On Jul 13, 2010, at 7:01 PM, Charl Barnard wrote:
>
>>
>> Which at least partially explains the lower cost... there aren't any
>> TDM
>> circuits involved.
>>
>
> I always suspected the Big Transition to IP was more vendor push than
> operator pull, so I'm keen to hear why a sharp guy like Kev
neers couldn't
explain the details to me.
Joe
On 7/13/2010 3:14 PM, Bryan Scott wrote:
> I could be wrong, but I don't think libss7 handles TCAP queries (currently).
>
> I'd be inclined to ask very big company how they put the TCAP over VPN (is it
> over SIGTRAN over an
I could be wrong, but I don't think libss7 handles TCAP queries (currently).
I'd be inclined to ask very big company how they put the TCAP over VPN (is it
over SIGTRAN over an IPSEC tunnel?).
(If libss7 does support TCAP queries, I too am interested...)
-- Bryan
On Jul 13, 2010, at 12:42 PM,
I would think it's kind of obvious, but since it isn't :)
Don't put a dchan in spans 2, 3, and 4, and don't block out those channels as
available cics. Also, aren't there 32 channels in an E1, not 31?
--
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-32
dchan=16
span=2,2,0,cc
There's really no "easy" way to do that without moving the IVR to
Asterisk. I can't think of a way to have the IVR system tell Asterisk
to close it's connection and redirect the call to the PSTN without
using the manager interface.
Now, if you have code on the IVR that can remotely control Asteri
I've seen this before too.
I have a couple of Asterisk boxes hanging off a switch using SIP. I also used
to explicitly Answer() as the call came in from the switch. My guess is our
switch converts SIP signaling straight to SS7, because anytime a call came in
from a specific in-state AT&T tand
Matt, Paul,
How do you set the CAC? Is that something that can be done by Asterisk SS7
(Matt et al), or (Paul) are you doing translations in your switch for the trunk
group?
Local calls on our network don't need a carrier, but toll calls do.
--
_
On May 3, 2010, at 11:11 AM, Matthew Fredrickson wrote:
>
> Bryan Scott wrote:
>> They do match I have them separated by "pointcode"...
>>
>> Switched back to 1.6.2.6 (1.6.1.18 was giving me grief on some SIP RTP
>> stuff).
>>
>>
&g
On May 3, 2010, at 11:11 AM, Matthew Fredrickson wrote:
>
> Bryan Scott wrote:
>> They do match I have them separated by "pointcode"...
>>
>> Switched back to 1.6.2.6 (1.6.1.18 was giving me grief on some SIP RTP
>> stuff).
>>
>>
&g
> -- Sent from my Palm Pre
> ____
> On May 3, 2010 8:33 AM, Bryan Scott wrote:
>
> OK. Downgraded to 1.6.1.18, DAHDI is at 2.3.0, libss7 is 1.0.2 with
> SLC patch (not that it's needed).
>
> Can't get into TMT to see isupMsgTrace yet...
OK. Downgraded to 1.6.1.18, DAHDI is at 2.3.0, libss7 is 1.0.2 with
SLC patch (not that it's needed).
Can't get into TMT to see isupMsgTrace yet... (need to call around to
get login info; never used it except when TAC has already opened it).
chan_dahdi:
group=2
signalling=ss7
ss7type = ansi
net
; grayed out so if yours differ, you'll have to recreate the node.
>
> My f-link is indeed SLC 0, apolgies for implying otherwise, btw.
>
>
Bryan Scott wrote:
> I've moved the f-link to the last channel (48), but otherwise yes, it
> looks like my other SS7 links and
What about just as a sniffer, I.e. Not having to interrupt a circuit
to put in the bridge, but maybe as an analyzer on a monitor port?
On Friday, April 30, 2010, Horacio J. Peña wrote:
> Answering myself, hope it is useful for somebody:
>
> #include
> #include
> #include
> #include
> #include
ISUP in open manager and click on the point code,
> you see the range of 00026 through 00048 on the right pane.
>
> Can you select all of those, right click and choose "Query Circuit
> Range" and let me know what that says?
>
> -Paul
>
> Bryan Scott wrote:
>
ry it and tell me if works.
>
> Saludos,
> Nahuel Greco.
>
>
>
> On Fri, Apr 30, 2010 at 7:01 PM, Bryan Scott wrote:
>>>>> Set your SS7 trunk to SLC 1. I couldn't make it work
>>>>> on SLC 0. No idea why.
>>>>>
>>>>
>>> Set your SS7 trunk to SLC 1. I couldn't make it work
>>> on SLC 0. No idea why.
>>>
>>
>> The Taqua is 1 too? Regardless of Asterisk's setting, it shows the SLC 0.
>> Setting the Taqua to 1 causes the F-Link to go down and Asterisk complains.
>>
> Yes. Both the taqua and the asterisk ar
Not at my desk to do some copy/paste of chan_dahdi.conf, but I suspect
it is ignoring the 'slc' parameter. After reloading Asterisk still
insists on using SLC of 0. I'm going to play a little more to see
what I can put in.
On Friday, April 30, 2010, Paul Timmins wrote:
>
On Apr 29, 2010, at 6:23 PM, Paul Timmins wrote:
> I started with CIC 73 so they would match the asterisk channels for
> better debugging. Set your SS7 trunk to SLC 1. I couldn't make it work
> on SLC 0. No idea why.
The Taqua is 1 too? Regardless of Asterisk's setting, it shows the SLC 0.
I hate feeling like a noob, but I'm at my wit's end and need a second (or
third, or fourth) set of eyes.
I have a 4 T1 Digium card connected to one of our Taqua T7000 switches and a
customer.
The customer is on span 1 (first T1 of the card). That PRI is working fine,
for all intents and purp
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