Hi,
I have a client using Mitel 5212 phones in SIP mode with a 1.4.21.1
Asterisk server (and a couple of previous 1.4 versions). They're
mostly happy with the combination except for this one issue.
For incoming calls only, either originating from other local SIP
phones or from a PRI, calls
Matt Watson wrote:
On July 19, 2008 11:22:08 am Mark Wiater wrote:
Hi,
I have a client using Mitel 5212 phones in SIP mode with a 1.4.21.1
Asterisk server (and a couple of previous 1.4 versions). They're
mostly happy with the combination except for this one issue.
For incoming calls only
Hello all,
I've got dialout= and callback= set in my voicemail.conf so that I
can have users return calls to folks who have left messages. They
really like this feature.
But when the callback is over, a normal hangup occurs instead of the
caller being put back into voicemail at the next message.
isn't snapanumber turning into ADA?
http://dl1.digium.com/ADA/ADA_MIS.pdf
There's a forum too.
Looks an awful lot the same.
Can't get it to work with my Thunderbird contacts the way
snapanumber does though.
Maybe it's a work in progress?
Mark
Dean Collins wrote:
Wow they must have bought
Lincoln King-Cliby wrote:
-Original Message-
Then starting at packet 3217 there are a series 6 of ICMP
Destination unreachable (Port Unreachable) messages from the
Asterisk server to the phone, with an RTP packet from the Phone
to the Asterisk server before each Destination
I asked the [EMAIL PROTECTED] for the documents and the tools that
are referenced in the admin guides and was told that I had to become
a registered user in the support section of the ww.sipura.com website.
They wanted name, title, phone # and type of support I provide for
the devices.
I think I
Russell Bryant wrote:
Ron McCarthy wrote:
Asterisk 1.4 im guessing? I did not know the Snom's worked with that,
Ill have to check it out then!
The way it is implemented in Asterisk is a bit interesting. It uses the
existing device state support (hints, BLF) to manage the buttons for shared
the mitel 3300 sends SMDR on TCP 1752. It spews software and hardware logs in
the same manner, different ports.
On 1/27/2010 11:00 AM, Steve Howes said:
On 27 Jan 2010, at 15:48, Jeff LaCoursiere wrote:
Sounds good to me, but without the spec I'm stuck in a catch 22!
tcpdump? (assuming
On 5/25/2012 3:18 AM, Lee, John (Sydney) said:
-- Executing [*1223*1**1900@incoming:78] Set(SIP/1900-08ee1da8,
DEVSTATE(Custom:cfalw1900)=INUSE) in new stack
I use
'Set(DEVICE_STATE(Custom:var)=BUSY)'
in my 1.4 dialplans to set device state.
mark
--
On 10/28/2013 3:59 PM, Ron Wheeler said:
I am reaching the same level of frustration.
I have tried to find the source of the problems.
We have IAX2 to our VoIP provider and SIP phones attached to the Asterisk - No
analogue.
I don't have any problems with IAX, but I hear some do.
We have a
On 6/18/2015 1:27 PM, Greg Woods wrote:
My provider claims that I am somehow sending an old number that
doesn't appear anywhere
snip
(I just moved from a POTS provider Century Link to a VOIP provider).
Set(CALLERID(number)=${var}) works fine for me.
Perhaps some debugging on the channel
On 10/27/2015 8:56 AM, Jonas Kellens wrote:
>
> I have changed this setting at Google but it brings me no success.
>
Jonas,
I've been using google calendar and Asterisk 1.8 for a couple of years
now without issue.
I have a note in my configuration that says that I'm using the Private
ICAL URL
In my case, username is the BTN. I also set the fromdomain to be the sbc
that I'm registering with. Externip might help also?
[paetec]
host=10.250.0.5
username=btn
fromdomain=10.250.0.5
dtmfmode=rfc2833
externip=10.255.0.2
I've used these settings on both registering and non-registering trunks,
On 1/18/2017 9:58 AM, Tech Support wrote:
For reconfiguring SIP phones? Can you give an example or short
explanation?
One can send a SIP notify with a check-config to the phone and have the
phone re-download it's configuration files from a provisioning server.
In the CLI, you can do a
On 8/31/2016 9:57 PM, D'Arcy J.M. Cain wrote:
exten => 55,1,Verbose(Door buzzer calling)
same => n,Set(toRing=)
same => n,ExecIf($["${DEVICE_STATE(SIP/user1)}" = "NOT IN
USE"]?Set(toRing=${toRing}/user1)
Failed. I checked the online docs and the syntax seems to be correct
but I get
On 9/1/2016 6:55 AM, D'Arcy J.M. Cain wrote:
So does the Dial command go directly to the registered device or does
it use the extension?
Yes, that's why you provide the technology part (SIP/, IAX/, DAHDI/)
I was assuming that it was going to the
extension's voice mail if it wasn't there
I think you had asked what phone works well with VPN's. I've had very
good experiences with Yealink using OpenVPN, never an issue.
I think I've heard that Snom does OpenVPN as well.
Mark
--
_
-- Bandwidth and Colocation
Yves,
Didn't you say that
AsteriskServer: 192.168.1.211
SIP-user: 165
?
On 12/21/2016 4:24 AM, Yves wrote:
. It is sure for 100% that there is no firewall or something else
mangeling
in between... another Hardphone works as expected using the same
Netzworkcable on the same Networkplug with
On 12/19/2016 10:26 AM, Yves wrote:
There are no SIP Packets arriving at my asterisk at all... and it has
nothing to do with a firewall or similar...
I can ping the phone from the asterisk,
If both of these items are true, then I'd look at the phone
configurations. Does the provisioning
On 4/18/2017 7:40 PM, Ernie Dunbar wrote:
Server network: 192.168.0.0/24
OpenVPN network: 10.8.0.0/24
Asus network: 192.168.1.0/24
The Asterisk SIP registration appears to be responding properly to
this - this is what I see when I do a 'sip show peer' for an Aastra
phone that's connecting
I've had Lefteris' code running for a few years without a problem.
I don't have a service key but I have entered my API key in the script
in the 'User defined parameters' section. You did that, right? What do
the other user defined parameters in your script look like?
On 7/19/2017 4:37 AM, Rahul
On 7/12/2017 5:30 PM, Holger Freyther wrote:
> I have to copy/mirror/forward the RTP streams for some selected call
> to an external address/port
I'd think that what you want to do might be best done outside of
Asterisk. If you're working with SIP, I'd suggest packet capture tools.
--
On 6/20/2017 8:42 AM, Tech Support wrote:
> I appreciate all the feedback, and replication seems to be a logical
> solution, but I was initially thinking about how to implement a solution
> within Asterisk to write the CDR's to two databases. Is that possible? Now
> I'm just curious.
On 5/25/2017 11:11 AM, Tech Support wrote:
I need to be able to tell whether or not the far end extension picked up
might a waitForSilence come in useful here?
--
_
-- Bandwidth and Colocation Provided by
On 5/31/2017 3:36 PM, Steve Edwards wrote:
I want to capture all SIP messages.
I have about 30 hosts in about 6 colos.
My first thought was dumpcap, but the output file name format bugs me.
What do you use for long term SIP capture?
voipmonitor is what you want.
--
Hi folks.
I have a couple of questions regarding RTP.
The background of my inquiry is that I have packet captures of SIP and
RTP traffic on an Asterisk and Broadworks SIP trunk and the RTP many
times has a time stamp that rewinds by 480 using g.711u. The Sequence
number continues to increment
On 8/30/2017 5:03 AM, Steve Davies wrote:
> Mark,
>
> You have cropped the image you inserted above and removed a very
> important part of the line you highlighted. I think is says ",Mark"
> after the time value - You can even see the un-cropped comma in your
> picture.
Thanks Steve,
I did omit
On 3/25/2019 4:45 PM, Mike Diehl wrote:
>
> > So, I don't think it's their network. I've taken pcaps of both legs of
>
> > example calls. On the provider-side, I see 2-way audio. On the
>
> > client-side, I only hear one side.
>
Mike,
In those pcaps, are you seeing the exact same RTP traffic
These two phones are not using the same extension, are they?
On 2/6/2019 8:49 AM, basti wrote:
> both phones are registered. and the hardware phone can also make calls.
> but an incoming call is not displayed and also not hearing.
>
> Call Waiting is also disabled.
>
> On 06.02.19 14:07, Cyril
On 4/19/2019 1:49 PM, Dovid Bender wrote:
> Mark,
>
> I am using PHP agi and when forking the call does not continue util
> the forked process is done. Am I doing it wrong?
>
>
> On Wed, Apr 10, 2019 at 4:27 PM Mark Wiater <mailto:mark.wia...@greybeam.com>> wrote
On 4/10/2019 3:54 PM, Dovid Bender wrote:
> I have an AGI that can sometimes take time complete. I don't want the
> dialplan to be held up by the agi. Is there any way to call it and have
> Asterisk continue with the dialplan?
>
Is there a reason you can't fork in the AGI and just return to the
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