphy sip:[EMAIL PROTECTED];tag=2f9ab5f04daa9e4e To: sip:[EMAIL PROTECTED];tag=as584623c0 Call-ID: [EMAIL PROTECTED] CSeq: 7711 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=7c6b65eb Content-Length: 0 to 82.145.37.29:5060 Sip read: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 82.145.37.29;branch=z9hG4bKd27e2ee695c179d0 From: Pete Murphy sip:[EMAIL PROTECTED];tag=2f9ab5f04daa9e4e To: sip:[EMAIL PROTECTED];tag=as584623c0 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 7711 ACK User-Agent: Grandstream SIP UA 1.0.4.26 Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE Content-Length: 0 11 headers, 0 lines Sip read: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 82.145.37.29;branch=z9hG4bKd04bc72a0d281db8 From: Pete Murphy sip:[EMAIL PROTECTED];tag=2f9ab5f04daa9e4e To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Proxy-Authorization: DIGEST username=2003, realm=asterisk, algorithm=MD5, uri=sip:[EMAIL PROTECTED], nonce=7c6b65eb, response=2d2400a30b257419c48ac5dd6747 Call-ID: [EMAIL PROTECTED] CSeq: 7712 INVITE User-Agent: Grandstream SIP UA 1.0.4.26 Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp Content-Length: 270 v=0 o=2003 8000 8000 IN IP4 82.145.37.29 s=SIP Call c=IN IP4 82.145.37.29 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 2 15 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000 a=ptime:20 13 headers, 13 lines Using latest request as basis request Sending to 82.145.37.29 : 5060 (non-NAT) Found audio format UNKN Found audio format ALAW Found audio format ULAW Found audio format UNKN Found audio format GSM Found audio format UNKN Found description format PCMU Found description format PCMA Found description format G723 Found description format G729 Found description format G726-32 Found description format G728 Capabilities: us - 524302, them - 285/0, combined - 12 Non-codec capabilities: us - 1, them - 0, combined - 0 Looking for 2000 in remote list_route: hop: sip:[EMAIL PROTECTED] Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 82.145.37.29;branch=z9hG4bKd04bc72a0d281db8 From: Pete Murphy sip:[EMAIL PROTECTED];tag=2f9ab5f04daa9e4e To: sip:[EMAIL PROTECTED];tag=as17a6c60a Call-ID: [EMAIL PROTECTED] CSeq: 7712 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL,
Re: [Asterisk-Users] internal external SIP
On Friday 09 July 2004 15:30, Soren Rathje wrote: What are your codec settings in sip.conf ?? Could you try (can be set at client level): disallow=all allow=ulaw codec's are set to allow all. I can't see how this would help. I can talk fine from local client to remote so the codecs must be correct. Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] internal external SIP
On Friday 09 July 2004 18:42, Wolfgang S. Rupprecht wrote: [EMAIL PROTECTED] (Jon Lawrence) writes: codec's are set to allow all. Thats your problem. No it's not. I'm not saying that it won't fix it - it might. I've just put my local phone back on the internal network, moved the remote phone onto a vlan that have a ipsec vpn to my internal network - guess what, everything worked. If the problem was down to me having all codec's allowed then this should not have worked - at least I don't think it should have :) I tried this too as an experiment and asterisk appears to take all to mean all codecs you can think of, not just the ones you have converters for. Instead of all you may want to try listing the codecs asterisk actually has (this is from -current): ; ; codecs: a_mu adpcm alaw g726 gsm ilbc lpc10 ulaw ; disallow=all allow=ulaw allow=alaw allow=gsm allow=adpcm allow=g726 allow=ilbc ;; allow=lpc10 (robotman) I'll try this any way - since it's something I've not tried. If this does cure my problems, I'll be throughly confused. Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] internal external SIP
In response to myself. Setting specific codecs has indeed fixed the problem. Q - how and why ? when the remote phone has a vpn directly to my internal LAN, everything works perfectly when codecs=all. But when it's connecting in from a public IP everything goes pear shaped. Can anyone give a even a clue as to why this happens ? Or is it like many other things that don't make sense - it just does :) Thanks to everyone that offered advise. Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] internal external SIP
Hi all, I've got a problem with external sip clients. My * box has 2 nics, one to my internal network and one on a public IP. There are external sip clients (on public IPs) and internal clients on the internal nic. both clients can register fine. I can phone external clients from the internal clients and the connection works perfectly. But if an external client phones an internal one, the internal phone rings, but when the phone is picked up the external call disappears. Both internal and external have canreinvite=no Can anyone give me any ideas where to start looking into this. Regards, Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk-Users List Etiquette
On Tuesday 15 June 2004 20:45, Aaron J. Angel wrote: And for those of you who don't like HTML email with different fonts or colors, etc., there's this thing called CSS . And for the rest of us, there's /dev/null which is where html email belongs. If you expect everyone else to use a client that bottom-posts, then I expect you to use an HTML-capable email client that supports CSS for accessibility. Expect away. I like many other won't have to read it 'cos it never gets as far as the inbox. Yes, I hear your reply, and you're right, switching clients is not an answer for everyone. Deal with it. Yes, switching clients isn't even an option in some situations. However configuring a client correctly is always an option. Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Caller ID with BT CD50
On Wednesday 26 May 2004 19:42, Jon Lawrence wrote: It looks like my missing digit problems are down to the dect phone I have connected to my handytone ata-286. When i have my Binatone dect connected, I only get the first 8 digits, if I connect my panasonic dect then I see all the digits - looks like I need a different dect phone :( Any ways, It looks like the patch works perfectly to me. It also works fine on my Telewest (Eurobell). I'm even more confused now. If I have the number in the phones phone book then it will show the relevant name, otherwise it only shows the first 8 digits. Has anyone ever heard of anything like this ? TIA Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Caller ID with BT CD50
On Wednesday 26 May 2004 17:07, Karl Dyson wrote: Can confirm it works with Generic X101P *BIG* Thank you :) I can confirm it works with my generic X100P (at least I think that's what it is :) ). The full callerID is put into my database, so I know it's receiving the complete CID. The phone only seems to get sent the first 8 digit's - I'm sure this is something in my configs, but I've not had chance to look into it yet. Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Caller ID with BT CD50
The full callerID is put into my database, so I know it's receiving the complete CID. The phone only seems to get sent the first 8 digit's - I'm sure this is something in my configs, but I've not had chance to look into it yet. It looks like my missing digit problems are down to the dect phone I have connected to my handytone ata-286. When i have my Binatone dect connected, I only get the first 8 digits, if I connect my panasonic dect then I see all the digits - looks like I need a different dect phone :( Any ways, It looks like the patch works perfectly to me. It also works fine on my Telewest (Eurobell). Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Caller ID Re: [Asterisk-Users] Re: Support Digium
On Saturday 01 May 2004 16:42, Gavin Hamill wrote: PLEASE PLEASE PLEASE PLEASE PLEASE PLEASE PLEASE can some Kind and Worthy soul spend a little time in getting this really important feature implemented? You would have the undying gratitude of thousands of X100P users all round the world! :D I emailed sales at digium asking whether the new module supported international (ie non bellcore) cli. The answer was yes, but it's not yet implemented in the driver - driver implementation is in the pipeline apparently. Whether this means that the detection is in the hardware or not I don't know. Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help choosing a UK IAX provider
On Wednesday 21 April 2004 21:48, Nicolas Bougues wrote: This last hop may be the source of your problem. Since I believe it's not a trans-continent link, it's either : - a very congestioned link - a router with serious problems at hop 13 (or maybe 12). You should contact whoever manages westloc.com We are aware of the problem. It is indeed a router problem and we are working on fixing it - main headache is as with all strange problems when I think I'm getting close to the cure, the problem disappears :( If it affected only one IP address (or subnet) I'd stand a chance - but the problem seems to move around our subnets appearingly at random. I'm taking delivery of a new set of routers so I can just replace both in one hit and then fault find in the lab :) Jon westloc.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 SIP Images
On Tuesday 30 March 2004 12:34, Terence Parker wrote: Wanta take a guess what would happen if Cisco decide to really enforce the legal rules? I'll bite: Their market share would plummet in all their markets, and then smaller, more innovative companies would become more able to compete with them, and the overall marketplace would be vastly improved because of more participants and more choices? B. I can't wait for that day. I don't deny that cisco make some nice products, but I don't like companies who have the attitude that since they're big and powerful they can invent whatever pricing policy they want and rip off the consumer. Of course, the argument is that as a consumer I can simply choose not to buy if I don't want to - and indeed we are now turning towards Polycom phones rather than Cisco. Cisco phones are already expensive enough - it is simply cheeky that they should have to charge further for the software that runs on the phone. That is a joke. All hardware includes software to some degree, yet one doesn't have to pay creative labs for the drivers that power their soundcards, nor Vegastream for the bundled web manage interface. And when bugs are fixed, it should be the responsibility of manufacturers to update them - the bugs shouldn't exist in the first place. Reading through some of the arguments on this thread (both pro anti Cisco) it is interesting how some feel that we should be paying Cisco the money they are demanding because it funds research and development - ironic considering this very list is about community support for a community made project. Asterisk, like many other open source projects, prove that innovation CAN and DOES take place without direct financial incentive - indeed the likes of sendmail, bind, apache etc... were around years before Microshaft came out with its equivalent tripe - and they charge piss loads for what is effectively a piece of shite. For the Cisco phones we DO have, we don't have any purchased licenses and I don't ever intend on getting any either. Cisco can sue my ass if they really want to. I have no problem with the idea of paying cisco for software that they write. In fact I have no problem with with paying for software full stop. But I'd love to have enough money to sue them if that software proved to have security issues or proved to be not fit for purpose - eg if a phone had a bug in its implementation of SIP. If people/companies want to charge for software fine (after all it takes time/money to develop) but they should be willing to take the responsibility that goes with it. Most companies don't - at least if you cantact cisco with a problem then they'll do their best to fix it or at least come up with a work-around, which is more than a certain other companies do. Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 SIP Images
On Tuesday 30 March 2004 19:01, Brian Cuthie wrote: My beef with Cisco is that the software license doesn't travel with the device. Without the license you can't buy an upgrade even if you want to. Indeed that bit is a complete joke. I can't think of anything that could be done about it though. Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Graphical Interface to display Asterisk CDR / php
On Wednesday 24 March 2004 06:49, Brian Capouch wrote: It might be helpful for us all if the author could let us know more about the environment in which this application was built. . I'm getting all kinds of errors when I try to run it, and I suspect that either my Postgres or PHP installations are incompatible with yours. My problem is with MySQL, it doesn't seem to like the queries. I think it would be useful if the author could post a list of his packages ie MySQL/PG version, php version etc. Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] UK BT caller ID revisted
Hi all, Does anyone know the procedure for adding a serial output to a cheap caller display unit. If I can find a way of doing this then I'm sure there will be away for linux to take the CallerID info, write it to a file, * to open that file an read the number from it. TIA Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: UK BT caller ID revisted
On Saturday 20 March 2004 18:51, Patrick Lidstone (Personal E-mail) wrote: In the meantime, there's some good info on hacking CID boxes here: http://www.automatedhome.co.uk/modules.php?name=Newsfile=printsid=1207 Cheers. That'll do the job. No to rip apart a few Caller ID units I've got lying around and fit out what chips are in them :) Jkn ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Should List be Moderated?
On Thursday 18 March 2004 17:21, Panny Malialis wrote: So give us a commercial list. Please :) Panny I can imagine that a commercial list could be useful for inter-provider stuff. But from a user point of view, having commercial things on this list is very useful - if a commercial provider releases a new (clever or completely different) service, it's very useful to find out about it without having to trawl around the internet. Perhaps commercial postings should be made to have [commercial] in the subject line. Then users who don't want to receive that traffic can simply filter them. Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] x100p CLI in the UK
On Tuesday 16 March 2004 08:42, Patrick Lidstone (Personal E-mail) wrote: Not quite this way, but yes. I use a Pegasus Meteor to achieve the same end result. See www.crucible-technologies.co.uk/caller-display.asp Although it is expensive, it is completely reliable - anecdotal evidence suggests that USR modems don't perform consistently with BT CID. If you are handy with a soldering iron, you could also consider modifying caller display units - they can be had for less than a fiver at The Link or similar, and just require the addition of a serial driver chip to interface them to a PC. The serial data output from these units is virtually identical to the meteor and is well documented (since it's specified and documented as part of the BT standard). Have you interfaced this with *. If so how ? Also, have you any more details on altering the caller display units ? urls would be great. TIA Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] extensions problem
Hi, I've got 2 x100p's installed in my system. Both execute the same incoming contexts as follows: [inboundA] include = dialjon [inboundB] include = dialjon|09:00-16:30|Mon-Fri|*|* [dialjon] exten = s,1,answer exten = s,2,Dial(SIP/2000,15) exten = s,3,Playback(noone) exten = s,103,Goto(onphone,s,1) snip Am I right in saying: if no one answers at ext 2000 then s,3 is executed. if ext 2000 is busy then 103 is executed. If so then sometihng is wrong. If I'm already on a call, I want 103 to be executed however, this isn't happening. If a new call comes in (whilst I'm talking on ext 2000) I here it ringing on my handset. Can anyone point out where I've gone wrong ? TIA Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] extensions problem (SIP)
On Monday 15 March 2004 16:00, Olle E. Johansson wrote: It depends on your SIP device. Asterisk places the call to your SIP device regardless, since by SIP protocol design the UA is not a slave, it is free. So the SIP ua must answer busy for Asterisk to understand that you're busy. If not, the call is placed to you and Asterisk has no knowledge that you are busy. Check you SIp phone if you can limit the number of concurrent calls. So does anyone know if the Grandstream handytone-286 sends this busy answer ? I'm guessing it doesn't. In that case, what other ways are there of connecting my dect phones to a voip * system ? - can I just connect them into the x100p's phone socket (how do I send calls to that port) or do I need to get a fxs card and run wire's everywhere - not an option :( How does everyone else connect up DECT phones to a * based system. Surely * should know if a phone is in use ? After all it initiated/took part in the call in the first place ;) Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] extensions problem (SIP)
On Monday 15 March 2004 20:35, Walker Haddock wrote: On Mon, Mar 15, 2004 at 09:28:00PM +0100, Olle E. Johansson wrote: The incominglimit limits how many simultaneous calls a UA may place to Asterisk. I'm pretty sure that the incominglimit specifies how many calls that * can send to the SIP device. If you set incominglimit=1 and then do a SIP SHOW INUSE from the *CLI then you will see the limit set. The behavior of * then will consider the device busy if there is a call in progress and the inuse count is incremented. Paul Lieu did some work on this a few months ago and I've been using it on my Cisco 7960 and Grandstream BT-102 phones. The interface to my handytone is identical to a BT-102 so it may also work with the handytone :). Where did you specify incominglimit=1 - is it in the sip.conf for that UA ? Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] extensions problem
On Monday 15 March 2004 16:26, Asterisk DEV. Mailing List wrote: Your phone supports call waiting, so isn't giving out busy. I had the same problem with a budgetone 102, you can't turn this off on the phone but you can work round it by adding Incominglimit=1 Into the sip.conf entry for the phone I can imagine situations where call waiting might be useful, but only if I can acknowledge the call with the phone either rejecting it to a queue or ditching the current call and picking up the incoming one - something to play with in the future (once I've found a way of getting UK callerID working). I've added the Incominglimit=1 and that's fixed my immediate problem. Thanks everyone. Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] UK callerID on BT line
Hi all, Has anyone found a way of extracting callerID information from a BT pstn line. I'm currently using a x100p which as we all know can't detect the callerID on a UK BT pstn line. Has anyone found any hardware which can detect the callerID which could be interfaced to the * system in some way or other. Alternatively has anyone any news on the availability of fxo modules for the TDM400 TIA, Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk CAPI DECT problem
On Thursday 11 March 2004 11:41, Ignace CARIA wrote: - Plug the DECT base into a X100P Digium Card. Plug the DECT phone into a Handytone-286 which is in turn plugged into your network. It works fine for me. HTH Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P dial in/out to sip phones
On Monday 08 March 2004 00:27, Isamar Maia wrote: Caller ID does not work in the UK, well not on my BT or Telewest line's. What I didn't understand yet about * + X100P with caller id not working in some countries is, it's a hardware or software limitation? Isamar I tihnk that it's primarily a hardware limitation. The callerID on a UK BT line is sent before the first ring tone. It is initiated by a line reversal - from what I can gather the x100p can't detect this. There are supposed to be some new modules coming out for the TDM400P which will work - but then they were supposed to be out before Christmas - does anyone know when/if they are going to get released. Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P dial in/out to sip phones
On Sunday 07 March 2004 20:08, Simon Chappell wrote: Thanks for your help David Your configs are a little to complicated for this complete asterisk newbie though. All i am actually after is how to get a sip phone to ring when the X100P is dialed on out landline, and how to get a sipphone to dial out through the X100P. I have saved all your configs and had a trawl through them though. I am a great believer in start simple then build it up and step by step it seems simple in the end but I keep stumbling on this task. once i have this i will look at call parking,conferencing (all the fun stuff) etc.. but at the moment all i would like to acheive is bridging the gap from sip to BT :-)IF you have any quick pointers to help me acheive that I would be very pleased. Thanks again for taking the time to reply (especially on a sunday evening with the roast going cold) I've emailed you my configs off list :) Like you, I'm not yet looking to do anything complicated with my asterisk setup and I've just finished implementing exactly what you're trying to do. HTH Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P dial in/out to sip phones
On Sunday 07 March 2004 20:08, Simon Chappell wrote: Thanks for your help David Your configs are a little to complicated for this complete asterisk newbie though. All i am actually after is how to get a sip phone to ring when the X100P is dialed on out landline, and how to get a sipphone to dial out through the X100P. I have saved all your configs and had a trawl through them though. I am a great believer in start simple then build it up and step by step it seems simple in the end but I keep stumbling on this task. once i have this i will look at call parking,conferencing (all the fun stuff) etc.. but at the moment all i would like to acheive is bridging the gap from sip to BT :-)IF you have any quick pointers to help me acheive that I would be very pleased. Thanks again for taking the time to reply (especially on a sunday evening with the roast going cold) Simon Hopefully the attached configs will be of help to you. They're pretty basic :) The one's you'll be interested in are zapata.com, extensions.conf and possibly sip.conf. HTH Jon Lawrence asterisk_configs.tgz Description: application/tgz
Re: [Asterisk-Users] X100P dial in/out to sip phones
On Sunday 07 March 2004 21:28, Simon Chappell wrote: thanks so much.. I have dialed from my mobile and nearly fell off my chair when the Sip phoone rang ,,!! then was sad enough to answer it and have a chat with myself!! Is there any provision for dialing out in these configs ? and if so is it dial 9 ? Thanks again as this has been a four day headache so far.. To dial out, simply prefix by *2 Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] x100p Q.
Hi everyone. I've now managed to my basic voip setup working, but I have a problem with my fxo cards. If I plug the cards into the pstn line whilst a normall phone is also plugged in, the normal phone continually rings. I'm convinced that this is a problem with the wiring but I don't know what/why. The * box works perfectly (with the exception of the callerid) so long as I don't have another phone plugged in. I can't just unplug all the other phones - the sky box + alarm system must remain plugged in. I can still ring out on the other phones and also on the * box, but the constant ringing is obviously a problem :) Is this normal ? Has anyone else seen this ? fyi I'm based in the UK. TIA Jon Lawrence ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] x100p Q.
On Saturday 06 March 2004 11:16, you wrote: I was having a similar problem with all the none asterisk'd phones ringing, but it was only occasionally. I solved the problem by plugging everything through asterisk! *8-) I'd be surprised if the alarm can't also be configured in a similar way. It probably can, but it's hardwired and would cost to have it rewired. There must be a way around this - surely an x100p can't expect to be thev only thing on the line. Darren. PS, is anyone working on CallerID in the UK? It would be very handy, but I don't know enough about programming/phones to do anything bar testing! From what I've read, the fxo modules for the tdm400p should solve the problem - if they ever get released. Jon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
