http://bugs.digium.com/bug_view_page.php?bug_id=0001589
Has anyone else heard an audible blip, break or
garble between answer and the native bridge attempt using sip?
If I change the usleep(50); to usleep(5000); in
rtp.c the proble totally goes away... even the note above it says it ne
Craig wrote:
Greetings all,
I have searched all over and have found bits and pieces on low bit rate
codecs, however I have found it very difficult to compare apples with
apples.
The conclusions I have come to are as follows, I would appreciate if
anyone has some feedback, or point me to where I m
Try enabling busy detect and set it to a value between 4 and 6. If you set
it too low you might start getting random call drops. I think this problem
is due to some providers allowing only the called party to hang up.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
They also structure the fees in such a way that it is impossible to
actually use the full value on the card. IIRC, they term this
"breakage" and it means you end up with at least some amount of unusable
value left on the card at end of use.
THX/BDH
On Sat, 2004-05-08 at
60 second increments, per call connect charges that range in the 39-99 cent
range.
bkw
- Original Message -
From: "Jeremy McNamara" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Saturday, May 08, 2004 4:57 PM
Subject: Re: [Asterisk-Users] 1800 Provider
> Jim Onnet wrote:
> > How d
Asterisk CVS instructions:
http://www.asterisk.org/index.php?menu=download Download CVS HEAD or
CVS STABLE for Asterisk. Zaptel and LIBPRI do not have a CVS stable
branch, only a CVS head branch. I recommend the stable branch.
The code you are looking for is only in the HEAD branch.
On Sat, 20
Jim Onnet wrote:
> How do the phone cards company with
2cents/minute rate do it by giving out 1800 access number?
By lying to their customers about the actual rate they are being charged.
Jeremy McNamara
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CAn you help me on
how to take that code out?
I tried with CVS export but it says there is no directory with that name..
Tx.
HQ.
- Original Message -
From: "Eric Wieling" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Saturday, May 08, 2004 6:40 PM
Subject: Re: [Asterisk-Users] DIAL w
was posted on a day or two ago
Thorsten Gehrig wrote:
hi
anybody running with german SIPGATE?
my configuration don't works :-(
regards
[EMAIL PROTECTED]
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Greetings all,
I have searched all over and have found bits and pieces on low bit rate
codecs, however I have found it very difficult to compare apples with
apples.
The conclusions I have come to are as follows, I would appreciate if
anyone has some feedback, or point me to where I might find th
we do
in sip.conf
register => userid:[EMAIL PROTECTED]/userid
[sipgate1]
type=friend
username=userid
secret=password
host=sipgate.de
fromuser=userid
fromdomain=sipgate.net
nat=no
;dtmfband=inband
context=incoming-ip
canreinvite=yes
then in extensions.conf create the following exten in conte
Is it possible to strip some numbers from the *end* of a number?
I know that ${EXTEN:1} will remove 1 position from the beggining... but
how to remove N numbers from the end?
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hi
anybody running with german SIPGATE?
my configuration don't works :-(
regards
[EMAIL PROTECTED]
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To UNSUBSCRIBE or update options visit:
htt
Title: Message
Even
if you could get that to work properly, which I dont know... the callprogress
detection is horrible; if you want to do that reliably you need a T1,ISDN or
IP interface to the switch (something that actually provides proper call
progress)
Sam
-Original Message-
I have an X100P connected to an extension of
a Panasonic PBX. When a call from the PSTN comes in, it is routed
directly to the extension where the x100p is . I want * to answer
the call, play a message and then transfer the call to another extension
via the Zap channel where the call was r
On Sat, 2004-05-08 at 16:28, HCQ wrote:
> I want to dial a sip device from extensions.conf but not connect the
> other party when it picks up,
> so I can send some DTMF before I connect the call.
> Is that possible ?
You must have missed this message:
From:
Eric Wieling
<[EMAIL PROTEC
Hi,
I want to dial a sip device from
extensions.conf but not connect the other party when it picks
up,
so I can send some DTMF before I connect the
call.
Is that possible ?
Thanks a lot!
HQ
Their current firmware doesn't allow to write to
the section for SIP registration. I am able to communicate with
it by dialing [EMAIL PROTECTED].
Also, you have to protect this box with Firewall
otherwise the whole world will be able to call through it.
Regards,
Wojtek
- Original Mess
>
> Don't know how far you've tried to take the 1204 in terms of functions,
> but we did the same thing over a two month period and found:
>
> 1. handling outbound calls on a "per pstn line" basis (eg, directing
> certain calls to certain pstn lines) is very non-standard and subject
> to future f
To All,
I am experiencing very strange behavior. It compiles just fine, on startup I
can see that it is connecting and authenticating properly (to mySQL),
however it's not using the DB. I can not access any mailboxes while using
mySQL module. Can not connect to check for messages, users can not lea
I have an X100P connected to an extension of
a Panasonic PBX. When a call from the PSTN comes in, it is routed
directly to the extension where the x100p is . I want * to answer
the call, play a message and then transfer the call to another extension
via the Zap channel where the call was rec
Not with the voiceage system, but with the new system you will be able to.
Mark
On Sat, 8 May 2004, nicolas wrote:
> if i have ordered one lic.
> and now i have realized i need two lic for one call (2 cannels one to
> provider one to sipphone)
> can i install 2. lic with another reg code ?
>
> n
After much reading and fiddling - I have the gnugk GateKeeper running
and can make calls from the H323 phone to the sip phone. Voice works
bi-directionally..
Calling from SIP to H323 gives me a problem...
Both gnuGK and Asterisk are on the same box. Someone said this was OK.
Others said No. I added
This hack is a tiny bit better:
http://lists.digium.com/pipermail/asterisk-users/2004-March/040186.html
John Baker
John Todd wrote:
At 10:31 AM -0400 5/8/04, Billy Huddleston wrote:
That won't work.. That'll DIAL multiple phones/extensions, but will only
bridge 1 of them when it auto-answers..
On Sat, 2004-05-08 at 20:43, Ryan Courtnage wrote:
> On 8-May-04, at 12:09 PM, Paul Tyreman wrote:
> > I have a problem with my Grandstream phone. I have set it up to use
> > DTMFMODE=info and I am able to transfer calls that have been made from
> > that
> > phone, but I am unable to transfer ca
We can provide you 2.2 cents a minute 1800 number
through SIP or H.323. We can also provide local access numbers and great
worldwide termination rates.
Regards,
Aram Ter-Martirosyan Senior Account Manager Hi-Tech Gateway, Inc. http://www.hi-teck.com 1225 Grand Central Ave. Glend
Hi list,
I'm interested in receiving incoming call to my Asterisk PBX thru an 1800 number. Anybody knows a provider with best minute rate? I heard that that Nufone can provide this service for around 3cents/min for calls made within 48 continental states. Any provider that can give better ra
I currently have
several asterisk servers geographically distributed (for automatic fail-over in
the event of either a network or server problem). My carrier delivers to each
server based on the same priorities that I have set in the DNS SRV records
which the clients point to.
Users always
Can you tell me if the md5secret stuff is broken? I noticed lastnight I
went by the wiki instructions and it didnt work. Alos if you change from
secret to md5secret then reload and do a sip show peer XXX it will say both
are set.
bkw
- Original Message -
From: "Olle E. Johansson" <[EMA
In cvs head version of chan_sip, there's two new CLI commands:
* sip show peer
Show details of peer - configuration, registration status etc
* sip show subscriptions
List active SIP subscriptions to extension state changes in Asterisk
/Olle
___
Asteris
Holger,
>From the Asterisk CLI, type: sip show peers
This will show you all users currently registered with Asterisk.
Brian D'Arcy
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Holger
Zimmermann
Sent: Saturday, May 08, 2004 9:46 AM
To: [EMAIL PROTECTE
On 8-May-04, at 12:09 PM, Paul Tyreman wrote:
Hi,
I have a problem with my Grandstream phone. I have set it up to use
DTMFMODE=info and I am able to transfer calls that have been made from
that
phone, but I am unable to transfer calls made TO that phone ??
I have the same problem (attempting to
I have the same problem with 2 ATA-286s, DTMFMODE=info and Dial command with
Tt options. Only the caller is able to transfer the call with the # key. The
callee is not able to transfer the call using # key, unless the codec is
ULAW and the DTMFMODE is inband. I suspect the problem is due the GS uni
Sorry to post this here also, but the biz list doesn't seem to have much
traffic yet.
I have a brand new SNOM 200 IP phone and also a new Siptone II phone
available on eBay, see http://tinyurl.com/2pbng
They are surplus after a customer cancelled an order. Please direct all
followup questions or bi
Hi,
I have a problem with my Grandstream phone. I have set it up to use
DTMFMODE=info and I am able to transfer calls that have been made from that
phone, but I am unable to transfer calls made TO that phone ??
I have tried every conbination of T and t in the extensions.conf file, but
all to no
> On Sat, 2004-05-08 at 10:52, Chris Wilson wrote:
> > I want to run different lines directly to different extensions on two
> > FXO analog interfaces. ie; Zap/1 goes to Ext. 101, Zap/2 goes to
> > extensions 102
> > Does anyone know of a way to do this?
>
> Yup! Check your trash folder. This was
> is it possible to see all online users?
>
> I have configure a isdn2sip gateway in the company I work.
> Now, the question: Is it possible to show all colleague which
> people where reachable with this gateway?
Have a look at Monastery http://www.unslept.com/monastery/
and/or http://graphics
On Sat, 2004-05-08 at 10:52, Chris Wilson wrote:
> I want to run different lines directly to different extensions on two
> FXO analog interfaces. ie; Zap/1 goes to Ext. 101, Zap/2 goes to
> extensions 102
> Does anyone know of a way to do this?
Yup! Check your trash folder. This was discussed on
Hello list,
is it possible to see all online users?
I have configure a isdn2sip gateway in the company I work.
Now, the question: Is it possible to show all colleague which people where reachable
with this gateway?
greetings and thanks,
Holger
__
At 9:27 AM -0400 on 5/8/04, Todd Lieberman wrote:
John, i think MGCP has this feature.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of John Todd
Sent: Friday, May 07, 2004 5:55 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Concept for line appearances an
At 10:31 AM -0400 5/8/04, Billy Huddleston wrote:
That won't work.. That'll DIAL multiple phones/extensions, but will only
bridge 1 of them when it auto-answers..
What we need is a way to have something like meetme call multiple extensions
and bridge them to a meetme confrence (all of them muted bu
Hi,
i want forward a call witch is comming over isdn (avm b1 witch i have) out to isdn
(same card 2. b channel).
The call is comming (one b channel open one is free) the forwarding is processed (snom
200) all seems correctly.
Then the message that the b channels all busy, but so is it not.
Forw
Hey everyone,
I want to run different lines directly to different extensions on two
FXO analog interfaces. ie; Zap/1 goes to Ext. 101, Zap/2 goes to
extensions 102
Does anyone know of a way to do this?
Thanks!
Chris
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[
Just a quick FYI..
I now only use iconnecthere for incoming
calls, I am phasing that out. If a company doesn’t want to give us the
information to properly use their services, then they don’t need my money. I am
now using voice pulse and I love it. I just wish they would have more exchan
Vic,
The problem you're having has been discussed multiple times on this list,
and can be easily seen using ethereal to inspect the timestamps contained
within the rtp packets sent "to" the 7960 phone. There are several issues
involved, including:
1. the cisco phones drop any rtp packet that is n
I am connecting a t100p to a b8zs, superframe, loopstart t1. Previously
I've attached to e&m wink and pri lines with no problems; however I seem
to be missing something.
Should it be fxols in the zaptel.conf (smartjack to x100p) or
fxsls?
With e&m wink the dnis set was 100, so it was easy to make
That won't work.. That'll DIAL multiple phones/extensions, but will only
bridge 1 of them when it auto-answers..
What we need is a way to have something like meetme call multiple extensions
and bridge them to a meetme confrence (all of them muted but the admin of
course, as it's a one way page) an
On Fri, 7 May 2004, Ian A. Underwood wrote:
> Joe Antkowiak wrote:
>
> > exten => 5101,1,Dial(SIP/5101,10,tA(intercom-tone))
> > exten => 5101,2,Congestion
>
> That's not too bad, but how do you page a group of phones...like a real
> intercom? That's what I'm dying to know!
in extensions.conf
You might try "callprogress=no"
It sort of sounds like noise (or analog phones) on the pstn side are
signaling the x100p to go off hook and possibly do other things.
> Happens quite often. X100P FXO card puts the PSTN line offhook, so that no
> calls go out or come in. The
Jeremy McNamara wrote:
Togan Muftuoglu wrote:
and what will happen if the box has more than one ethernet card
Mark is smarter than Voiceagehe will make it work.
Jeremy McNamara
That isn't saying much. The village idiot is smarter than VoiceAge. :-)
Regards,
Steve
_
On Fri, 7 May 2004, Brian Cuthie wrote:
> It seems that each time I get a new checkout of * from CVS my Cisco 7960
> works worse than before. I know this stuff's in flux, so I mention this
> in case it's news. Anyone else having trouble? What I'm seeing (er,
> hearing) is really choppy audio.
Togan Muftuoglu wrote:
and what will happen if the box has more than one ethernet card
Mark is smarter than Voiceagehe will make it work.
Jeremy McNamara
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Shahid wrote:
Happens quite often. X100P FXO card puts the PSTN line offhook, so that no
calls go out or come in. The outside callers get a busy siganl while inside
callers cant dial PSTN.
Its a DELL optiplex P3 128MB ram 500MHz processor.
Here is some more info: (see the zapata.conf in the end)
P
John, i think MGCP has this feature.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of John Todd
Sent: Friday, May 07, 2004 5:55 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Concept for line appearances and bridging:
anyone?
OK, here's a configuration
> if i have ordered one lic.
> and now i have realized i need two lic for one call (2 cannels one to
> provider one to sipphone)
> can i install 2. lic with another reg code ?
You shouldn't need two if the SIP phone and the provider are both using g.729
so long as you dont' expect Asterisk to "se
if i have ordered one lic.
and now i have realized i need two lic for one call (2 cannels one to
provider one to sipphone)
can i install 2. lic with another reg code ?
nico
Mark Elkins wrote:
> I Purchased 4 licences for my SCSI only machine. I do have a CDROM -
> with a mounted CD. The Registra
Hi folks,
I am using opengk to handle h323 calls.
* and my clients register at opengk successfully.
But everyone can register to my gk??
Is there a way to restrict the clients by using the authorisation of
h323.conf ??
Cheers,
Harald
___
Asterisk-Users
Hi,
I am stuck with my extensions.conf and would appreciate a small hint from the
ISDN experts.
What is the correct way to indicate a busy condition to a calling ISDN zap
channel (TE410P) when a local SIP ext. is busy?
I have
[pstn-in]
exten => *591,1,Dial(SIP/${EXTEN},45,r)
exten =>
This isn't really the issue. Up until a week ago or so everything worked
fine with a hallf duplex hub. Now it doesn't - so I suspect some code
change made in * is responsible. I think * must maintain backwards
compatibility with existing hardware or many people will get fed up with
constant
Nope I works differently, they use it in a few hospitals here in Sydney,
basically it works like the new gsm 'push to talk' service being rolled
out, basically limited number of frequencies, voice 'envelope' being
delivered as a best case availability basis.
It's not a 'held up' tdma style call fl
Use mpg123 version 0.59r
On Fri, 2004-05-07 at 17:57, Kyle Hagan wrote:
> When I put someone on hold audio doesnt play and i get
>
>
>
> mpg123: unknown option "mono",
>
>
>
> Any ideas. I searched wifi and archives.
>
>
>
> Kyle
--
Eric Wieling * BTEL Consulting * 504-899-1
Not sure if my other message got through. Wifi limitations with voip are a
function of # of concurrent active calls per access point (in addtion to which
codecs used). A single floor of the hospital might have many many access points.
If you just need a way to contact nurses on call, my guess is yo
Happens quite often. X100P FXO card puts the PSTN line offhook, so that no
calls go out or come in. The outside callers get a busy siganl while inside
callers cant dial PSTN.
Its a DELL optiplex P3 128MB ram 500MHz processor.
Here is some more info: (see the zapata.conf in the end)
Please direct m
Anyone got callwaitingcallerid working succesfull on
nortel/aastra/.../... 390 ADSI Phone?
It will be great if someone share some ADSI Scripts for these phones also.
Thanks
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At 4:00 PM -0500 on 5/7/04, Mark Spencer wrote:
> I Purchased 4 licences for my SCSI only machine. I do have a CDROM -
with a mounted CD. The Registration binary gives me a 'Segmentation
Fault'. Is this like telling me I can't register the licence?
Unfortunately - I only seriously scanned the m
Mark Spencer wrote:
I Purchased 4 licences for my SCSI only machine. I do have a CDROM -
with a mounted CD. The Registration binary gives me a 'Segmentation
Fault'. Is this like telling me I can't register the licence?
Unfortunately - I only seriously scanned the mailing list after buying
the keys
Joe Antkowiak wrote:
exten => 5101,1,Dial(SIP/5101,10,tA(intercom-tone))
exten => 5101,2,Congestion
That's not too bad, but how do you page a group of phones...like a real
intercom? That's what I'm dying to know!
--
/* Ian A. Underwood - [EMAIL PROTECTED] - http://www.agentgreen.org
There ar
I guess vocera doesn't have any RF engineers to tell them they can't do it.
Paul Mahler
[EMAIL PROTECTED]
Signate, LLC
PO Box 60430
Palo Alto, CA
94306
VoIP Systems, Training & Consulting
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Mu
John,
Check out www.vocera.com instead then.
Built for this exact situation.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James Moran
Sent: Saturday, 8 May 2004 2:27 AM
To: Asterisk
Subject: Re: [Asterisk-Users] WI FI IP phones??
No I'm
Excellent :)
That was the clunkiest thing I have ever seen.
Looking forward to your beta code.
Also, please make sure that users switching from the "old binary" to
the "new binary" does not lose their licenseI remember reading
something about only being able to register 3x.
On Fri, 7 May 20
Got so many people asking for it, here’s
what I used for the intercom announce:
http://www.jsci.net/asterisk/intercom-tone.gsm
It’s not great, but it does the
job. Actually trying to find something better…
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTEC
When I put someone on hold audio doesnt play and i
get
mpg123: unknown option "mono",
Any ideas. I searched wifi and
archives.
Kyle
Ok, here's a good one -- I've tried a lot of different things, searched
the archives, etc.
I've signed up with a VOIP provider and coaxed the SIP settings out of
them.
I have username (=phone), password, sip.provider.com and
proxy.provider.com. Using these settings, in Asterisk or X-lite, I can
Hi bkw
Yep, which is going to be a huge problem since it's only taking a line and
not doing any transmittal until after you get a line out, the line of course
is being rejected before I can even get there :(
Of course I can't even establish connectivity to the telco whilst having it
peered to the
try setting immediate=no
for that span
Jason
At 18:12 07/05/2004 +0100, you wrote:
Hi all
OK this may sound like a good one but maybe someone can tell me.
Simple context is - I want to unplug my existing conventional PBX from the
Telco and place * with it's TE410P in between.
Now the difficult
OK, here's a configuration challenge: I want to have certain line
appearances able to be "interrupted" by various other line apperances
elsewhere in the office. This is harder to describe than it is to
demonstrate, so I'll do that:
Let's assume I have Cisco 7960's on all desks.
1) Call com
On Fri, 2004-05-07 at 23:00, Mark Spencer wrote:
> > I Purchased 4 licences for my SCSI only machine. I do have a CDROM -
> > with a mounted CD. The Registration binary gives me a 'Segmentation
> > Fault'. Is this like telling me I can't register the licence?
> If you'll just be patient for a litt
I'm sure I saw a posting about someone updating the CVS with a more
richly featured voicemail system. What happened? Am I wrong?
Can't seem to find anything on this...
--
. . ___. .__ Posix Systems - Sth Africa
/| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE
/
* Mark Spencer; <[EMAIL PROTECTED]> on 07 May, 2004 wrote:
I Purchased 4 licences for my SCSI only machine. I do have a CDROM -
with a mounted CD. The Registration binary gives me a 'Segmentation
Fault'. Is this like telling me I can't register the licence?
Unfortunately - I only seriously scanned
Hello Everyone,
My company is about to deploy * as replacement for our existing
commercial Altigen PBX. Meanwhile, I've been trying to find the best
cost effective SIP VoIP phone which we can use in office for 20-30
employees, as well as a few remote staff.
Normally I wouldn't post about a VoIP
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