On Sun, 2004-12-19 at 02:11, David Uzzell wrote:
Then the other thing if mem serves me you are running 2.6 kernel so why
not run ztdummy? With the 2.6 kernel this does not require any
specialist Hardware or anything!
Sorry, but maybe you should have read his posts more thoroughly. ztdummy
is
Bruno Hertz wrote:
On Sun, 2004-12-19 at 13:11 +1100, David Uzzell wrote:
I have * running on Mandrake 10.1 and I to had similar problems in the
begging but as soon as I had ztdummy configured correctly everything
seemed to just fall into place and work with IAX and *, not that I have
got a
Yes, of course you can do that. I have this very setup working for the
office, with * aggregating voip and isdn incoming calls and forwarding
them to my laptop wherever I am.
just follow the instructions on the FWD website, and run iax2 debug from
the console to see what's happening in
On Sun, 19 Dec 2004, Eric Bishop wrote:
Apart from the the coolness factor can anyone explain to me in what
situation one would use TDMoE rather than IAX for communication
betwwen 2 Asterisk servers?
There are two advantages with TDMoE:
* low latency (prevents far end echo from going from
Martin List-Petersen wrote:
On Sun, 2004-12-19 at 02:11, David Uzzell wrote:
Then the other thing if mem serves me you are running 2.6 kernel so why
not run ztdummy? With the 2.6 kernel this does not require any
specialist Hardware or anything!
Sorry, but maybe you should have read his posts
Hi all.
Is there a way to use asterisk for call screening?
Meaning, a call comes in, asterisk answers with voicemail after I don't
pickup, and the voicemail prompt + the caller's message a played via the
sound card on asterisk. If I wan't to pick up, I do so by picking up the
phone and dialing
Yes
U can do it with asterisk and by dialing *98 on your Ip Phone you can listen
to your message
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Shoval Tomer
Sent: Sunday, December 19, 2004 1:40 PM
To: Asterisk Users Mailing List - Non-Commercial
Sorry, I don't follow.
Dialing *98 will achieve what?
Up until the time I decide to take the call, I want to be able to hear
the person leaving a message interactively, so when I decide to pick up
the call he's still there.
Like a regular answering machine
-Original Message-
From:
Hi,
Currently I am using a ISDN BRI PCI FRITZ card (works), would I get
any benefits switching to a HFC card? Or it would be a better choice to
switch to a ISDN with a DSP processor?
Currently I have echo on my CAPI channel when calling analog lines,
if call a cell phone, ISDN
Sorry
I mean the voice mail
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Shoval Tomer
Sent: Sunday, December 19, 2004 2:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] call screening
Sorry, I don't
It seems that all my CDR is dumping into the Master.csv file. There is a way
to create per user/extension CDR but I have looked endlessly in the Wiki,
docs, README.CDR, mailing list archives etc.. I can't seem to find a way to
do this..
Any help would be appreciated.
Thanks!
--
Start Your Own
HI;
I have an Asterisk with 10 "SIP" ip-phones, our pbx
features are now: Voicemail and Call Transfer.
How can I serve both "Call Waiting / 3 way calling"
for our SIP Phones.?/
Appreciate Any Help
Mohammad
___
Asterisk-Users mailing list
I have * running on Mandrake 10.1 and I to had similar problems in the
begging but as soon as I had ztdummy configured correctly everything
seemed to just fall into place and work with IAX and *, not that I have
got a perfect dialplan as that confuse's me but hey thats another subject.
Hi
I am struggling with hardware choices to get started with. My options are
narrowed down to SIP phones - Polycom IP500, IP600 and Cisco 7940G.
of importance is:
- functionality / integration with asterisk
- headset functionality and use
- voice quality
- build quality
Is there much of a
http://www.voip-info.org/wiki-RTP+Silence+Suppression
http://lists.digium.com/pipermail/asterisk-users/2003-August/018670.html
So I am confused. The first says that VAD is supported in RTP. Ok, I know
that. The
second is kinda ambiguous and seems to imply that *
doesnt support
how do youintegrateGnugk and Asterisk billing?
Are you using Asterisk's H323 channel?Voip Business [EMAIL PROTECTED] wrote:
I integrate Gnugk and a gnugk billing system working like a charm.regardsHAOn Sat, 18 Dec 2004 01:48:56 -0800, Inam <[EMAIL PROTECTED]>wrote: HI Alll this is my first post
On Sun, Dec 19, 2004 at 12:21:28AM -0600, Matthew Boehm wrote:
I'm having a similar problem. Do you have operator=yes in your
voicemail.conf under [general]?
Argh, thats it, solved!
Thanks a lot :)
...cut
--
Tho/\/\as
___
Asterisk-Users mailing
OK. I now have call recording working for both incoming and outgoing
calls.
Now I want to make those wavs into mp3. I could launch a script from
cron that checks for new wavs and converts them. But that wouldn't be
so elegant.
Launching it from * on hangup would be nicer. How is it done?
He's trying to use sip, not iax. It would appear he's got both a fwd
registration issue and an incoming fwd context issue. They don't appear
to be in sync (probably an understanding of context issue actually).
Yes, of course you can do that. I have this very setup working for the
office,
Hi
I've bought the Wildcard TE110 some days ago but I'm unable to get it to work
with Siemens HiCom 300.
I've tried this so far:
1. I've used standard cat5 cable cut off on one edge and twisted wires 1 to 4
and 2 to 3 which is according to cisco a short circuit for E1 lines. Wildcard
takes a
Gonzalo,
Have you tried IAX, I see yo are behind NAT, and my experiences with IAX
behind NAT are much less painful :-)
I've FWD via IAX, receiveing calls (in fact, last time was a nearby
person in Portugal :-) that tested it).
One last thing, you mention dialup client, I guess she is not in
Hello--
I've done some coding for call screening in Asterisk. It's not in
Asterisk yet, mainly because we're waiting for
prompts from Allyson so it sounds like the rest of the system. But
patches, prototype sound files, etc, are all
filed at:
On Sat, Dec 18, 2004 at 06:28:54PM +, Antony Stone wrote:
On Saturday 18 December 2004 18:07, Dorn Hetzel wrote:
I wouldn't say I hate SIP, it sucks less than H.323 and
so on by a large margin. But, having said that, if you
can use IAX, it sucks even much than SIP does :)
Um, are
From the trace it appears that you are not getting any Layer 2
communication. All the broadcast messages like SETUP and the TEI
assignments are being sent (and because your phone rings, it is hearing
what Asterisk is saying to it. Your ISDN phone does not appear to be
responding. This looks
On Fri, 2004-12-17 at 16:47 -0800, Shahed wrote:
Hello all,
(Not sure if this is more appropriate for user or dev list)
Does asterisk have any sort of standards based api that can enable
an application to do call control on the switch ?
For example, if I am developing a call center
Steven Wang wrote:
Hello
I try to set up voicemails for extension. When VoicemailMain gets called, it
prompts for mailbox and password. It seems not able to read from the phone.
So the authentication always fails.
I desparately need help to understand what is wrong. Here is a part of my
On Sunday 19 December 2004 06:31, Wilson Pickett wrote:
Is it possible to send the incoming PSTN caller ID to a Grandstream
Budge Tone-100 SIP phone? I've configured the extensions.conf file
and the log is
As Eric notes, the BT100 phones won't show letters. If a call comes
in without
It BT100. it works.
thanks!
steven
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Russ
Beaupre, P.E.
Sent: Sunday, December 19, 2004 8:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] VoicemailMain can't read
On Sun, 2004-12-19 at 00:40 -0800, Chris Miller wrote:
From what I have read the issue with choppy sound under the demo voice
seems to be due to a timing issue
Taking the risk of appearing notorious, I again emphasize that I don't
believe that.
I have asterisk right now with ztdummy running
[EMAIL PROTECTED] wrote:
hi
any chance of making asterisk support these?
http://voipstore.atacomm.com/shops/ViewItem.aspx/27934028032-3835624908
8.htm
According to the manufacturer, they already do:
http://www.ipvolution.com/
Cheers,
Jim.
--
No virus found in this outgoing message.
Hello,
I would like to parse inbound Asterisk IAX2 7-digit numbers in the form of
123-4567 and strip out the first four digits, and then dial whatever number
digits remain. If I only have three digits (000-999) and have a mix of
channels (ZAP, SIP, IAX2) could someone please point out how I can
mohammad wrote:
I have an Asterisk with 10 SIP ip-phones, our pbx features are now: Voicemail
and Call Transfer.
How can I serve both Call Waiting / 3 way calling for our SIP Phones.?/
This is what I call one of the dirty little secrets of SIP. On SIP
phones (and H323) all the call control is
Matthew Boehm wrote:
Hey gang,
Getting ready to run some test bills for customers. Most SIP phones have
both an extension and a DID. If a person calls a DID asterisk redirects the
call to the right extension:
exten = 8005551212,1,Goto(companyA-internal,3022,1)
The problem is, that if someone
Steven Wang wrote:
Hello
I try to set up voicemails for extension. When VoicemailMain gets called, it
prompts for mailbox and password. It seems not able to read from the phone.
So the authentication always fails.
This is almost ALWAYS a DTMF problem. Usually a DTMF mode mismatch
between the
Jim Van Meggelen wrote:
According to the manufacturer, they already do:
http://www.ipvolution.com/
Wow... if that board actually ships as promised, with Asterisk support,
that will be amazing. Up to 8 T1/E1 in a singe PCI slot, with onboard
codecs and echo cancellation... and a price that is
On 19/12/2004 16:40 Chris Miller said the following:
seems to be due to a timing issue, one that can't be solved under
FreeBSD with the zaprtp (linux) stuff, and I haven't seen anything as
the ztdummy pseudo timer works well under freebsd 4.x and 5.x. i used it
for a bit before i got my digium
On 19/12/2004 20:38 Rich Adamson said the following:
I'm 95% sure iax is not dependent on the ztdummy type timers.
trunked iax requires a timer, either ztdummy or a digium card.
--
Regards, /\_/\ All dogs go to heaven.
[EMAIL PROTECTED](0 0)
I did have fromuser set in my sip.conf so I went in and commented the line
out (thanks for the help on that). This is what I have in my
extensions.conf file:
exten = s,1,SetCallerID(${CALLERID}) ; Set the caller ID
exten = s,2,Wait(2)
exten = s,3,Dial(SIP/1234,20,tr) ; Dial our office SIP phone
On Sun, 19 Dec 2004, Jens Kübler wrote:
I've bought the Wildcard TE110 some days ago but I'm unable to get it to work
with Siemens HiCom 300.
I've tried this so far:
1. I've used standard cat5 cable cut off on one edge and twisted wires 1 to 4
and 2 to 3 which is according to cisco a
I forgot to ask, since the BT100 can't take characters (only numbers), I
would have assumed that there was a function to extract a number from an
incoming PSTN CID, is that possible?
Thanks again,
David
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Hi.
I gave up with the IBM NetFinity, so I'm going to buy new hardware. I'm
going to install:
1-)One X100P (1 FXO module)
2-)One TDM03B (3 FXO modules)
I'll have the 4 FXO channels busy almost all the time, and I would like
quality to be as good as possible without going to the high-level
I forgot to ask, since the BT100 can't take characters (only numbers), I
would have assumed that there was a function to extract a number from an
incoming PSTN CID, is that possible?
Try this
exten = s,5,SetCIDNum(1234)
and see if the phone displays it
This is almost ALWAYS a DTMF problem. Usually a DTMF mode mismatch
between the phone and Asterisk. For most phones you want to use RFC2833
for both the phone and for the entry for that phone in sip.conf.
Yep, and the BT will only work right with certain codecs. I think it's
iLBC that
I've read quite a bit in the older mailing list posts and the wiki but
I'm missing some simple point.
1) What is required to send an SMS to a mobile outside the office given:
Channel: ZAP/1
send it to $SMS_RECIPIENT (which includes the final extra digit)
via
$SMS_CENTER=the national message
I'm interested in this, too. I find that when I use Xten or SjPhone
software locally the quality is quite good, but when I use it remotely
across the internet, I get quite a crackly response.
*however*, if I use some SIP hardware, such as a Grandstream 236 or an IP
phone (still use alaw just
On Sun, 2004-12-19 at 20:10 +0100, Rodolfo Grave wrote:
Hi.
I gave up with the IBM NetFinity, so I'm going to buy new hardware. I'm
going to install:
1-)One X100P (1 FXO module)
2-)One TDM03B (3 FXO modules)
I'll have the 4 FXO channels busy almost all the time, and I would like
The SMS in asterisk is not SMS like you're thinking... Its not for sending
to mobile phones and not something usable in the US.
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Wilson Pickett
Sent: Sunday, December 19, 2004 1:42 PM
HI,
basic question. I've got a TE110P card and I'm trying to set it up with
ztcfg with polish zone.
ioctl(ZT_LOADZONE) failed: Invalid argument
Notice: Configuration file is /etc/zaptel.conf
line 206: Unable to register tone zone 'pl'
I've got loadzone and defaultzone set to pl, and there is a
Wilson Pickett wrote:
This is almost ALWAYS a DTMF problem. Usually a DTMF mode mismatch
between the phone and Asterisk. For most phones you want to use RFC2833
for both the phone and for the entry for that phone in sip.conf.
Yep, and the BT will only work right with certain codecs. I think
Am Sonntag, 19. Dezember 2004 21:40 schrieb Marcin Mazurek:
HI,
basic question. I've got a TE110P card and I'm trying to set it up with
ztcfg with polish zone.
ioctl(ZT_LOADZONE) failed: Invalid argument
Notice: Configuration file is /etc/zaptel.conf
line 206: Unable to register tone zone
I am running * in a development environment, adding functionality as I
go.
The * box has a X100P card in it which ztcfg enabled as channel 1 with
fxsks signalling (fxsks=1).
Everything worked fine and I was able to make inbound and outbound calls
to/from the PSTN, the only issue being that some
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Samudra E. Haque
Sent: Sunday, December 19, 2004 12:58 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] dialplan selection
Hello,
I would like to parse inbound Asterisk IAX2 7-digit numbers
[globals]
X1000=SIP/1000
X1001=ZAP/1001
X1002=IAX2/1002
X1003=SIP/1003
[outbound]
exten = _123,1,Dial(${X${EXTEN:4}},10)
Oops, that line should read:
exten = _123,1,Dial(${X${EXTEN:3}},10)
___
Asterisk-Users mailing list
[EMAIL
Hi All,
I dont know too much about the technical specs on DPNSS, but can
support for it be developed in software, like libpri ?
I guess what I am asking is, if DPNSS is just another
signalling protocol, I suppose it can be built using software,
as a layer over zaptel using a digium digital E1
On Sun, Dec 19, 2004 at 06:56:19PM +1100, Eric Bishop wrote:
Hi all,
Information on this topic seems a little scarce, so I thought I'd try
the list
Apart from the the coolness factor can anyone explain to me in what
situation one would use TDMoE rather than IAX for communication
I was wondering how to make asterisk transfer a sip call automatically as sip endpoint. For example, SIP call comes to asterisk from a SIPproxy/Endpoint that offer Call Transfer feature, I want Asterisk send SIP REFER (transfer) tothat SIP proxy/Endpointso thatCaller transfersthatcall to another
On Sunday 19 December 2004 20:18, Brian West wrote:
The SMS in asterisk is not SMS like you're thinking... Its not for sending
to mobile phones and not something usable in the US.
Um, sorry, but if SMS is not for sending to mobile phones, then what is it for
(if it matters, I'm not in the US)
I was dloading cvs over the top of a stable branch... (Matthew told me
that
was a no-no...)
No. That is not what I said. I said that when you do cvs update inside
a previously CVS'd download of STABLE you are NOT getting the most recent
version of asterisk.
There are two ways to download
This is something we would deffinatly be interested in. Our only beef with
the digium cards is that you can only get 1 in a machine, unless you want to
start messing with all that IRQ problems people complain about.
If we want to handle 12 PRI's worth of calls, we will have to buy 3 machines
between asterisk boxes and fixed line SMS I believe but never was 100%
sure on this either.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
Me wrote:
It seems that all my CDR is dumping into the Master.csv file. There is a
way to create per user/extension CDR but I have looked endlessly in the
Wiki, docs, README.CDR, mailing list archives etc.. I can't seem to find
a way to do this..
I'm probably not the right person to answer
Hello,
I've spent the last few days installing asterisk, and the support and
documentation available here and on the wiki has been exceptional. I
have now configured an E100P, with about 20 internal SIP extensions
(snom 190), and a handful of international SIP extensions. Everything
is
If each account has an account code it should spawn off a CSV CDR or
you can just do a mass select from SQL by account code.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or
On 2004.12.19 10:17 Eric Wieling aka ManxPower wrote:
Personally I don't really approve of a company just taking Digium's
design and cloning it.
Huh? To what hardware are you referring? Certainly you wouldn't be
indicating that the GPL only permits one licensee.
On Sun, Dec 19, 2004 at 12:09:48PM +0200, Shoval Tomer spake thusly:
Is there a way to use asterisk for call screening?
Meaning, a call comes in, asterisk answers with voicemail after I don't
pickup, and the voicemail prompt + the caller's message a played via the
sound card on asterisk. If
On Sunday 19 December 2004 21:35, Antony Stone wrote:
On Sunday 19 December 2004 20:18, Brian West wrote:
The SMS in asterisk is not SMS like you're thinking... Its not for
sending to mobile phones and not something usable in the US.
Um, sorry, but if SMS is not for sending to mobile
I'm pretty sure if you assign account codes to your SIP and/or IAX clients
in their respective .conf files then cdr files will automatically be
generated for each individual account code in addition to the master.
No idea about how it works with real time.
hth.
Aaron
- Original Message
Hi!
Everything is fine up to 190 channels, but the 191st call fails every
time with errors like:
Dec 14 15:44:00 WARNING[1215]: Unable to start PBX on Zap/201-1
Dec 14 15:44:00 WARNING[1215]: Failed to create update thread!
Dec 14 15:44:00 WARNING[1215]: Unable to start PBX on channel 0/9,
According to this it exists:
http://www.voip-info.org/wiki-Asterisk+cmd+Dial
However I'm testing it for the last 8 hours with no success.
Recompiling after reading this:
http://bugs.digium.com/bug_view_page.php?bug_id=0002905
will post back
On Sun, 19 Dec 2004 14:46:01 -0800, Tracy R Reed
Um, sorry, but if SMS is not for sending to mobile phones, then what is it
for (if it matters, I'm not in the US) ?
i am in germany and use app_sms to send sms messgaes to mobile phones.
app_sms does not talk directly to mobile phones but to the sms message
center that in turn sends the sms
Should be an account code field in the DB table that can be used in
queries to just pull 1 accounts records
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
On Sun, 19 Dec 2004 12:52:40 +, w fm3 wrote:
Hi
I am struggling with hardware choices to get started with. My options are
narrowed down to SIP phones - Polycom IP500, IP600 and Cisco 7940G.
of importance is:
- functionality / integration with asterisk
- headset functionality and use
-
On Sun, 2004-12-19 at 14:57 -0800, Lee Howard wrote:
On 2004.12.19 10:17 Eric Wieling aka ManxPower wrote:
Personally I don't really approve of a company just taking Digium's
design and cloning it.
Huh? To what hardware are you referring? Certainly you wouldn't be
indicating that
Shahed wrote:
Hi All,
I dont know too much about the technical specs on DPNSS, but can
support for it be developed in software, like libpri ?
I guess what I am asking is, if DPNSS is just another
signalling protocol, I suppose it can be built using software,
as a layer over zaptel using a digium
Right now I'm stuck at this point:
[default]
exten = 1002,Macro(stdcs,1002,SIP/1002)
[macro-stdcs]
;; arg1 exten
;; arg2 device
exten = s,1,Wait(0.2)
exten = s,2,Playback(vm-rec-name)
exten = s,3,SetVar(SCREEN_FILE=/tmp/${CALLERIDNUM}-${EPOCH})
exten = s,4,Record(${SCREEN_FILE}:gsm|2|4)
exten =
Could you (or anyone else who got SMS working) please send some config files?
--
Socrates.
On Sun, 19 Dec 2004 23:39:45 +, Stefan Reuter [EMAIL PROTECTED] wrote:
Um, sorry, but if SMS is not for sending to mobile phones, then what is it
for (if it matters, I'm not in the US) ?
i am
Hi Matt,
I have a coupple of question yet,
First a couple of keys, so we know we're talking about the same things.
Your setup (as I understand it) is:
IAXy - Asterisk A --IAX-- Asterisk B
Ok, as I see my current setup is:
LANInternetLAN
(IAXy A)
Do the paths to each of the include files exist?
If not, you will need to edit the Makefile in that directory to point
to the right include directories.
- James
On 18/12/2004, at 1:14 PM, David Adade wrote:
Hi,
Can anyone help? I get the following error when trying to complie the
h323
Not sure if anyone on here has heard of this before, kind of
OT but still very interesting to me and Im sure several people here.
Any thoughts?
http://telephonyonline.com/ar/telecom_callwave_launches_voip/index.htm
Cheers,
Dean
Steve Underwood wrote:
It might be hard to get anyone outside the UK to
take any interest in it.
You are right about that.
However, if there is anyone on this list who
has any thoughts on how this can be done,
could you please contact me OFF list
to exchange ideas ?
Thanks
Shahed
I feel this is a slap in the face for those of us that have been here and I
don't feel I should HAVE to pay to be certified... I think me and MANY
others are about to walk out of the project over this. I have already
spoken with many people that are close to the project. You're hurting US
and
I feel this is a slap in the face for those of us that have been here and I
don't feel I should HAVE to pay to be certified... I think me and MANY
others are about to walk out of the project over this. I have already
spoken with many people that are close to the project. You're hurting US
and
On Mon, 20 Dec 2004, James wrote:
Do the paths to each of the include files exist?
If not, you will need to edit the Makefile in that directory to point
to the right include directories.
- James
On 18/12/2004, at 1:14 PM, David Adade wrote:
Hi,
Can anyone help? I get the
Brian West wrote:
I feel this is a slap in the face for those of us that have been here and I
don't feel I should HAVE to pay to be certified... I think me and MANY
others are about to walk out of the project over this. I have already
spoken with many people that are close to the project.
Hello
I have one phone (phone1) in one network, the other (phone2) in public
network. both can call the other side; phone1 can be heard by phone2, phone2
can't be heard. I don't have NAT set on both end, but I use rtpproxy on SER.
Is NAT still necessary to be set on both phones?
Thank you!
OK I now know what was/is worng, my SIP is wrong it doesn't give 2 way
audio, so first I'm going to fix this and then we will see.
On Sun, 19 Dec 2004 19:26:59 -0500, C F [EMAIL PROTECTED] wrote:
Right now I'm stuck at this point:
[default]
exten = 1002,Macro(stdcs,1002,SIP/1002)
http://bugs.digium.com/bug_view_page.php?bug_id=0002905
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of C F
Sent: Sunday, December 19, 2004 8:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
I downloaded the latest CVS today, and since then I have only one way
audio on my sip channels the callee can't hear the caller. whats
wrong?
I did the follwoing:
cvs checkout asterisk
make clean
make
make install
running FC3 linux 2.6 64bit
___
What is the most efficient way to allow inbound callers to
dial internal users yet restrict them from outbound PSTN calls? Today I have a
basic greeting that after a welcome message allows inbound callers the ability
to dial any of my users. However, it seems that since I transfer the
Chad Brown wrote:
What is the most efficient way to allow inbound callers to dial internal
users yet restrict them from outbound PSTN calls? Today I have a basic
greeting that after a welcome message allows inbound callers the ability
to dial any of my users. However, it seems that since I
dear all,
does anyone have a clue why in the event messages it
show that Unregistered '1000' (AUTHENTICATED) if i'm
using iaxfriends ?
if using iax.conf text file configuration ... the
status showed Registered '1000' (AUTHENTICATED)
i'm using asterisk 1.0.3 and iaxcomm-linux (pre CVS 28
Feb
Hello
I have one phone (phone1) in one network, the other (phone2) in public
network. both can call the other side; phone1 can be heard by phone2,
phone2
can't be heard. I don't have NAT set on both end, but I use rtpproxy on
SER.
Is NAT still necessary to be set on both phones?
Thank
It seems that would be pretty easy to setup with
Asterisk. I wonder what amounts of usage are included at that
price?
Not sure if anyone on here has
heard of this before, kind of OT but still very interesting to me and Im sure
several people here.
Any
thoughts?
7. How Much Does It Cost?
Sign up today for a RISK-FREE 30-day trial of CallWave! Keep it, and
you'll pay a special, introductory rate of only $3.95 per month.
Cancel any time before your trial ends and you pay nothing.
Hmm seems they aren't exactly sure what to expect. TOS didn't seem to
have
Steven Critchfield wrote:
I would suggest something in a serverworks board. So far we have had a
PIII 850 on a serverworks chipset and SCSI drive running for a long
time. Our main PSTN gateway has a 418 day uptime and asterisk has been
running non-stop for nearly 20 weeks. We take nearly 500
Hi, Julio,
thanks for the tip, IAX and the incoming calls confi did the trick! FWD is up and running!
THANKS! and happy holidays!
Do you Yahoo!?
Yahoo! Mail - now with 250MB free storage. Learn more.___
Asterisk-Users mailing list
Hello
I am trying to compile asterisk-oh323-0.7.0 with pwlib-Janus_patch4
and openh323-Janus_patch4 downloaded from inaccessnetworks so I did
this:
tar -zxvf openh323-Janus_patch4-src-tar.gz
cd openh323
patch -p1 /root/asterisk-oh323-0.7.0/openh323_1.13.5-make.patch
./configure
make opt
cd
Hi all
I have MFCR2 successfully
installed but seems to get warnings a s seen below when I start asterisk. Am
running on Redhat 9.
Asterisk Ready.*CLI Dec 20
08:40:38 WARNING[1175077440]: chan_unicall.c:634 unicall_error: UniCall: mfcr2
far_unblocking_expiredDec 20 08:40:38
I am having problems getting incoming caller id to work on a Telstra
Onramp 10.
I have changed /DEFAULT_CIDRINGS 2/
Is there something i'm missing ?
My Cisco 7960 just shows asterisk
Thanks,
Nathan
[zapata.conf]
context=incoming
usecallingpres=yes
relaxdtmf=no
rxgain=0.0
txgain=0.0
busydetect=no
Nathan Alberti wrote:
I am having problems getting incoming caller id to work on a Telstra
Onramp 10.
I have changed /DEFAULT_CIDRINGS 2/
Is there something i'm missing ?
My Cisco 7960 just shows asterisk
Thanks,
Nathan
SNIP
linux*CLI show channel Zap/2-1
-- General --
Name: Zap/2-1
1 - 100 of 113 matches
Mail list logo
