to help you if you're behind NAT but it
seems that changing the register host from natrelay.deltathree.com
to sipauth.deltathree.com fixed it for me.
-- Luke
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the SIP retransmission
patch is?
-- Luke
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I've also noticed that my SIP phones (snom [12]00) seem to deregister
themselves after some time, and not be able to re-register until
Asterisk is restarted. This problem only manifested with the latest
CVS.
-- Luke
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Luke Howard | PADL Software Pty Ltd | www.padl.com
Turn off reinvites and that will likely fix this.
Thank you; it does.
-- Luke
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what is WMI ?
Typo: MWI == Message Waiting Indication.
-- Luke
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'
in macro 'iconnecthere'
== Spawn extension (local, s, 1) exited non-zero on 'SIP/515-Office-b922'
-- Luke
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I can confirm that.
After 10/15 seconds, outgoing sip calls are
destroyed.
Interestingly, I'm seeing incoming SIP calls destroyed after 10-15
seconds, and outgoing SIP calls destroyed almost immediately.
-- Luke
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Not Exist back from 1.2.3.85
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on the asterisk *CLI interface? Do I have a conf messed up? If
not have any SIP phones registered, the output is fine.
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While I'm at it, does anyone have details of other suppliers who can supply
the snom phones. At one stage I though digium were doing that, but they
don't seem to now?
In Australia, try Nick Haler at Salient Networks. His e-mail address is
Nick.Haler at salientnetworks.com.au.
-- Luke
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Luke
can get the gateway to work with g723.1 and gsm, but I can't get it to
work with ulaw or alaw. My phone device is a quicknet linecard.
The problem I'm having appears to be purely a signalling one.
-- Luke
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Luke Howard | PADL Software Pty Ltd | www.padl.com
I remember at some point getting 488 media errors if I didn't enable
gsm.
As I mentioned, I'm getting 480 Temporarily not available, not
488 media errors.
I tried the grotesque hack of making handle_response() ignore 480
errors, which *seems* to work. Hmm.
-- Luke
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Luke Howard | PADL
I tried the grotesque hack of making handle_response() ignore 480
errors, which *seems* to work. Hmm.
I tried that, and at least for me it has a number of subtle side effects:
Or maybe we should send an ACK to them -- I need to read the SIP RFC...
-- Luke
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Luke Howard | PADL Software
Moreover, if anyone has a packet trace of iConnectHere's SIP client
making a call (which presumably does work), then please send it
along... it would be interesting to see whether Asterisk, at fault
or not, can be made to work around this properly.
-- Luke
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Luke Howard | PADL Software Pty
I should add that in my config I'm totally behind NAT, both asterisk and
an ATA186 that talks to it.
Hmm, both our SIP phones and Asterisk are on visible IPs.
-- Luke
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) exited non-zero on 'SIP/515-Office-b8b1'
in macro 'iconnecthere'
== Spawn extension (local, 933145207135, 1) exited non-zero on 'SIP/515-Office-b8b1'
-- Luke
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that
sip show
registry now shows nonsense characters (wrong pointer to the strings?)
BTW: thanks for cleaning up the $(PROC) settings in the Makefiles! I can now tell gcc
to
make code for my Via C3 without all the Makefile editing!!
Michiel
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PSTN...
-- Luke
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. Is anyone planning on licensing G 723.1 for use with Asterisk? As
I understand it, buying a LineJACK won't suffice if the card's DSP is
not actually used.
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necessarily
needing an Asterisk server there too (although doing so would, relative to
bandwidth costs, still be cheaper :-)).
I assume 64k ISDN is OK for GSM + SIP overhead?
-- Luke
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