Everyone's credentials should still work. If you do not have access
to something you should have, please let me know. I'll send out more
details tomorrow, but very tired right now. Lot's of people to thank
for getting this done.
Couple items of note
- if you subscribed to receive digest on mai
Hi,
Adding asterisk was quite simple. The peer config can be seen in my initial
post of the thread.
I simply had to add a user in sipx (for giving asterisk the right of placing
external calls over other gateways) and added asterisk as an unmanaged
gateway.
Dialplan configuration debends on your sy
On Mon, Jul 26, 2010 at 4:13 PM, Michael Scheidell wrote:
> you got ATT? Verizon? Level3 trying to sell you enterprise SIP trunks?
>
> they want you to send to a different port? want to send to you on a
> different port?
>
> Got it nailed.
nice
>
> (hint: you don't use sipxbridge as your SBC.
On Mon, Jul 26, 2010 at 6:48 PM, McIlvin, Don
wrote:
> I must have some part of set up incomplete. What is needed for the “send
> profiles” to work properly? What service is sending to the phone to initiate
> the restart (i.e. message apparently timing out)? How does the phone know to
> obey ( aut
wiki, tracker, mailing list as we move to new infrastructure. if we
run into problems like we did last night, we may be swapping back,
I'll keep you posted.
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i had that problem with Bria and it turned out to be the silence detection.
turned that off and it started working properly...
Mike
On Mon, Jul 26, 2010 at 7:34 PM, Jean-Hugues Royer wrote:
>
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You should get a trace for this to debug it, but there is a
chance you run into a reported problem of ACK not b
So SipXecs has no event logging when a Job fails to see what happened?
I changed a polycom phone (IP650) to sip 3.1.3revC, made sure it was
registered. In SipConfig added a speed dial to the user, navigated to
the phone and pressed "send profiles". Sure enough the Profile job
completed and the
That is what we do. I just wasn't going into the details.
On 7/26/2010 3:25 PM, Tony Graziano wrote:
In the event you are not supporting remote users, that should be
simple enough. If you are supporting remote users you would need to do
that ONLY for originating calls from the ITSP.
On Mon,
I figured out how to have Mixed Versions of firmware for Polycom Phones
In this instance.. 3.1.3revC for my IP450, IP550, and IP650s.. And 3.2.3
for the IP335s (not supported in 3.1.3revC).
I do a lot of phone restarts with updates, so having to manually toggle
between the versions dependin
autentication, NOT registration, as they keep telling me.
we don't REGISTER, we AUTENTICATE.
and, hell! set up your SBC to be port 5060, on a DIFFERENT PUBLIC IP
address, and use pfsense to xlate it.
more to come.
(but, a normal ITSP account, trying to send to port 5070, like they
wanted, w
In the event you are not supporting remote users, that should be simple
enough. If you are supporting remote users you would need to do that ONLY
for originating calls from the ITSP.
On Mon, Jul 26, 2010 at 4:21 PM, Matthew Kitchin (public/usenet) <
mkitchin.pub...@gmail.com> wrote:
> Just an FY
Just an FYI. For Verizon, I simply translate inbound 5060 -> to 5080
at the router level, and all is well.
On 7/26/2010 3:13 PM, Michael Scheidell wrote:
Ready to document this and write it up.
you got ATT? Verizon? Level3 trying to sell you enterprise SIP trunks?
they want you to send to a
I don't think AT&T will agree to change the port to anything but 5060. They
have always flatly refused in the past. Not sure they use Acme packet in all
of their SIP products either.
In other words your server is behind NAT supporting remote users and you
state your RTP ports. You disable sipxbrid
Ready to document this and write it up.
you got ATT? Verizon? Level3 trying to sell you enterprise SIP trunks?
they want you to send to a different port? want to send to you on a
different port?
Got it nailed.
(hint: you don't use sipxbridge as your SBC. they go an acme SBC in
front of th
It's not an Asterisk configuration problem, it's an Asterisk problem
(I've experienced this before as well). FreeSWITCH has the same issue
currently. I'd report this to Digium but don't expect it to get fixed.
Josh Patten
Assistant Network Administrator
Brazos County IT Dept.
(979) 361-4676
O
on the 'static ip registration' and attempting to send to udp port 5070.
SEEMS that if you don't have something in the user field under itsp
account, and/or you don't have a p-asserted identity, it won't send to
port 5070, no matter what you do.
this has happened, on and off, its almost impos
Hello list members,
we have successfully connected an asterisk as a gateway to our sipx
installation. This gateway is only used for outbound calls and everything
seems to be working fine.
The only problem we figured out is the attended transfer While blind
transfer works without any problems the a
We are getting a strange error when we try to receive inbound calls. The
specific error is in the attached XML file, on lines 8 and 17 via sipviewer.
Specifically it appears our bridge is translating(?) the contact wrong, we
are getting:
CALL #1: From Bridge -> Primary SIP
484 Address Incomple
This is in regards to this thread I had started before -
http://forum.sipfoundry.org/index.php?t=msg&th=13427&goto=47362&S=173b77
b3f1eb007e6c353c47189b19db#msg_47362
(I did not know how to reply back to the list from that thread since I
no longer had that email from the list, and posting a reply
Hi all,
I'm experimenting the ACD feature.
When all agents of a queue are logged off, sipx give the busy tone and
close the call.
Is it possible to play a message before closing the call?
Currently I'm running sipx 4.0.4.
Thanks
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I am using Grandstream HT503 to connect analog lines to
Sipx. I have one line that is causing problems. When it
connects I ge
> Hi Dale,
>
> I know they are, unfortunately, following the "wide" SIP standard.
>
> Since media is relayed though sipX, Media Relay, when
> sipXBridge is used, I was wondering if it was possible to get
> an event when first media arrives? This in not relevant for a
> normal sipX deployment,
From: sipx-users-boun...@list.sipfoundry.org
[sipx-users-boun...@list.sipfoundry.org] On Behalf Of Talbot, Peter
[peter.tal...@nrtnortheast.com]
Currently inbound calls are coming in from our ITSP, hitting our sipbridge, but
then from the bridge they are
From: Sven Evensen [sven.even...@onrelay.com]
I know they are, unfortunately, following the "wide" SIP standard.
Since media is relayed though sipX, Media Relay, when sipXBridge is
used, I was wondering if it was possible to get an event when first
media a
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