> -Original Message-
> From: John Novack [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, August 15, 2006 10:40 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Manager Interface API's
>
>
> I, for one, didn't take his comment as anything other th
> -Original Message-
> From: Moises Silva [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, August 15, 2006 10:20 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Manager Interface API's
>
>
> Douglas. Please take this as a constructive comment. I
you dont like something, then change it yourself, they are not
> >> providing a payed service. The source is available AS-IS
> if you want
> >> it, and if you like it, take it; If you dont, just ignore
> it, try to
> >> not make peyorative comments.
redundancy also. That way, if one asterisk box
goes down, you don't have 50-100 clients completely
down.-brandon
On 8/16/06, Douglas
Garstang <[EMAIL PROTECTED]>
wrote:
Well, we're talking about
several dozen, maybe 100, companies, per Aster
> -Original Message-
> From: Jeremy McNamara [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, August 16, 2006 12:23 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Asterisk 'Hosting'
>
>
> Douglas Garstang w
P SIGNED MESSAGE-
> Hash: SHA1
>
> Douglas Garstang wrote:
> > Has anyone ever tried to run multiple instances of Asterisk
> on a single system, running each with a different username,
> and each in a separate base directory? Something like
> /home/pbx/business-1, home/pb
erisk-users] Asterisk
'Hosting'You might be able to use virtual NICs to
eliminate the problem with "non-standard" ports for a company's SIP
phones. Or real NICs using a couple of multi-homed cards.I
haven't tried it, though.
On 8/16/06, Douglas
Garstang
Has anyone ever tried to run multiple instances of Asterisk on a single system,
running each with a different username, and each in a separate base directory?
Something like /home/pbx/business-1, home/pbx/business-2 etc?
Did it work? I assume for every service that Asterisk runs, on each instanc
How
did you find out about 468*??? It's sure as poop not documented in the Polycom
Admin Guide anywhere.
-Original Message-From: Dovid Bender
[mailto:[EMAIL PROTECTED]Sent: Tuesday, August 15, 2006
11:16 PMTo: Asterisk Users Mailing List - Non-Commercial
DiscussionSubject:
> -Original Message-
> From: kjcsb [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, August 16, 2006 12:37 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] OPENSER / SER and Asterisk
>
>
> Absolutely. The SER/OpenSER documentation is terrible, an
*lol* The cryptic replies have been exactly my problem as well!
> -Original Message-
> From: kjcsb [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, August 16, 2006 12:37 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] OPENSER / SER and Asterisk
If it's open source and it's free...Then offer
them some money to make documentation for it hehe...
- Original Message -
From:
Douglas Garstang
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Tuesday, August 15, 2006
Can anyone recommend
the best Manager Interface API, putting language preferences
aside?
The python and perl
ones have bupkiss documentation. I can't understand why anyone would even write
an api and make it publically available without documenting
it.
Doug.
__
Yes, it might be a problem in our situation. We have three Asterisk boxes in a
'cluster'. The sip.conf is identical on all three. In that case, all three of
the Asterisk boxes in our cluster are going to send sip options messages to the
phones, which is silly.
Only the Asterisk box that a pho
re sent out at a set
interval, which, as you can see, is 60 seconds. If the device was previously
determined to be UNREACHABLE, the qualify packets will then be sent out every
10 seconds instead.
- Original Message -
From: Douglas Garstang <[EMAIL PROTECTED]>
- Non-Commercial Discussion
Cc:
Subject: Re: [asterisk-users] SIP Qualify
- Original Message -
From: Douglas Garstang
[mailto:[EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Non-Commercial Discussion [mailto:[EMAIL
Ok, what's the deal with qualify in sip.conf. The docs on the voip wiki at:
http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+qualify
state that it can take either yes, no, of a number which represents how long in
milliseconds between polling. I set it to 1000, (ie qualify=1000), did a
I thought I'd bounce
this around here before I opened a bug.
Phone A makes a
call to phone B. Phone B is still registered, but is physically turned
off.
Asterisk takes the
INVITE message from phone A.
Now,
1) It sends RINGING
back to phone A before it has even sent an INVITE to phon
Anyone using pyAst?
There's absolutely no docs.
It doesn't seem to
work anyway...
foo =
asterisk.manager.Manager()foo.connect("pbx1.xxx.com",5038)foo.login("callout","password")res
= foo.command("Show Channels")print res
this
yields:
[14:[EMAIL PROTECTED]:~]# ./test1.py Follows
even
article <[EMAIL PROTECTED]>,
Douglas Garstang <[EMAIL PROTECTED]> wrote:
> Can someone tell me why this is not valid...
>
> [start]
> exten => 1000,1,Answer
> exten => 1000,2,Wait,1
> exten => 1000,3,Ag
Can someone tell me why this is not valid...
[start]
exten => 1000,1,Answer
exten => 1000,2,Wait,1
exten => 1000,3,AgentcallbackLogin(1000||[EMAIL PROTECTED])
exten => 2000,1,Macro(DialProxy,115551212)
exten => 3000,1,Queue(testq45)
while this is:
[start]
exten => 1000,1,Answer
exten => 1000
Does anyone know how
I can set/increase the inter digit timeout on Asterisk assisted
transfers?
Doug.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.
We encountered a problem recently where, if an agent who received a call from a
queue, tried to transfer the caller with the Polycom transfer keys, it would
cause the Queue application to completely lock up.
This bug seems to be similar.
http://bugs.digium.com/view.php?id=7458
and this seems to
Where? I don't see any files dated Aug 2006.
Doug.
> -Original Message-
> From: Matt Florell [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, August 08, 2006 3:14 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Polycom 1.6.7 firmware?
>
>
> htt
When you say flaky howso?
It wouldn't be that when you try to edit buddies(ie directory) directly on the
phone, it freezes and reboots, is it?
Doug.
> -Original Message-
> From: Louis-David Mitterrand
> [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, August 08, 2006 10:43 AM
> To: asteri
I've discovered that music on hold, in general, doesn't behave in a way that
makes sense.
Here's my sip.conf:
sip.conf:
[general]
musicclass=natv_default
[3254101]
musicclass=classic
[3254103]
musicclass=natv_silence
Here's what happens with these settings:
3254101 calls 3254103:
325
Ok, this is weird..
sip.conf:
[general]
musicclass=natv_default
[3254101]
musicclass=classic
[3254103]
musicclass=natv_silence
When 3254101 puts 3254103 on hold, moh played to 3254103 is natv_default.
When 3254103 puts 3254101 on hold, moh played to 3254101 is classic.
I'm not following the ru
What's the best way to have silence for someone's moh?
Should I set their moh class to some bogus value?
Should I generate a file of silence, and use that?
Doug.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To
: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Re: DUNDi with SIP
>
>
> Works fine for me.
>
> Douglas Garstang wrote:
> > Didn't work Brad. I changed fromuser to dundisip1 from
> dundisip2. The
> > first Asterisk b
nd it fails the call.
> -Original Message-
> From: Steve Totaro [mailto:[EMAIL PROTECTED]
> Sent: Friday, August 04, 2006 8:51 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Re: DUNDi with SIP
>
>
> Works fine for
g a
fromuser setting to yoyr peer entries.-
Brad -Original Message-From: Douglas
Garstang [mailto:[EMAIL PROTECTED]]Sent:
Thu Aug 03 15:58:50 2006To: Asterisk Users
Mailing List - Non-Commercial
DiscussionSubject: RE:
[asterisk-users] Re: DUNDi
chmod
u-w -directory.xml
The
phone will quietly not be able to write to the contacts
directory.
Doug.
-Original Message-From: Stephen Murphy
[mailto:[EMAIL PROTECTED]Sent: Thursday, August 03, 2006
4:09 PMTo: asterisk-users@lists.digium.comSubject:
[asterisk-users] Pre
erisk Users Mailing List - Non-Commercial
DiscussionSubject: RE: [asterisk-users] Re: DUNDi with
SIP
I forget if this does what you want, but try adding a fromuser
setting to yoyr peer entries.- Brad -Original
Message-----From: Douglas Garstang [mailto:[EMAIL PROTECTED]]Sent:
List - Non-Commercial
DiscussionSubject: RE: [asterisk-users] Re: DUNDi with
SIP
I forget if this does what you want, but try adding a fromuser
setting to yoyr peer entries.- Brad -Original
Message-----From: Douglas Garstang [mailto:[EMAIL PROTECTED]]Sent:
Thu Aug 03 15:58
Brad...
Here's the INVITE that the second asterisk box receives from
the first Asterisk box, after the second asterisk box sends a
Proxy auth message to the first. The first sends the dundisip
userid, but for some reason the second asterisk box is
matching it against the From: 3254101 address.
nf, were
> the secret= lines literal, or did you replace the actual
> secret with "password"?
>
> - Brad
>
> -----Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Douglas Garstang
> Sent: Thursday, August 03, 200
es literal, or did you replace the actual
> secret with "password"?
>
> - Brad
>
> -Original Message-----
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Douglas Garstang
> Sent: Thursday, August 03, 2006 11:43 AM
> To: Asterisk Use
If you want to pass
> a variable
> set the variable, but prefix it with __ (2 underscores)
>
> Set(__DNID=${DNID})
>
> Douglas Garstang wrote:
> > Oh... That's real nice. I was considering using SIP instead
> of IAX to trunk calls between Asterisk boxes as IAX has
> -Original Message-
> From: Douglas Garstang
> Sent: Thursday, August 03, 2006 10:30 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] IAX Variables
>
>
> I'm a little confused about something. If IAX2 doesn't
I'm a little confused about something. If IAX2 doesn't support passing
variables like the accountcode for example, between Asterisk servers, why do
all the examples have it defined in iax.conf? If the protocol doesn't support
it, then why have it set for an IAX user agent?
Doug.
___
Ok... it'd be great if someone could explain this to me...
User A on pbx1 wants to dial User B on pbx2. We do a local lookup and don't
find user B on pbx1, so we do a DUNDi lookup of user B, get a result, and place
the call to user B on pbx2 with IAX2.
When pbx2 calls the AGI script that dialls
Oh... That's real nice. I was considering using SIP instead of IAX to trunk
calls between Asterisk boxes as IAX has some severe limitations in regards to
passing variables. A few people said 'use SIP!' because you can set the SIP
headers. Looks like that isn't an option!
> -Original Message
c86
To: ;tag=as004358bd
Call-ID: [EMAIL PROTECTED]
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact:
Content-Length: 0
---
<-- SIP read from xxx.yyy.142.162:5060:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP xxx.yyy.142.
d)
> definitions. You appear to be attempting to authenticate as the
> originating callerid (3254101).
>
> - Brad
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Douglas
> Garstang
> Sent: Wednesday, August 02, 2006 6:45 PM
&
, so I only list it's contexts in it's own file. Then,
> sgw1 publishes what IT can handle. There's a matching e164
> and internal
> context on each server to tell the others what it can take.
>
> On Wed, 2006-08-02 at 16:35 -0600, Douglas Garstang wrote:
> >
> -Original Message-
> From: Watkins, Bradley [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, August 02, 2006 4:45 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [asterisk-users] Re: DUNDi with SIP
>
>
> Errr... I should learn to pay attention to what I'm wr
> -Original Message-
> From: Aaron Daniel [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, August 02, 2006 4:02 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [asterisk-users] Re: DUNDi with SIP
>
>
> On Wed, 2006-08-02 at 15:52 -0600
> -Original Message-
> From: Aaron Daniel [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, August 02, 2006 4:02 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [asterisk-users] Re: DUNDi with SIP
>
>
> On Wed, 2006-08-02 at 15:52 -0600
> -Original Message-
> From: Watkins, Bradley [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, August 02, 2006 4:07 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [asterisk-users] Re: DUNDi with SIP
>
>
> The way to make this work is to define a sip user/peer
> -Original Message-
> From: Watkins, Bradley [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, August 02, 2006 4:07 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [asterisk-users] Re: DUNDi with SIP
>
>
> The way to make this work is to define a sip user/peer
; driver is being
> parsed.
>
> - Brad
>
> -Original Message-----
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Douglas
> Garstang
> Sent: Wednesday, August 02, 2006 5:59 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
>
> -Original Message-
> From: Tony Mountifield [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, August 02, 2006 3:49 PM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Re: DUNDi with SIP
>
>
> In article
> <[EMAIL PROTECTED]>,
> Dougla
> -Original Message-
> From: Tony Mountifield [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, August 02, 2006 3:49 PM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Re: DUNDi with SIP
>
>
> In article
> <[EMAIL PROTECTED]>,
> Dougla
ll? If not, is the CLI output the same as when you try it via DUNDi?
Second, are your keys generated properly, with public keys shared
between the two boxes OK? I had a lot of DUNDi problems initially, and found
that my keys were the problem.Alex
On 8/2/06, Douglas
Garstang <[EM
: [asterisk-users] DUNDi with
SIPYou can use an unchanging password. It's not as
secure, but it will provide functionality.Alex
On 8/2/06, Douglas
Garstang <
[EMAIL PROTECTED]> wrote:
So
what are the options?> -Original Message- > From:
Aaron Dan
SIP
>
>
> I'm talking about the rotating DUNDi secret that is stored in dbsecret
> in iax.conf. It doesn't exist in the SIP channel.
>
> On Wed, 2006-08-02 at 13:43 -0600, Douglas Garstang wrote:
> > Secret? Do you mean sbsecret in sip.conf?
> >
> &
;m talking about the rotating DUNDi secret that is stored in dbsecret
> in iax.conf. It doesn't exist in the SIP channel.
>
> On Wed, 2006-08-02 at 13:43 -0600, Douglas Garstang wrote:
> > Secret? Do you mean sbsecret in sip.conf?
> >
> > > -Original Message---
> -Original Message-
> From: Tony Mountifield [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, August 02, 2006 2:01 PM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Re: DUNDi with SIP
>
>
> In article
> <[EMAIL PROTECTED]>,
> Dougla
;
> Using the SECRET variable for sip doesn't work.
>
> On Wed, 2006-08-02 at 13:11 -0600, Douglas Garstang wrote:
> > I've trying to use DUNDi with SIP to see if it works around
> some limitations of IAX2.
> >
> > I do a DUNDi lookup, get my SIP path, a
I've trying to use DUNDi with SIP to see if it works around some limitations of
IAX2.
I do a DUNDi lookup, get my SIP path, and try to dial it. Asterisk immediately
says 'No such host', eventhough that's the path is just returned!
[Aug 2 13:07:05] == Spawn extension (global_vmdeposit, u92203
> -Original Message-
> From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, August 02, 2006 11:54 AM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Limitations of IAX
>
>
> On Wednesday 02 August 2006 09:42, Joshua Colp wrote:
> > As previously pointe
Ok... it'd be great if someone could explain this to me...
User A on pbx1 wants to dial User B on pbx2. We do a local lookup and don't
find user B on pbx1, so we do a DUNDi lookup of user B, get a result, and place
the call to user B on pbx2 with IAX2.
When pbx2 calls the AGI script that dialls
> -Original Message-
> From: Joshua Colp [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, August 02, 2006 7:42 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Limitations of IAX
>
>
> - Original Message -
I'm about ready to give up on IAX2. It seems to have some SERIOUS limitations.
Incoming PSTN call comes into to user A on pbx1. We look for user A locally,
and don't find them. We then do a DUNDi lookup, get a path, and dial user A on
pbx2 with IAX2. User A picks up the call.
When IAX passes th
> -Original Message-
> From: Michiel van Baak [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, August 01, 2006 3:57 PM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Dundi and Dial Arguments
>
>
> On 15:39, Tue 01 Aug 06, Douglas Garstang wr
> -Original Message-
> From: Mitch Sharp [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, August 01, 2006 3:06 PM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Dundi and Dial Arguments
>
>
> Dundi question:
>
> Is there a way to pass dial arguments to switch => DUNDi as
>
> -Original Message-
> From: Joshua Colp [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, August 01, 2006 10:25 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [asterisk-users] IAX and Accountcode
>
>
> - Original Message -
> -Original Message-
> From: Joshua Colp [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, August 01, 2006 8:34 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] IAX and Accountcode
>
>
> - Original Message -
Does the accountcode from a SIP user agent get passed to IAX when trunking a
call from one asterisk box to another? The SIP caller id, extension etc do get
passsed, so why not the account code? It's a standard field.
Doing a 'iax2 debug' doesn't even show the accountcode field.
Good grief. IAX2
You have a config generator script for the Polycom XML files? What did you
build that with?
-Original Message-
From: Greg Boehnlein [mailto:[EMAIL PROTECTED]
Sent: Sat 7/29/2006 7:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
I actually did get it to work, by removing _all_ spaces from the for line...
for (x=0;${x}<3;x=${x}+1) {
This works for me. It's just a matter of finding WHICH space is breaking it.
-Original Message-
From: Rushowr [mailto:[EMAIL PROTECTED]
Sent: Sat 7/29/2006
What
about sip registration replication?
What
about SIP subscription replication?
What
about BLF replication?
What
about using DUNDi to replicate applications for redundancy?
How
would you handle different phones ability to failover if they don't do it so
well?
How
would handle the fact
I was just experimenting with AEL2, and tried to use a for loop as per the
example.
Here's what I have.
context new_pbx_betty_start {
_X. => {
for (x=0; ${x} < 3; x=${x} + 1) {
Verbose(x is ${x} !);
}
};
}
Here's the output.
The var x
> -Original Message-
> From: Steven [mailto:[EMAIL PROTECTED]
> Sent: Friday, July 28, 2006 6:44 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Re: bugs.digium.com
>
>
> "This is not a bug. It is just the way it works.
>
> The "sip debug" output is "verbose" output
> -Original Message-
> From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED]
> Sent: Thursday, July 27, 2006 8:48 AM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] bugs.digium.com
>
>
> On Thursday 27 July 2006 10:32, Douglas Garstang wrote:
>
I opened bug
#0007490 the other day. The issue was that when you do a 'sip debug' on the
Asterisk console, there was no way to have this output go _only_ to the messages
file. Someone with the id of 'russell' in his infinite wisdom has deemed that
this isn't a bug, closed it, and given me -2
Aside from the main .cfg file, You can use https or ftps to get the rest
of the xml configuration files.
Doug.
-Original Message-
From: C F [mailto:[EMAIL PROTECTED]
Sent: Wed 7/26/2006 8:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussio
...
>
>
> On Mon, 24 Jul 2006, Douglas Garstang wrote:
> > Not for our users. We held focus groups, and the Polycom's
> won in terms of ease-of-use over all the other phones investigated.
>
> Which other phones did you investigate specifically?
>
> Our us
> -Original Message-
> From: C F [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, July 25, 2006 4:46 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Caller ID on Transfers
>
>
> so you are telling me that all the time you have been bitching you
>
> -Original Message-
> From: C F [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, July 25, 2006 4:34 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Caller ID on Transfers
>
>
> what version of asterisk you using? I use the ${BLINDTRANSFER} a lot
> -Original Message-
> From: Anthony Rodgers [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, July 25, 2006 4:21 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Caller ID on Transfers
>
>
> The 'o' option to the Dial() command, along with using b
ded transfer, in our case a receptionist.
On 7/25/06, Doug
Lytle <[EMAIL PROTECTED]>
wrote:
Douglas
Garstang wrote:>> talking to). Blind transfers show
the original caller.>>>> Doesn't seem to be
happening that way with Poly
> -Original Message-
> From: C F [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, July 25, 2006 4:06 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Caller ID on Transfers
>
>
> In my experience (although I didn't test this as I type now) using t
> -Original Message-
> From: Joshua Colp [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, July 25, 2006 10:31 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [asterisk-users] Caller ID on Transfers
>
>
> - Original Message -
> -Original Message-
> From: Doug Lytle [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, July 25, 2006 2:27 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Caller ID on Transfers
>
>
> Douglas Garstang wrote:
> >&g
n 7/25/06, Doug
Lytle <[EMAIL PROTECTED]>
wrote:
Douglas
Garstang wrote:>> talking to). Blind transfers show the
original caller.>>>> Doesn't seem to be happening
that way with Polycom phones and blind/attended transfers.> Both are
> -Original Message-
> From: Joshua Colp [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, July 25, 2006 9:47 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [asterisk-users] Caller ID on Transfers
>
>
> - Original Message -
> -Original Message-
> From: Doug Lytle [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, July 25, 2006 1:37 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Caller ID on Transfers
>
>
> Douglas Garstang wrote:
> >&
> -Original Message-
> From: Brian Capouch [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, July 25, 2006 12:53 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] RDNIS and IAX2
>
>
> Douglas Garstang wrote:
> > I&#x
> -Original Message-
> From: Joshua Colp [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, July 25, 2006 8:54 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [asterisk-users] Caller ID on Transfers
>
>
> - Original Message -
> -Original Message-
> From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, July 25, 2006 12:48 PM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Caller ID on Transfers
>
>
> On Tuesday 25 July 2006 14:37, Douglas Garstang w
> -Original Message-
> From: Joshua Colp [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, July 25, 2006 8:41 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Caller ID on Transfers
>
>
>
>
> - Original Message
I have
three phones here with extensions 2944093, 3254103 and
9220371.
2944093 calls 3254103. 3254103 transfers 2944093 to 9220371. We want the
caller id of 2944093 to be presented on the display of
9220371.
However, the caller id of the transferer, 3254103, is appearing. This
doesn't mak
I stumbled across a draft rfc for IAX2 at:
http://www.eguy.org/presentations/2006/draft-guy-iax-01.html#sec.new
It describes all the IAX2 'information elements' sent when a new call is
established.
Here's the list.
http://www.eguy.org/presentations/2006/draft-guy-iax-01.html#sec.new
When I issu
I'll probably get blasted for this. I hope I'm wrong, and then a little
blasting is ok. It appears that Asterisk may have let us down again as a
'carrier grade' solution.
1. User A calls User B. The call is bridged.
2. User B wants to transfer User A to user C. When this happens, User B's phone
Not for our users. We held focus groups, and the Polycom's won in terms of
ease-of-use over all the other phones investigated.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Mon 7/24/2006 7:08 PM
To: Asterisk Users Mailing Lis
I'll probably get blasted for this. I hope I'm wrong, and then a little
blasting is ok. It appears that Asterisk may have let us down again as a
'carrier grade' solution.
1. User A calls User B. The call is bridged.
2. User B wants to transfer User A to user C. When this happens, User B's phone
> -Original Message-
> From: Doug Lytle [mailto:[EMAIL PROTECTED]
> Sent: Monday, July 24, 2006 2:15 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Transfers - No ringback or moh
>
>
> Douglas Garstang wrote:
&g
> -Original Message-
> From: Doug Lytle [mailto:[EMAIL PROTECTED]
> Sent: Monday, July 24, 2006 1:35 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Transfers - No ringback or moh
>
>
> Douglas Garstang w
> -Original Message-
> From: Doug Lytle [mailto:[EMAIL PROTECTED]
> Sent: Monday, July 24, 2006 12:56 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Transfers - No ringback or moh
>
>
> Douglas Garstang wrote:
&g
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