Re: [Asterisk-Users] AudioCodes MP108 8-Port FXO Analog Gateway (SIP)
For the FXS unit:
1) it doesn't recognise voicemail waiting messages, so your analog
phones won't receive a stuttered dial tone.
2) it doesn't seem to recognise the transfer (#) button since it seems
to use different payload numbers (rtp codec 100
and 96). We will be submitting an email shortly to the bug tracker database.
As long as you have the coefficients file defined for your region then call
handling should be fine. Our gateways are configured for uk use.
Thanks
Tan
telappliant.com
- Original Message -
From: "Ing. Angel Gomez Garcia" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, August 21, 2003 6:39 AM
Subject: Re: [Asterisk-Users] AudioCodes MP108 8-Port FXO Analog Gateway
(SIP)
Hello Ernest.
I'm setting up two * boxes using mp108 ( FXO and FXS ), and it is
working fine. The only issue with the FXO box is that it does not
support remote disconnect supervision so you have to make sure that the
reorder tone ( wich is used for disconnect ) is adequate for your country.
I have this in sip.conf:
--
[mp108out]
type=friend
host=x.x.x.x; <-- Fixed ip assigned to the mp108
dtmfmode=inband
[mp108in] ; <-- mp108 configured to register with user
mp108in
type=friend
host=dynamic
dtmfmode=inband
context=inbound
And this in extensions.conf
[turnklongdistance]
exten => _91NXXNXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten => _91NXXNXX,2,Congestion()
exten => _91NXXNXX,102,Congestion()
[longdistance]
ignorepat => 9
include => trunklongdistance
...
...
[inbound]
exten => s,1,Wait(2)
exten => s,2,Answer()
. ; Basically your ivr main menu
exten => 100,1,Goto(s|1)
exten => 200,1,Dial(ZAP/2-1)
; Handling of exceptions
... ; More extensions
---
on inbound calls you have to configure the mp108 to forward incoming
calls to extension 100.
on outbound calls you have to configure the mp108 to One step dialing.
choose the order of your codecs and i think thats it.
Good luck.
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[Asterisk-Users] Configuring iptables to allow sip and dynamically allocate rtp ports
Hi, We have an asterisk box, with 2 nics, one with internal addressing, and the other with a public address. The firewall (iptables) is configured for nat routing. Now we want to allow this box to receive sip registrations from the internet. Does anyone know if you can use iptables to allow the dynamic creation of rtp ports? T
Re: [Asterisk-Users] Grandstream Budgetone
RRP: $75 for 101, $85 for 102 - Original Message - From: <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Sunday, August 17, 2003 1:22 PM Subject: [Asterisk-Users] Grandstream Budgetone Does anyone know what the Grandstream Budgetone is going for $$$ in the US? I didn't immediately see pricing on the phones page. AJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Does Wildcard x100p support BT Caller ID in UK?
We use it, but with no caller id. Tan telappliant.com - Original Message - From: "Dave Wilson" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, August 05, 2003 1:53 PM Subject: [Asterisk-Users] Does Wildcard x100p support BT Caller ID in UK? Hi all, I can't seem to find any info on this anywhere on the web, except that BT caller ID doesnt use the standard bellcore system in use in the US. So, if anyone here in the UK is onlist and using the x100p successfully, please let me know. TIA, Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Grandstream power supplies..
We do the replacement adapters for £12+VAT if interested. You'll still need the US-to-UK adapter though. Contact me offline if you need one. You could go to cpc as was suggested. www.cpc.co.uk Tan telappliant.com - Original Message - From: "WipeOut ." <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, August 12, 2003 1:09 PM Subject: [Asterisk-Users] OT: Grandstream power supplies.. Hi, Quick question to all the electronics gurus out there.. I unpacked my second GS phone yesterday (had it for about a month!) and set it up.. This morning the power supply is dead.. I have looked for a new one online (In the UK using Maplin let me know if you know a better place.) becasue it would probbaly take too long to get one sent from China or the US and I need to get that station operational again.. There seem to be many choices for power supplies.. Looking on the bottom of the broken one it is a 5VDC 400mA output.. When I looked online for a new one the choices are for regulated, unregulated and switch mode power supplies with the regulated and switch mode ones being VERY much more expensive than the unregulated ones.. Which kind would do the job?? Also most of the variable output adapters are 4.5v or 6v, Not the required 5v.. Any help would be appreciated.. Thanks.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Running Asterisk behind NAT?
You can run SIP reliably behind nat with a SIP-aware adsl router. We have set this up for a few customers, as well as for ourselves and it works fine. http://www.telappliant.com/intertex_sip_aware.htm. Tan - Original Message - From: <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, August 12, 2003 5:35 AM Subject: RE: [Asterisk-Users] Running Asterisk behind NAT? Yes, you can run Asterisk behind a NAT. NO, you CAN'T (reliably, easily) run SIP behind a NAT. For FWD think about using their behindnat and fwdproxy addresses. Maybe a STUN would help. Also, test your setup infront of NAT also, make sure they work, before you head behind a NAT. -- wasim This mail is confidential & intended solely for the use of the addressee. On Tue, 12 Aug 2003, Terence Chan wrote: > I would like to ask if it is possible to run Asterisk behind NAT. I have a > linksys router that forwarded the port UDP 5082 to the local IP of my > Asterisk box, I got the error 479 when I try to register my Asterisk box > with FWD. (see detail below). > > Have anyone got Asterisk working behind NAT and successfully registers with > FWD? > > Any pointer or information will be appreciated. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Bad sound quality with G729A on SNOMs
Hi, We are testing with G729 from remote offices to a central asterisk machine. With a SNOM 200 the g729 is terrible. We notice the following: 1) when dialling voicemail, the first part of the announcement is missed. 2) the sound is very quiet, and sound quality is terrible (tinny sound, humming in background). Has anyone else had the same problems?
Re: [Asterisk-Users] Snome-200 with Asterisk
Hi, Some questions: 1) Are you behind NAT? 2) If the answer to (1) is Yes, then be aware that if you have the latest firmware (1.16w) then you should choose the the appropriate setting under "NAT detection". The "Automatic" setting doesn't seem to work some of our customers behind nat. There are options such as UPnP and STUN and you will have to choose the appropriate one. 3) If you aren't behind NAT, then my guess is that you have a codec issue. Tan telappliant.com - Original Message - From: denzel-infotechs To: [EMAIL PROTECTED] Sent: Friday, August 08, 2003 9:46 AM Subject: [Asterisk-Users] Snome-200 with Asterisk hi We are using snome 200 IP phone with *. It works OK. But after a period of time we can't hear any sounds for any icoming or outgoing calls. I've got two of these phones. Same symptoms occur to both of these( not at the same time ) and the problem remains until the phone is completely rebooted. Don't know whether this's * or Snomes' prob. Any help would be appreciated. Thanks In advance. Denzel.
Re: [Asterisk-Users] Grandstream Budgettone 100 & 102
These guys charge £79+VAT for the 102, and that includes postage to anywhere in the UK. The $75 doesn't include the VAT tax which has to be paid on top if shipping to places like the uk. Tan (telappliant.com) - Original Message - From: "Skuse, Phil" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Thursday, July 31, 2003 9:48 AM Subject: RE: [Asterisk-Users] Grandstream Budgettone 100 & 102 We bought two 100's for $75 each, and IIRC they charged an extra $100 or so for shipping to the UK (which seemed a little excessive to me - I asked our finance people to look into it). -Original Message- From: Reed Wade [mailto:[EMAIL PROTECTED] Sent: 31 July 2003 06:08 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Grandstream Budgettone 100 & 102 With shipping, I recall my 102 came to $97. I think it was $85 but I'd need to look it up and don't have the papers nearby. -reed At 06:39 PM 7/30/2003 -0500, you wrote: >I was quoted $75 and $85 USD today. > >Ricardo Villa >http://www.telesip.net > >- Original Message - >From: "Joe Cooke" <[EMAIL PROTECTED]> >To: <[EMAIL PROTECTED]> >Sent: Wednesday, July 30, 2003 6:31 PM >Subject: Re: [Asterisk-Users] Grandstream Budgettone 100 & 102 > > > > I was quoted the $75 and $85 USD prices from Grandstream direct about 2 > > months ago. I'm not sure if it makes a difference, but I live in the US. > > > > - Joe > > - Original Message - > > From: "marrandy" <[EMAIL PROTECTED]> > > To: <[EMAIL PROTECTED]> > > Sent: Wednesday, July 30, 2003 7:17 PM > > Subject: [Asterisk-Users] Grandstream Budgettone 100 & 102 > > > > > > > > > > Checking the earlier mails, it stated that the phones were $75 (100) & >$85 > > > (102) ref :- > > > > > > http://lists.digium.com/pipermail/asterisk-users/2003-June/013483.html > > > > > > Well, I just called Ovislink/dgtimes and was quoted $90 & $100 and the > > person > > > said there was no price change. > > > > > > Anyone on this list actually bought them at the $75 & $85 rate ??? > > > > > > Regards...Martin > > > -- > > > Too much is just enough. > > > -- Mark Twain, on whiskey > > > > > > ___ > > > Asterisk-Users mailing list > > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] UK call termination..
We use our own gateway for h323 and sip shortly. Contact me offline. Tan - Original Message - From: "WipeOut ." <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, July 21, 2003 1:43 PM Subject: [Asterisk-Users] UK call termination.. Hi, I am looking for call termination in the UK so that I can place calls via my internet line instead of buying more PSTN lines.. anyone know of amy providers in the UK.. somthing like nufone.net in the UK would be perfect.. Later.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Authentication bug?
Hi, I don't know whether only we are experiencing this problem but it seems that if authentication is used on a couple of phones, and then the authentication is removed (i.e. remove the secret parameter from each of the extensions), then this isn't reflected in asterisk after a reload. Instead we actually have to restart asterisk for the authentication to be removed. Has anyone else seen this? Tan
Re: [Asterisk-Users] New budgetone firmware
Has anyone noticed that sometimes when you boot the phone you don't hear a dialling tone? I've logged the issue with Grandstream nevertheless. Tan - Original Message - From: "WipeOut ." <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, July 15, 2003 11:17 AM Subject: RE: [Asterisk-Users] New budgetone firmware Thanks for that I will give it a try.. only problem is that I have some extensions that use a # which is going to cause a problem.. I guess I will have to rethink these.. The ability to transfer is more important.. Maybe I will be able to just replace # with * in the dialplan.. :) Later.. > For me works, providing that: > > * 'Use # as Dial Key:' must be set to yes > > And to trasnfer you do the following: > receive a call > press trasnfer > dial the exten + # > > the call is trasnferred now. > > if you simply dial the exten + hangup as stated in the manual, > that doesn't work (also if 'Use # as Dial Key:' is set to no) > > Matteo. > > > Il mar, 2003-07-15 alle 10:46, WipeOut . ha scritto: > > Yes, The External NTP issue has been around for a while now.. I was hoping it would be fixed in this release.. > > > > Also the ability transfer a call using the "Transfer" button is still broken.. (Unless it requires some special configuration to make it work with *)... Anyone know?? > > > > These problems are not listed in the "Known Problems" section of the release notes for .77 release.. > > > > > > > > > Hi > > > > > > I upgraded earlier today and so far have found that if the Daylight Saving > > > option is on one hour is added to the time received from the NTP server > > > regardless of date. > > > > > > This is using my internal NTP server but I can't get it to work with any > > > external NTP server, it simply does not download the date > > > > > > Other than this I have seen no change since I upgraded. > > > > > > Although there appears to be no Release notes with this release > > > > > > Regards > > > Paul > > > > > > -Original Message- > > > From: [EMAIL PROTECTED] > > > [mailto:[EMAIL PROTECTED] Behalf Of Brancaleoni > > > Matteo > > > Sent: 14 July 2003 20:42 > > > To: [EMAIL PROTECTED] > > > Subject: [Asterisk-Users] New budgetone firmware > > > > > > > > > Hi. > > > Has anyone experienced with the new firmware .77 ? > > > There's Day Light Saving time now, but haven't > > > time to play with it, till now. > > > > > > Matteo. > > > > > > -- > > > Matteo Brancaleoni > > > Espia System Administrator - IT services > > > Website : http://www.espia.it > > > Email : [EMAIL PROTECTED] > > > > > > > > > > > > ___ > > > Asterisk-Users mailing list > > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > ___ > > > Asterisk-Users mailing list > > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] caller id
Use SetCallerID(1234567). Tan telappliant.com - Original Message - From: "Marian Danisek" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, July 09, 2003 3:23 PM Subject: [Asterisk-Users] caller id Hello, is it possible to change how are caller id on incoming call from isdn, capi lines displayed od sip phones ? ( e.g. SNOM ) standard is [EMAIL PROTECTED] I just want only 1234567 to be displayed. is it possible ? regards Marian -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/ A mind is like a parachute... it only works when it's open. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Direct entry to your own voice mailbox
e.g.
exten => 8501,1,VoiceMailMain2(${CALLERIDNUM})
Tan
telappliant.com
- Original Message -
From: "Dan" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, July 07, 2003 4:47 PM
Subject: [Asterisk-Users] Direct entry to your own voice mailbox
Hi,
There is any possibility to dial a specific extension and then enter in your
own mailbox (the one defined for that specific SIP phone) without asking for
the exxtension number but only for the password?
I want to be the same extension for all phones, not a specific one for each
of them.
It is possible to have a time stamp in the recorded message? I want to know
when the message has been recorded.
Thanks,
Dan
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Re: [Asterisk-Users] Accurate Billing
Steve, For analog, isn't it just a case of getting asterisk to listen out for specific tones such as busy, or "ringing". Isn't this what the "callprogress" option is for in zapata.conf? I thought that it works for the US at the moment but no where else. Tan - Original Message - From: "Steven Critchfield" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, July 07, 2003 6:01 AM Subject: RE: [Asterisk-Users] Accurate Billing On Sun, 2003-07-06 at 23:17, Kim C. Callis wrote: > Steve, > > What exactly would be classified as a digital ZAP device? T1/E1 interfaces, so T100P, E100P, T400P, E400P If you need to see examples, I could probably dig up CDR records where busy is indicated, and where no answer is indicated and there is a definate difference between call duration and stop-start duration. > > -Original Message- > > From: [EMAIL PROTECTED] [mailto:asterisk-users- > > [EMAIL PROTECTED] On Behalf Of Steven Critchfield > > Sent: Sunday, July 06, 2003 8:58 PM > > To: [EMAIL PROTECTED] > > Subject: Re: [Asterisk-Users] Accurate Billing > > > > On Sun, 2003-07-06 at 22:07, [EMAIL PROTECTED] wrote: > > > hi everyone, > > > > > > I know this issue has been raised many times before, i think still > the > > > problem remains. When a call is made through a Zap channel, whether > it > > > is actually made or not (irrespective of whether, engaged, busy, or > > > actually answered), asterisk logs it in CDRs as a call made. This > > > makes it impossible to do an accurate billing. Has anybody found a > way > > > to overcome this problem, if yes, please let me/us know. > > > > If you are on a digital Zap interface, then it is known. If you are on > > an analog interface, then there is no way to know the other answered > or > > not. > > > > -- > > Steven Critchfield <[EMAIL PROTECTED]> > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and Hot Desks??
How about the logon wizard of the snom 100? I think that does something simlar to what you want. It's designed to allow different people to login to a single phone. - Original Message - From: "WipeOut ." <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, July 02, 2003 11:21 AM Subject: [Asterisk-Users] Asterisk and Hot Desks?? Hi, Has anyone worked out a way to use Asterisk in a Hot Desk environment?? I have not been able to think of a way for the user to have control over which IP phone will ring when that users extension is dialed without the user needing to reconfigure the phone.. Something like this would be cool.. User dials *8555 (or similar) and is prompted to enter their extension and then password, after successfully validating the user is then prompted for phone number (being some IP phone ID number or an external Mobile or Home phone number).. All calls made to that users DID number or extension are now routed to the registered destination device.. Any calls the user makes from any IP Phones carry the correct caller ID information as well.. Anyone got something like this or any form of user manageable extension control or hot desk type solution working?? Or any suggestions how it could be archived?? Later.. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with echo
Could you provide details of which sip phones you are using. For instance, the SNOM 200 has echo problems on firmware ver 1.16b. Upgrading to 1.16k resolves most of the echo issue. Tan (telappliant.com) - Original Message - From: "Dave Packham" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, July 01, 2003 5:15 PM Subject: Re: [Asterisk-Users] Problem with echo Same prob here. 15 SIP phones only get eco when going to the PSTN... if you find something let me know Dave >>> [EMAIL PROTECTED] 7/1/2003 8:53:13 AM >>> Hello, I can't have asterisk working without echo when I place a call from IP phone (SIP or H323) to a PSTN Phone. The called number as no problem with echo but there is a very audible echo in the SIP phone. This situation occurs either when connected with ISDN card thru i4linux driver and with my openline card from voicetronix. Do you have any suggestion fo that? Regards, Daniel ANDRE -- Daniel ANDRE (mailto:[EMAIL PROTECTED]) IRIS Technologies - http://www.iris-tech.com Serveur kwartz - http://www.kwartz.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip call pickup ?
Just want to know if this feature was implemented. Also, how do I do a supervised transfer with sip phones? Tan - Original Message - From: "Mark Spencer" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Thursday, March 13, 2003 5:26 PM Subject: Re: [Asterisk-Users] Sip call pickup ? This feature is in development currently. Mark On Thu, 13 Mar 2003, Matteo Brancaleoni wrote: > Hi, I have a mixed sip-zap evironment > in my office. I was wondering if is > possible to do remote call pickup > from a sip phone, like from zap. > > Any hint? > > Matteo Brancaleoni > [EMAIL PROTECTED] > Emmegi System Administrator > > EspiA - EMMEGI Srl - e*solution provider > Uffici: Via Pascoli, 37 > 20129 Milano - Italy > Sede Legale: Corso Sempione 67 > 20149 Milano - Italy > Tel. +39 0270633354 > Fax. +39 0245487890 > http://www.espia.it > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Minimum budget question ...
Hi, Yes, each port can be addressed by *, as it behaves like a separate sip / h323 endpoint. The "connector" is just a way to allow the voip box to have 24 connections, and they are just standard rj11 connections. Another way is just to use 3 x 8-port gateways on separate IP addresses. You can use g729 and run * either with "safe_asterisk", or using the screen command e.g. screen -d -m asterisk -vvvc. Contact me offline for pricing info. Tan [EMAIL PROTECTED] - Original Message - From: "Andy Powell" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, June 30, 2003 7:51 PM Subject: Re: [Asterisk-Users] Minimum budget question ... Hi Tan, Thanks for the reply. I'll end up asking a load more questions now... What sort of prices are we talking about for the 24 port VoIP gateway? I assume that each port is individually addresable by *? As I recall the 24 port gateways tend to be terminated at the FXS side as some 'wierd' connector (wierd in that it's not rj45/11) do you just wire this to a patch panel? What codec is in use to get all 24 ports 'running' at the same time..G729? Does this cause problems since iirc * needs to run in console mode for the G729 codec to work properly Thanks for the info... interesting site too :D Andy *** REPLY SEPARATOR *** On 30/06/2003 at 19:21 Tan Aks wrote: >Hi, > >We provide asterisk-based solutions to customers based in the uk. One of >our >customers (9 users) is trialling our low-end solution which comprises of a >box with 2 x X100P (analogue line) cards installed, and a voip carrier for >outgoing calls. This customer intends to have 13 extensions in his "live" >scenario. The way to use multiple analogue phones is: > >1) get a T100P card and use a T1 channel bank sourced from the US >2) use a couple of TDM400P cards to give 8 extensions, and use IP >phones for the other extensions >3) use a voip gateway to provide up to 24 x analogue extensions per >IP address. VoIP gateways are commonly available and convert analogue lines >into a SIP/H323 VoIP stream. > >You can get an E1 terminated with an RJ45. If you have a coax termination >then you can use a balun to get rj45 connectivity. > >Hope that helps. >Tan (telappliant.com) > > > > >- Original Message - >From: "Andy Powell" <[EMAIL PROTECTED]> >To: <[EMAIL PROTECTED]> >Sent: Monday, June 30, 2003 5:26 PM >Subject: RE: [Asterisk-Users] Minimum budget question ... > > >Tim, > >a good comprehensive answer to the question...certainly gave me a few >things >to think about. I do have a few questions though, since I'm in Europe. > >Has anyone in Europe set up something equivalent to what Tim suggested? > >What sort of prices did it work out at? > >How did you solve the channel bank 'issue' in Europe? > >I keep reading that E1 lines are coax terminated, is this correct or do you >usually get a choice from your teleco? > >Were there any other issues to contend with? > >I'd certainly be interested in the experiences of anyone in Europe... > >Thanks > >Andy > > > > >On 30/06/2003 at 10:55 [EMAIL PROTECTED] wrote: > >>If this is for commercial use, especially if you are going to be selling >>this solution, I would suggest that you don't even offer the choice of >>analog lines except in the smallest of offices. Unless you like to >>spend a lot of unbillable time supporting them :) >> > > > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users > > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Minimum budget question ...
Hi, We provide asterisk-based solutions to customers based in the uk. One of our customers (9 users) is trialling our low-end solution which comprises of a box with 2 x X100P (analogue line) cards installed, and a voip carrier for outgoing calls. This customer intends to have 13 extensions in his "live" scenario. The way to use multiple analogue phones is: 1) get a T100P card and use a T1 channel bank sourced from the US 2) use a couple of TDM400P cards to give 8 extensions, and use IP phones for the other extensions 3) use a voip gateway to provide up to 24 x analogue extensions per IP address. VoIP gateways are commonly available and convert analogue lines into a SIP/H323 VoIP stream. You can get an E1 terminated with an RJ45. If you have a coax termination then you can use a balun to get rj45 connectivity. Hope that helps. Tan (telappliant.com) - Original Message - From: "Andy Powell" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, June 30, 2003 5:26 PM Subject: RE: [Asterisk-Users] Minimum budget question ... Tim, a good comprehensive answer to the question...certainly gave me a few things to think about. I do have a few questions though, since I'm in Europe. Has anyone in Europe set up something equivalent to what Tim suggested? What sort of prices did it work out at? How did you solve the channel bank 'issue' in Europe? I keep reading that E1 lines are coax terminated, is this correct or do you usually get a choice from your teleco? Were there any other issues to contend with? I'd certainly be interested in the experiences of anyone in Europe... Thanks Andy On 30/06/2003 at 10:55 [EMAIL PROTECTED] wrote: >If this is for commercial use, especially if you are going to be selling >this solution, I would suggest that you don't even offer the choice of >analog lines except in the smallest of offices. Unless you like to >spend a lot of unbillable time supporting them :) > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] where to get adsi phones in europe ?
We sell the CE approved versions of the PT390. Contact me offline and I'll give you details. Tan - Original Message - From: "Thomas Haeger" <[EMAIL PROTECTED]> To: "Asterisk User" <[EMAIL PROTECTED]> Sent: Friday, June 20, 2003 4:33 PM Subject: [Asterisk-Users] where to get adsi phones in europe ? Hi all, have anybody an idea where to get adsi phones in europe ? Thanks, Thomas. *** beroNet technologies GmbH Dipl.- Ing. Thomas Häger Potsdamer Str. 18 A 14513 Teltow FON:+49 (0) 3328 3077731 FAX:+49 (0) 3328 334779 Email: [EMAIL PROTECTED] *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Active ISDN PCMCIA card
We use and sell the AVM B1 PCI V4.0 card. It seems to work well with asterisk apart from slight echo that I noticed when receiving an isdn --> * --> remote sip phone call. Tan - Original Message - From: "Michael Manousos" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Friday, June 20, 2003 12:28 PM Subject: [Asterisk-Users] Active ISDN PCMCIA card Are there any suggestions for active ISDN CAPI PCMCIA cards that are known to work with Asterisk? Thanks, Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Billsec on CDR
Isn't there any way to make callprogress work for people in Europe? What is it that is needed to make it work? T - Original Message - From: "Martin Pycko" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Thursday, June 19, 2003 11:36 PM Subject: Re: [Asterisk-Users] Billsec on CDR It has to do with the fact that with analog channels like FXO we don't have a way to tell whether the call has been answered or not. So after the interfaces sends the called number we assume that the call got answered. This happens unless you have callprogress=yes in zapata.conf. But it's designed to be working only in US. Martin On Thu, 19 Jun 2003, Dan Fernandez wrote: > I have an X100P and when I place calls to the PSTN which are not answered, the Billsec field of the CDR still logs the seconds that the phone rang. > > Can someone please confirm that this has to do with the ringcadance of the indications.conf file? Is there anything else I need to check ? > > Thanks in advance > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SNOM 200 and MWI??
Yep, this works, but on the SNOM 200 when you press the MWI button, it seems to have the effect of turning off the MWI light, even if I cancel the call without actually reading my message. On the odd occasion the light remains on after cancelling the call. I'll have to inspect the sip debug. Has anyone else come across this? T - Original Message - From: "Martin Pycko" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Wednesday, June 18, 2003 9:01 PM Subject: Re: [Asterisk-Users] SNOM 200 and MWI?? You could always have exten => asterisk,1,VoicemailMain Martin On Wed, 18 Jun 2003, Test wrote: > Does anyone know if this was implemented? If not then where should I look to > try and make the mod? > > Thanks > Tan > > > > - Original Message - > From: "WipeOut ." <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Monday, April 28, 2003 9:03 AM > Subject: Re: [Asterisk-Users] SNOM 200 and MWI?? > > > Hi Mark, > > If you decied to impliment this please let me know when it is done.. > > Thanks.. > > > Sounds like the SNOM expects to use our "Contact" to get a hold of us. It > > should be simple to add something like "voicemail=" in the general > > section for setting the voicemail extension to use in the contact area. > > > > Mark > > > > On Thu, 24 Apr 2003, WipeOut . wrote: > > > > > Here is the trace if anyone is interested.. > > > > > > NOTIFY sip:[EMAIL PROTECTED] SIP/2.0 > > > Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK523b1b63 > > > From: "asterisk" ;tag=as3da6a846 > > > To: > > > Contact: > > > Call-ID: [EMAIL PROTECTED] > > > CSeq: 102 NOTIFY > > > User-Agent: Asterisk PBX > > > Event: message-summary > > > Content-Type: application/simple-message-summary > > > Content-Length: 36 > > > > > > Message-Waiting: yes > > > Voicemail: 1/0 > > > > > > > > > > Hi, > > > > > > > > The MWI is working on the SNOM 200 but the problem is that when you > press the MWI button it attempts to dial > > > > "asterisk" > > > > where 192.168.1.200 is the IP address of my * box. > > > > > > > > How can I modify this so the return path is correct, which on my setup > is extension 8500 for voicmailmain?? > > > > > > > > Thanks > > > > -- > > > > __ > > > > http://www.linuxmail.org/ > > > > Now with e-mail forwarding for only US$5.95/yr > > > > > > > > Powered by Outblaze > > > > ___ > > > > Asterisk-Users mailing list > > > > [EMAIL PROTECTED] > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > -- > > > __ > > > http://www.linuxmail.org/ > > > Now with e-mail forwarding for only US$5.95/yr > > > > > > Powered by Outblaze > > > ___ > > > Asterisk-Users mailing list > > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > __ > http://www.linuxmail.org/ > Now with e-mail forwarding for only US$5.95/yr > > Powered by Outblaze > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
