Dr. Rich Murphey wrote:
Has anyone approached VoicePulse Connect about forwarding Caller Name?
Given the interest, it could be an effective way to differentiate service
among competitors.
Rich
They might not be getting the NAME on calls. NI1 providioning on a PRI only
provides CID Number.
To get
That should be
exten => 911.,1,blah
and
exten => 9911.,1,blah
You don't want to not catch a call when the user is scared to death and hits too many
1's.
John
Steve Totaro wrote:
its probably a good idea to put extens for both 9911 and 911. all bases
covered and no need for stickers.
- Origina
Bill Merriam wrote:
I am trying find a way to help the local Kerry campaign and it occurs to
me that VOIP and Asterisk could be a big help. I have never worked on a
Bill,
You'll find that the FEC has VERY strict guidelines regarding things
like this.
John
_
Hariharan Gopalan wrote:
Hi all
Was wondering if anyone is aware of a colo provider who can terminate a 1800 phone line to my box in their colo. I just need one or may be two phone lines with the same 1800 number to go to my asterisk box.
Thanks for any help
Hariom
-
Brian Wilkins wrote:
Hi,
We are having an issue here. It seems that whenever we initialize Asterisk
on our network, the router that the Asterisk server is connected to (Cisco
7200) crashes and loses it configuration. This has happended five times and
each time we have tested it, it is always
Todd Lieberman wrote:
Asterisk and dial-up modemsLook at the ZapRAS 'show application ZapRAS'
this only work w/a PRI. TL
And from what I've seen, this will only work ISDN -> ISDN. Note the
fact that there is no modem emulation in ZapRAS.
John
___
Aste
Jean-Yves Avenard wrote:
I posted two days ago a problem I was facing that calls made over the
Internet to my Asterisk gateway would hang-up after just 5s (no NAT were
involved)
Answer I got was: it's your config
Well, it wasn't (as I was expecting).
I compiled Asterisk under a Linux RedHat 9 PC
Kevin P. Fleming wrote:
Andrew Kohlsmith wrote:
Do you have information on how to do this? This is *precisely* what I
want to do. I assumed you set this up with your telco and then set
the caller ID to the # matching the address you wanted, leaving the
telco to do the address match.
In discus
Andrew Kohlsmith wrote:
On Thursday 17 June 2004 11:38, John Fraizer wrote:
If you have PRI service into your * server, it is possible - though not
always easy - to set the ALI database information specific for each ANI
(DID number) that you use. I do this with our PRI's. Depending on
Joe Baptista wrote:
I understand that most VoIP providers allow for 911 calling but that 911
service is not the same as that available to PSTN.
From what I understand a 911 Call Will Go To A General Access Line at the
Public Safety Answering Point (PSAP). This is different from the 911
Emergency Re
oi geli wrote:
I want to buy a 7940 to use with Asterisk. Does all
the features (i.e. Transfer, Hold, call waiting, MWI,
etc)work?
How difficult it is to configure 7940?
Works great. It takes about 5mins to configure if you know what you're
doing. Maybe an hour if you're learning as you go.
Eve
Randy Bush wrote:
Time for Duane to start implementing DNS SRV, since it's from now on is
turned on by default in CVS head.
Unless you're planning on breaking other standards my A records will
keep on working just fine :)
except you (likely to be ex-) customers will have problems reaching
more an
Duane wrote:
The documentation, well what documentation there is, simply isn't
coherent enough, or detailed enough to explain these things, and the few
lines in the config file certainly doesn't explain anything either...
;srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;
> exten => 555,1,Dial(SIP/1000,30)
> exten => 555,102,Dial(SIP/2000,30)
> exten => 555,103,Dial(SIP/3000,30)
> exten => 555,104,Voicemail2(u3278)
> exten => 555,105,Hangup
> exten => 555,2,VoiceMail2(u3278)
> exten => 555,3,Hangup
...should be
exten => 555,1,Dial(SIP/1000,30) ; Unanswered = 2,
exten => _NXXNXX,1,Dial(Zap/g1/${EXTEN})
exten => _NXXNXX,2,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
exten => _NXXNXX,3,Congestion
The above will attempt to dial out your Zap interface first. If that
fails, it will dial out using "username" for the username and the
password, IP address
Duane wrote:
Chris Glover wrote:
I use voiptalk via my DSL connection. It seems to work very well.
Originally I was using the connection in Sip mode, but had problems with
DTMF, I could only get it to work on outgoing calls, or incoming if I
changed mode, but not both. I switched to using IAX last
I just noticed that incoming calls don't show up in user specific CDR files.
For example, if in sip.conf, you have the following entry:
[123]
callerid="Joe Blow" <123>
type=friend
username=123
secret=456
[EMAIL PROTECTED]
host=dynamic
context=123
canreinvite=no
dtmfmode=rfc2833
nat=yes
accountcode=
They do too have a "ToVoiceMail" button. It's the one that looks like
an envelope. You just have to set the phone up to use that button.
John
Maveric wrote:
What type of cisco phones? i'm using 7960's and i know they don't have
a to voice mail button. That annoys me.
At 02:59 PM 6/4/2004, y
Tony Hoyle wrote:
Eric Wieling wrote:
Why are you even looking at VoIP? Analog ports and phones are pretty
cheap. They are not "pretty", but they are cheap and all the smarts are
in the PBX.
Free calls to the US, basically, since the leased line is dirt cheap to
run. ie. the purpose of the exe
Senad Jordanovic wrote:
In United Kingdom, we have time based dialling pricing from most of
Telco's
based on time the call is placed! It is called PEAK (08.00- 18.00
Mon-Fri), OFF PEAK(18.00-08.00 Mon-Fri) and WEEKEND (all other times!
Could someone from any of other countries let me know if time b
Patrick Lidstone (Personal e-mail) wrote:
Please excuse me if this is a niaive question...
I have Cisco 7940 (but same applies to Snom's too), and it would be
convenient to have multiple extensions on the same phone registered
against the same asterisk instance. (E.g. one extension which is
associa
Try moving the ATA-186 to a port other then 5060.
John
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Tony Hoyle wrote:
John Fraizer wrote:
Asterisk handles transfer just fine. It's the P-O-S Grandstreams that
don't.
Even this analogue phone that's on my desk handles this... it's not
normally a function of the phone, but of the PBX.
How do companies that use asterisk han
Duane wrote:
John Fraizer wrote:
Spoken like a true n00b13.
If the current SIP bug isn't annoying enough to push people away from
asterisk you just have to chip in your 2 cents worth to push things that
little bit more...
You can *sometimes* get away with not having MX records. Yo
Tony Hoyle wrote:
Stephen R. Besch wrote:
Not as far as I know, at least not exactly the way you have outlined
it. Try this:
1. call comes to you
2. You hold the call and call other person.
3. You say "Someone wants to talk to you, OK, thanks"
3a. Other person then hangs up.
Michael Welter wrote:
Is there a Linux tool that will splice several gsm sound clips together
into one clip?
In my agi script, I would like to use 'get_data' with one clip instead
of multiple 'stream_file' so the user doesn't have to listen to the
entire spiel before responding.
Thanks,
cat c
Duane wrote:
Andrew Thompson wrote:
Does anyone know if any of these options are acceptable substitutes
for an
SRV record, or do I need to put in a ticket to have a SRV record
specifically created for me?
As with email you technically don't need MX records, an A record will
also work fine. I'm p
Eric Wieling wrote:
On Wed, 2004-06-02 at 01:01, Tracy R Reed wrote:
Note however that this WILL NOT work if one of the devices you are calling
is on a Zap channel. I have a PRI and I would love to ring my cell phone
AND my desk phone (SIP) at the same time but if I try only the Zap
interface rings
Kyle Hagan wrote:
Ok I have a testing version available at www.easyhomenetworks.net/astrec
There is a shot docs.txt in the directory you will need to read.
Its very very beta (alpha?).
There are a couple bugs right now. But give me your ideas and
CONSTRUCTIVE critisism please. :)
It will only tra
Andrew Kohlsmith wrote:
Please do not trim out attribution tags.
The double quoted is from Julien Levi <[EMAIL PROTECTED]>
Why not? I replied to Julien Levi's post, so the attribution should be
implied, just as I am replying to your post, and I don't have a "Steven
Critchfield sez" line... I'v
Steve Totaro wrote:
I went with voicepulse after I emailed NuFone sales twice about paying for
some 800 numbers that were never responded to.
After looking at clearpath's page they might get my business since it seems
they offer 800 service.
Who offers the ability to forward your number to a differ
lists wrote:
Humm that SCCP to start sorry
I went up to a signed load on the sccp NP using my CCM's but I can't get
the phone to load a SIP load. I am currently trying 7.1 as per cisco's
paper
http://www.cisco.com/en/US/products/sw/voicesw/ps4967/products_upgrade_guide
s09186a008022a968.html#wp10
lists wrote:
I have a done google seaches on convertion and so far they all failed.
Rich adamson and wheely-bin.co.uk
Here is what I have
Laptop running solarwinds tftp with the following files
OS79XX.txt <- POS30201
SIP.cnf.xml
That should be SIP.cnf
John
_
Andrew Kohlsmith wrote:
In short, you're making this problem worse. Answer the damn support emails
quickly and people won't see the need to post here. I get the "we got your
support question, your ticket # is .." email quickly but then it tends to
languish for a while. I've only had a few sup
Yes, I have. It's pretty simple to do.
John
Rob Franklin wrote:
Hi All
I was wondering if anyone knew whether there is a built in set of
functions for handling "speed dial", basically a set of numbers that can
be entered from that handsets and stored in locations such as 3000, 3001
etc. I can do
-5800 ext 115
On Mon, 24 May 2004, John Fraizer wrote:
Bruce Komito wrote:
> In sip.conf, try setting canreinvite=no for both lines.
>
> Bruce Komito
> High Sierra Networks, Inc.
> www.servers-r-us.com
> (775) 284-5800 ext 115
canreinvite=no will sometimes make a difference but, I
It looks like your cellphone carrier is actually "answering" the call
before they ring your phone. In their switch, they probably have the
equiv of:
exten => your.cellphone.number,1,Answer()
exten => your.cellphone.number,2,Ringing
exten => your.cellphone.number,3,Dial(CELL/${EXTEN},20)
exten =
the inside of the NAT trying to create identical flows. It
could have easilly done some mapping to change the source port on the
WAN side of the connection of one of the flows.
OK. This is the * list and not routing 701 so, I'll stop now. Suffice
it to say that it is a good idea to have u
Barry Fawthrop wrote:
The problem is probably that both of your SIP phones are using the same
port. I played with two 7960's behind a Linksys on Saturday and finally
got them playing right when I changed the following:
In Phone 1's SIP[macaddr].cnf:
voip_control_port: 5061
In Phone 2's SIP[maca
Chad Brown wrote:
I have 2 SIP phones (Cisco 7960 & XTen) behind a NAT provided by a
Linksys firewall that supports UPnP. The Asterisk server has a public
IP. Here are the problems that I am having with this configuration…
1. The 2 SIP phones can call MeetMe and have a conference but cann
Randy Bush wrote:
i try to place a call
exten => _X.,1,Dial(SIP/[EMAIL PROTECTED]:5061,60,Ttr)
where sip.conf has an entry
[foo]
secret=torture
callerid="local ext 103" <1914666>
type=friend
fromuser=asterisk
auth=both
host=dynamic
canreinvite=yes
context
Tor Roberts wrote:
It was my understanding that asterisk would not let you register the
same extension more than once. If that is not the case, I will try to
register the same extension to all 6 lines.
On the 7960's, * does not get upset with having multiple appearances of
the same "line" on a 7
Andres wrote:
[EMAIL PROTECTED] wrote:
Which providers give you a jitter buffer?
In Europe: VoipTalk and Magrathea. In the US: Iconnecthere. I am sure
there are more.
Clearpath gives jitter buffer as well. http://www.clearpath1.com/
John
___
Aste
Tor Roberts wrote:
Hi,
I am setting up a dispatch center where will have 4 call takers, all
with Polycom IP 600 Sip phones. Each phone will be setup with 6
extensions each. When a new call comes in, the first extension on all
the phones will ring. This works fine, the problem is when one of the
Steve Totaro wrote:
2600mhz cpn crunch whistle? bump the oper off the line?
Holy crap Batman! You've got a whistle that does 2.6Ghz? Perhaps you
should look into some RF exposure safety literature.
Hz = cycles of function per second (function being signwave, sawtooth,
squarewave, etc)
KHz =
Steven Critchfield wrote:
You have to do a dial somewhere to get to another place, put the wait(1)
in before that dial. What ever you do with the call, get the wait(1) in
so that the facility tag can be processed into the call before you
forward it, answer it, or whatever you do with it.
Got it.
Steven Critchfield wrote:
On Fri, 2004-05-14 at 09:44, John Fraizer wrote:
We just turned up a PRI with NI2 signaling for callerID & Name. We can
see the name in the CDR records but, it doesn't show up when a PSTN ->
SIP or PSTN -> IAX2 call is received. The phones only
We just turned up a PRI with NI2 signaling for callerID & Name. We can
see the name in the CDR records but, it doesn't show up when a PSTN ->
SIP or PSTN -> IAX2 call is received. The phones only get the number.
Where do I look to try to fix this problem? It seems to be timing related.
John
I did some searching to find out what I need to do to get CID&NAME
inbound on one of our PRI circuits. Right now, we're only getting
number (well, asterisk shows the number in both the name and number
portions of CID).
Is there anything special I need to do with Asterisk to have it accept
the
tmpm wrote:
Of course, and I suggest a firewall as well, but its NOT going to do
anything for a purloined email some infected machine in Bumsquatialand
sending it as you. Theres only so much you can do.
If you run your own mailserver, I suggest the following to keep your
braindead users from b
James Moran wrote:
No I'm not but it's a hospital that nurses are on call and need to have
a way to contact them. On Fri, 2004-05-07 at 09:52, John Fraizer wrote:
James Moran wrote:
We need to have about 30 phones on one floor
And you really think that WiFi phones are suite
James Moran wrote:
We need to have about 30 phones on one floor
And you really think that WiFi phones are suited for this application?
Not an RF engineer, are ya?
John
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Frederic Steinfels wrote:
When talking to me, people are complaining the volume was not high enough.
The phone only allows to change the volume of the speaker/earpiece. Is
there an alternative solution? Is it possible to increase the volume in
asterisk?
Frederic
___
Set up some .procmailrc filters or whatever filters you like and move
anything with "[asterisk-" in the subject into a different spool. Then,
you don't get bothered and can read whenever you want just by looking at
the spool.
You're old-school Randy. I'm surprised you haven't done this alread
Note to self: Don't be a dumbass and open up an unsolicited attachment
and execute the content, especially when it was sent off-topic to a
mailing list.
Any of you who did - please report to building 666 for termination.
You've now proven yourself to be simply sucking up oxygen from others.
F
Nick Knight wrote:
I haven’t used cisco phones as yet – do they work with asterisk, are
they good which models are the best?
I use a 7960 with Asterisk and absolutely love it. It blows the snot
out of the Nortel phone I used to use.
John
___
Asterisk
dkwok wrote:
Let's not cynical about the way I raised the question. I know there is
no matching peer. I do not using windoz myself and I am not accustomed
to pop up gui. But why there is no matching peer, this is not expected.
I presume ztdummy will provide the same timing device as zaptel car
Jeremy McNamara wrote:
It told you everything you need to know: no matching peer.
For the cheap seats: You need a peer.
Jeremy McNamara
I'm convinced that some people can't function without a M$ GUI. It doesn't
matter how verbose and descriptive an error message is, unless it's in a
pop-
vozip wrote:
group=1
signalling=fxo_ks
mailbox=2468
callerid="Phone 1" <2468>
channel=1
ZT_CHANCONFIG failed on channel 1: Invalid argument (22)
Did you forget that FXS interfaces are configured with FXO signalling
and that FXO interfaces use FXS signalling?
ANY IDEAS.!
CHEERS.!
VOZIP
The er
er wrote:
Price, quality, etc?
John Baker
On Wed, 2004-03-17 at 13:36, John Fraizer wrote:
Doug Harris wrote:
> Hi,
>
> Seems like there arn't any alternative to NuFone either ?
>
> Toll free Number + IAX/SIP + 2.9 Cents per minute, no strings attached.
>
> Doug
If
Doug Harris wrote:
> Hi,
>
> Seems like there arn't any alternative to NuFone either ?
>
> Toll free Number + IAX/SIP + 2.9 Cents per minute, no strings attached.
>
> Doug
If you want SIP/IAX termination from someone other than NuFone for the same
price, you can contact me. We can offer that.
Jo
I forward my cisco 7960 to a special extension that "preserves" the original
callerID information. It works just fine.
exten => _**73.,1,Setcallerid(${CALLERIDNAME} <${CALLERIDNUM}>|a)
exten => _**73.,2,Dial(IAX2/provider/${EXTEN:[EMAIL PROTECTED])
exten => _**73.,3,Congestion
exten => _**73.,4,
Cisco 7940/7960 will do what you want. They're not cheap but, then again,
they're not much more than "comparable" phones for my Nortel system cost.
You then set up a "main" extension (regular calls will be sent here), an
"intercom" extension for intercom and a "paging/voicecall" extension
(aut
Doug Meredith wrote:
John Fraizer <[EMAIL PROTECTED]> wrote:
And what I told you works just fine. I'm taking 20+ DIDs from a single IAX
provider with no problems what-so-ever. I'll be happy to consult for you at
my normal hourly rate if you still can't figure it out
You need to make sure the polarity is right. If I'm not mistaken, I think I
saw someone post that the Cisco cubes had the polarity reversed.
John
stan wrote:
Is anyone using a 3com 3CNJPSE to power a 7960G?
I have a couple of 7960Gs and 3CNJPSEs but no combination appears to
work. Both phone
Anton wrote:
> tiburon telecom
> florida
> we current have an actual record of 100% for the last year!
Ah, past year. Is it a young CLEC or did you have an outage at which time,
you restarted the clock? Still, congratulations no matter what.
John
___
Anton wrote:
sure, we are an actual running clec so we require ds3 level interfaces,
99.% uptime, and easy management.
6 nines huh? Just wondering. What is the name of your CLEC and in which
state are you registered? I'd love to look at your PUC and FCC reports to
find 6 nines overall up
Carlos Chavez wrote:
> Is there a way to know which Codec a particular phone is using? I have
> several devices which support different codecs and I would like to find out
> which one was negotiaded with Asterisk. Is there a CLI or Manager command to
> get this information?
>
> --
> Carlos C
The CID NAME is able to be sent to other SIP/IAX devices. That is how those
are used.
Don't get me wrong... I think that Asterisk sends the complete CLID (name +
number) on PRI and T1 interfaces to the telco. It is just that the telco
ignores the "name" portion and looks it up in its own data
(1) I don't think you can set CID on a POTS line (or BRI ISDN) at all so,
you'll need PRI or DS1.
(2) You can't set the name on what is sent to the telco. Only the number.
John
Carey Jung wrote:
Hi,
I've got a small asterisk server setup, dialing out on an X100 card and
would like to implemen
admin wrote:
From a windows box run tracert and you will get a better idea where the
problem lies.
Ew! Why on earth would you want to use winblows for traceroute? From a
*nix box, run traceroute . Now, that's better.
John
___
Asterisk-Users ma
Jeremy McNamara wrote:
started.Didja hear about RedHat buying out Digium ? Or was it
Microsoft?
Jeremy McNamara
You've got it all wrong, man! Microsoft bought SCO who bought RedHat who
bought Digium. ;)
John
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Asterisk-Users mailing list
[E
[no name provided] wrote:
You're right. It's symmetric so it only takes 83Kbits/sec for u-law.
IPTraf is confusing me :-)
IPTraf is a neat tool but, information it gives should be taken with a grain
of salt. If you're looking at "general" statistics, it is going to show you
the combined IN+OU
It is included with Asterisk.
John
hank wrote:
- Original Message -
From: hank
To: [EMAIL PROTECTED]
Sent: Wednesday, March 10, 2004 1:07 PM
Subject: where can I get Commedian mail at?
hello where can I get Commedian
Mail at?
thanks
hank
__
Steve Kennedy wrote:
It't not quite that simple, DSL in the UK is PPPoATM, so you need to
take into account who IP is encoded at the ATM layer etc. If you're
using a reasonable bit rate codec, you can't really expect to get much
more than 1 voice channel out of an ADSL service (assume something lik
Steve Creel wrote:
Can you test this with an extension that goes into VoiceMailMain(). My
7960 and 7960G phones both get the first couple letters of "Commedian
Mail" cut off (usually "...median Mail").
Just trying to quantify the delay we're talking about...
Steve
exten => 8500,1,Answer
exten =
Bisker, Scott (7805) wrote:
> What versions of Zaptel, Asterisk, and libpri?
>
>
I downloaded them all at the same time from CVS. I really couldn't tell you
though off the top of my head.
John
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You need to decide if you're going to measure both sides of the call or not.
ITU standard is 64Kbits/s. That is correct. It is a standard DS0. But,
guess what. That DS0 goes both directions so, "measured bandwidth per call"
is 128Kbits/s using your logic.
Only "consumer" grade DSL/Cable ba
For what it's worth, I don't have any delay between answer and audio with my
asterisk server and 7960G either originating or answering. It doesn't
matter if it's a call to/from another SIP/IAX device or to/from PSTN. It's
pretty much instant (not detectable by humans at least). So, there may
And what I told you works just fine. I'm taking 20+ DIDs from a single IAX
provider with no problems what-so-ever. I'll be happy to consult for you at
my normal hourly rate if you still can't figure it out.
John
Barton Hodges wrote:
Both register commands register with the iaxtel provider.
You do this with contexts attached to the [provider] section in the iax.conf
file and you provide coresponding contexts and extensions in your
extensions.conf file.
John
Barton Hodges wrote:
With entries in sip.conf, I can route incoming SIP calls with an
extension specified in the register com
Simon Chappell wrote:
Hi al
I have 3 GS 101's plugged into asterisk.
They work great and teh quality of sound I can not fault. Most people I am
speaking to now ask if I have a new phone because the quality is so much
better.
Don't ever use a Cisco phone if you're happy with your GS phones right no
g729 channels.
What codecs are you using ? Is there a conference config in the 7960 that
I'm missing ?
I can make inbound and outbound calls just fine on this phone using g279 to
other sip phones and the pstn all day long.
Thanks,
Chris
- Original Message -
From: "John Fraizer&qu
Chris Clifton wrote:
Anyone been able to get the conference feature on the 7960's to work without
using meetme ?
I get - warning, chan_sip.c:2103 process_sdp: No compatible codecs!
Thanks,
Chris
* is telling you exactly what the problem is. You're running incompatible
codecs on the sessions yo
Tilghman Lesher wrote:
>
> On 2004 Mar 05, at 05:18, atif wrote:
>
>> how to disable this DEBUG information...
>> I am getting this on Asterisk CLI
>>
>> ---
>> Mar 5 16:18:17 DEBUG[426001]: chan_zap.c:3397 __zt_exception:
>> Exce
Darren Wiebe wrote:
The complaint I'm getting from a few people is that when they hang up their phones, they still
cannot get dialtone for a while. Two people said last night that even
20 seconds after they hung up their phones, when they picked up again,
they still did not have a dial tone. I
David Uzzell wrote:
So for us Dummies out here :) who just know it works.
This would mean that if you had a 512/256 aDSL and a 256 ISDN connection
you would be able to have more channels over the ISDN?
David
It all depends on what excapsulation your aDSL uses. It boils down to
encapsulation ov
In your DNS zone file for the domain you are using, put:
_sip._udp SRV 0 0 5060sipproxy.yourdomain.com.
sipproxy300 IN A 1.2.3.4
John
Chris Lee wrote:
When handed a URL type address for telephony, is there a DNS "TX" record
(like MX but for tele
David Liu wrote:
Hi John,
Is this what you are getting?
Mar 2 10:28:47 WARNING[245776]: chan_sip.c:4978 handle_response: Got 200 OK
on REGISTER that isn't a register
This is what I got too and I have been puzzled. I have tried all sorts of
thing by varying the register message with FWD and it
On that note, I had been getting warnings about receiving a "200
Registration OK from x.x.x.x - Not a register) or something along those lines.
I tracked it down (finally) to FWD. When I removed the Register statements
for two FWD accounts, those messages went away. I tried adding only one of
Glenn Dalgliesh wrote:
I seem to be having trouble with cvs login. anyone having similar problems
It just hangs after entering the password
Make sure you actually have connectivity to the CVS server (ping/traceroute).
John
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Asterisk-Users mailing
This will definitely work for a wakeup call, processed from a call file:
;file sample.call
Channel: SIP/1234
MaxRetries: 2
RetryTime: 60
WaitTime: 30
Context: wakeup
Extension: s
Priority: 1
In extensions.conf:
[wakeup]
exten => s,1,Wait(2)
exten => s,2,Playback(tt-monkeys)
exten => s,3,Hangup
It
Carl wrote:
I'll give them a whirl. Cheers C.
If you email me a username/PW combo, I'll get you an account set up and
email you the particulars for this side (or telephone you if you include a
number) as soon as I get home from dinner.
John
___
Asteri
Carl wrote:
> I'll give them a whirl. Cheers C.
Carl, are you not getting my emails?
John
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That's all you need. At least, that's kinda how I have mine set up and it
works fine to dial-by-email.
WipeOut wrote:
If I want users to be able to call each other (or others to be able to
call users on our Asterisk system) using their email address
([EMAIL PROTECTED]) what would have to be do
carl wrote:
Anyone got an example of sip and extensions confs for Iconnect outgoing calls behind NAT.
Here you go:
[Scene starts out with you on the phone with IConnect technical support.]
You: "I know that Asterisk isn't one of your supported platforms. I'm not
asking you to support my 'devi
IE; If I tell my asterisk server to set
my callerID to "test" and call someplace, What I get on the
CLID display of the phone I dial is "John Fraizer" and my home number.
Since Powell has stated that we must provide E911 services, I am wondering
what precisely is g
dkwok wrote:
I cannot get MWI working either with GS101 firmwire 1.0.4.39
My sip.conf has the mailbox number specified. voicemail.conf has mailbox
set up. I have collecting mail fine.
If you're running any other voicemail contexts other than default (in your
voicemail.conf), you need to specif
I conference calls all the time. Asterisk is telling you what the problem
is. You are running a codec that it doesn't like for whatever reason.
I use ULAW and ALAW with absolutely no problem.
Chris Clifton wrote:
Is app_meetme the only way to conference calls on a 7960 with * ? It looks
as if
I would put it in a totally different light. IE; depending on who they use
as an IAX/SIP carrier, they may have potentially unlimited outbound and
inbound lines with the limit only imposed by the total number of indications
on the phones in the office and even then, new inbound calls can still
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