On Wed, Mar 15, 2006 at 11:26:45AM +, Paul Hayes wrote:
> The SPA-2100 is the only one to support T.38 at the moment though. SPA-2002
> has the ability to support t.38 (i.e. it has the processing power required)
> but
> the firmware support isn't there yet.
Per our sales contact at Cisco for
I'm trying to figure out how Asterisk decides how often it will send SIP
NOTIFY's to an ATA when a voicemail message is waiting for the user on the
server.
>From watching, it seems to be completely random. Sometimes 10 seconds
apart, then 33 seconds, then 13 seconds, etc. Each time causes a "rin
On Tue, Jan 03, 2006 at 06:43:16PM -0500, Michael Stearne wrote:
> I am having trouble with FC3.
>
> After doing a yum update (of 1264 packages) I still cannont compile
> 1.2.1 from source:
>
> make[1]: `libedit.a' is up to date.
> make[1]: Leaving directory `/usr/src/asterisk-1.2.1/editline'
> m
I generally use CentOS. Haven't tried CentOS 4 with Asterisk yet, but I'm
sure it'd work fine.
It's generally less of a "moving target" than Fedora is as far as updates
are concerned. CentOS 3.x will get updates as long as Red Hat is providing
them whereas FC1 servers and FC2 servers we set up a
On Sat, Dec 31, 2005 at 10:05:19AM +0100, Olle E Johansson wrote:
> We're currently planning a new generation of chan_sip that will have a
> different authentication scheme, not based on the from: header unless
> it's a local policy to require the From: header to be the same as the
> Digest auth
Posted this to -dev, but it may be more appropriate here as I haven't
released my "patches" for it...
I've run into a couple issues relating to RPID.
I have an Asterisk 1.2.1 installation doing SIP for SPA-2002 and PAP2-NA
ATA's. From the Asterisk box, we then do SIP to a VoIP provider who handl
On Wed, Oct 26, 2005 at 10:12:09AM -0400, Matt wrote:
> Does anyone know if SIPURA SPA-2002's support DNS SRV records?
Yep, it does (as does its brother PAP2-NA).
Ray
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Disregard my previous post. I was thinking the uplink to your telco was via
SIP.
Ray
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On Wed, Oct 19, 2005 at 09:16:48AM -0400, Dave Wise wrote:
> I am using a * w/a PRI for the TDM interface to telco.
> I am running Asterisk CVS-HEAD-05/29/05-03:59:44
> All was working well until I needed a SIP ATA to be unlisted.
>
> in sip.conf, on the account I used:
> restrictcid=yes
>
> I am
Perhaps they dont' like the codec you're offering in your INVITE message?
Ray
On Fri, Oct 14, 2005 at 01:36:17PM +, Obelix wrote:
>
> I have been receiving a lot these 488 "Not Acceptable Here" from a number of
> providers. What could the problem be?
>
> What is the most common cause of th
of lack of ANI.
My workaround would be to detect numbers which require ANI (1-8XX, 911, etc)
and ensure that the From header is always populated for these calls.
Mostly I want to find out how things are _supposed_ tow work though. :-)
Thanks for any info.
Ray
--
Ray Van Dolson
On Tue, Oct 04, 2005 at 09:59:56PM +0200, Olle E. Johansson wrote:
> I think this is a bug. Please open a report in the bug tracker,
> attaching all the requested information. If a re-invite fails, we should
> not cancel the call. I am afraid that is exactly what is happening here
> and would like
Opened bug #5384.
http://bugs.digium.com/view.php?id=5384
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On Tue, Oct 04, 2005 at 09:59:56PM +0200, Olle E. Johansson wrote:
> > 1. Asterisk sends the initial INVITE (requesting G711u)
> > 2. SIP/PSTN gateway says it's trying (100) and its media server begins
> > sending
> >G711U RTP traffic.
> > 3. SIP/PSTN gateway sends a 183 session progress messa
On Thu, Sep 29, 2005 at 08:54:42PM -0500, Kevin P. Fleming wrote:
> Ray Van Dolson wrote:
>
> >Our SIP/PSTN gateway provider seems to think that Asterisk should initiate
> >a
> >renegotiation to G711 when it sends the 488 message rejecting T38.
>
> This is not corr
When qualify is set to yes in sip.conf for a "friend" and the OPTIONS packet
gets returned with an ICMP port unreachable message, what is the behavior of
Asterisk?
It looks to me like Asterisk tries sending the OPTION request again right away
(well within a second or two).
Some of our devices are
On Fri, Sep 30, 2005 at 05:05:41PM -0700, Ray Van Dolson wrote:
> I'll have to do some tcpdumps later to see what SIP messages are sent now when
> I hit hold and what happens to the rtp streams.
Did some briefly.
No RTP stream initiates from the Asterisk server. When I hit hold on th
On Fri, Sep 30, 2005 at 06:47:51PM -0500, Kevin P. Fleming wrote:
> >The ATA's are Sipura SPA-2002's and I have MOH Server set to 899 on each.
>
> Take that out, you don't need it.
>
> None of this is needed; Asterisk will stream MOH to ATA 2 all by itself,
> just by the fact that ATA 1 put ATA
I've been having problems getting MusicOnHold to work, so I've dumbed down my
setup to as simple of a setup as I can.
Asterisk 1.0.9. SIP ATA's (Sipura SPA-2002's)
<---> <--->
Both ATA's have public IP's. No NAT'ing going on here. Reinvites are allowed
so the media stream bypases Asterisk
On Fri, Sep 30, 2005 at 10:20:12AM -0400, Joel Newkirk wrote:
> How can we achieve this, short of 'reciting' the unit number aloud at
> the beginning of the placed call?
Hmm, could you just put the full address (including unit no.) in the E911
database for the corresponding numbers assigned?
You
Disclaimer: Yes, I know faxing over G711 is unreliable. :-)
We're running Asterisk 1.0.9 which talks to a Audiocodes SIP Gateway. We're
running Sipura SPA-2002's as ATA's and faxing within our own voice network is
working. If we try and fax out to the world however, we're running into a
problem.
We have a small app that runs asterisk -rx "sip show peers" (etc) and gathers
some info and reports it back somewhere else. This is called from a cron job
essentially and doesn't really run all that often (once a minute). We noticed
some gaps in the results we were getting and dug in a little fur
On Tue, Aug 16, 2005 at 10:37:01AM -0600, Damon Estep wrote:
>
>Is there a method in SIP to set the CALLING number type to national and the
>calling number plan to isdn? I am dealing with an issue where a media
>gateway is not sending the correct values and would like to know if SIP h
On Mon, Aug 01, 2005 at 02:46:55PM -0400, Huddleston, Robert wrote:
> Is it my imagination or did I just drop off the list for several days
> somehow... I didn't get any posts since Friday...
>
See the IRC channel. The list has been broken for a couple days. If you look
at the archives on list
r RTP server, but it appears to me as if the traffic is
already going outbound although I have no way to know if it's valid or being
accepted, etc.
Just hoping someone can verify that I'm doing the correct setup. Let me know
if there's any additional info I can provide.
Ray
--
Ray
The STUN server was extremely easy to set up. Just check ou the MyStun
sources (you have to use CVS), compile and run the server executable. That's
about all there is to it.
Ray
On Tue, Jun 28, 2005 at 11:07:48AM -0700, hank wrote:
> how easy is it to set up a stun server? with asterisk amd wil
We've been feeling our way along with the NAT stuff (using SIP) as well.
At this point we are fairly small, so the keep-alive packets are not too bad.
What type of user load are you at and what are the specs on your Asterisk box?
I'm concerned we may run into this as well.
We do have the luxury t
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