On 6/13/13 16:20 , Matthew J. Roth wrote:
It's hard to be certain without seeing a full SIP trace, but I think the
INVITE
with the internal IP is actually a re-INVITE that Asterisk is sending to
establish a native bridge between the SIP friend and the SIP gateway to PSTN
converter.
It's
Andreas Sikkema wrote:
On 6/13/13 16:20 , Matthew J. Roth wrote:
It's hard to be certain without seeing a full SIP trace, but I think the
INVITE
with the internal IP is actually a re-INVITE that Asterisk is sending to
establish a native bridge between the SIP friend and the SIP gateway
Thanks to everyone for the responses. I really appreciate it. I'll
answer all questions and suggestions in this one email. (at the bottom)
On 6/14/2013 9:43 AM, Nunya Biznatch wrote:
Howdy All,
They say opinions are like belly buttons, everybody has one.
(that's the clean version of the
Hello all.
I'm having trouble resolving an issue with our Asterisk system that
seems to have popped up recently (no one knows for sure when the issue
started). I'm still somewhat of a Asterisk newb and have been tasked
with administrating the system as the previous administrator has left
the
...
For redundant/failover of Asterisk checkout HAAST at
www.generationd.comhttp://www.generationd.com The HAAST product sits between
Linux and Asterisk, monitors for failures etc, and then fails over to another
Asterisk box. It effectively creates a low-cost cluster, moving IP's etc to
Interesting product that I was very interested in, but the licensing has
one huge glaring problem. Be sure to read the FAQ carefully. If your
hardware fails and you replace almost anything in the machine, you have to
pay for the product again.
On Sat, Jun 15, 2013 at 10:42 AM, Michelle Dupuis
On 15/6/13 7:00 pm, Carlos Alvarez wrote:
Interesting product that I was very interested in, but the licensing has
one huge glaring problem. Be sure to read the FAQ carefully. If your
hardware fails and you replace almost anything in the machine, you have to
pay for the product again.
Not to
Jun 15 13:06:05 VERBOSE[30232]: -- Executing Dial(Zap/1-1,
zap/g1/1XX|20|tT) in new stack
Jun 15 13:06:05 VERBOSE[30232]: -- Called g1/1XX
Jun 15 13:06:08 VERBOSE[30232]: -- Zap/2-1 answered Zap/1-1
Jun 15 13:06:08 VERBOSE[30232]: -- Attempting native bridge of Zap/1-1
and
On Sat, Jun 15, 2013 at 2:18 PM, Daniel Tryba dan...@tryba.nl wrote:
Jun 15 13:06:05 VERBOSE[30232]: -- Executing Dial(Zap/1-1,
zap/g1/1XX|20|tT) in new stack
Jun 15 13:06:05 VERBOSE[30232]: -- Called g1/1XX
Jun 15 13:06:08 VERBOSE[30232]: -- Zap/2-1 answered Zap/1-1
Jun 15
They are either incompetant or lying to you. Call appears to succeed (it
is answered) and gets disconnected after 23s. You are not generating the
message, so the calls gets back to you telco.
Most likely someone is filtering on callerID (which is a good thing IMHO).
Set the callerid to one
What about projects like YATE, DiaStar, and mobicents (even though I
have no idea how to approach that project in terms of downloading
etc..). Are there any mature SS7/SIGTRAN stacks?
Kind Regards,
Nick.
--
_
-- Bandwidth and
On Sat, Jun 15, 2013 at 10:28:50AM -0600, Nunya Biznatch wrote:
Answer - There's a couple reasons I'm thinking this way, which may be
misguided so thanks for making me think about it. First is redundancy.
Offloading the PRIs and analog phones from the primary PBX means if
there's an issue, I
On Sat, Jun 15, 2013 at 03:02:41PM -0400, Andre Goree wrote:
Setting the CID did not work, unfortunately :(
[...]
I'm going to try another number that we have through them in hopes
that it'll complete and I'll let you know if that works. Do you have
any other suggestions on what you think
On Sat, Jun 15, 2013 at 4:03 PM, Daniel Tryba dan...@tryba.nl wrote:
On Sat, Jun 15, 2013 at 03:02:41PM -0400, Andre Goree wrote:
Setting the CID did not work, unfortunately :(
[...]
I'm going to try another number that we have through them in hopes
that it'll complete and I'll let you know
I am having an issue with iax2 for the fax.
when i am issuing iax2 show peers, i am getting
*CLI iax2 show peers
Name/UsernameHost Mask Port Status
Description
iaxmodem/iaxmod (null) (D) 255.255.255.255 0
Unmonitored
1 iax2 peers [0
Do a pri set debug (or whatever it is called in 1.4 (zap?)) Zap/Zap
bridging should work, it did on my PRIs and still does with DAHDI. Only
thing I can think of is the TON/NPI might be a problem (but doubt it
since SIP/Zap works).
Thanks so much for your suggestions.
I'm running 1.0.x
Enable logging and see what happens when you start and stop the iaxmodem. Obviously, it doesn't
register, but the messages might be helpful. Since iaxmodem is somewhat older, you might have to
disallow call tokens in iax.conf.
jg
--
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