SIP gateway <-> IP phone?

Have a question or want to start a discussion? Post it! No Registration Necessary.  Now with pictures!

Threaded View


Hi

Since I'm still stuck at how to make a Linksys 3102 work OK with the
free Axon PBX for Windows*... I was wondering if it's possible to
configure the 3012 (or the GrandStream HT-488 that I also have) to
have it ring an SIP phone directly, ie. no need for either an SIP PBX
or some VoIP account on the Net? For single use, that would solve the
issue.

Thank you!

* that stupid thing goes off-hook when forwarding incoming PSTN calls
to the PBX, regardless of whether an extension does ultimately pick up
the call

Re: SIP gateway <-> IP phone?


Quoted text here. Click to load it

From http://www.sipura.com/support/spa3000faq/Section_2.html#5 it would seem
it is possible, although I've never tried.

I'm not sure about going off-hook immediately; registering the SPA-3000 on
Asterisk it's definitely possible to avoid that problem, as explained at
http://voxilla.com/forum-viewtopic-t-1335.html . Perhaps that can be adapted
to the Axon PBX, of which I know nothing.

Enzo


Re: SIP gateway <-> IP phone?


On Mon, 21 Aug 2006 19:13:37 +0800, "Enzo Michelangeli"
Quoted text here. Click to load it

Thanks for the link. I'll check it out. It's always nice to keep
something as basic as possible.

Quoted text here. Click to load it

A sharp-eyed Axon user noticed that, by default, it's supposed to play
on-hold music (I hear silence) instead of just keep RINGing the remote
extension. Changing that setting solved that issue

But I'm still stuck with the internal softphone's microphone not being
hear by the remote PSTN caller, although it works find PSTN ->
softphone. I know just about nothing about SIP or RTP, so it might be
something obvious. Actually, that may also explain why I got no music
on hold before..

For the curious out there, here's the output of an incoming PSTN call
getting into the Linksys, being forwarded from FXS to FXO (at least,
that's how I think of it), RINGing extension 101 (a softphone) through
the Axon IP PBX, having a (one-way,sigh) conversation, and then
finally hanging up:

http://codecomplete.free.fr/syslog.txt

If anyone has any hint about why RTP data only seem to travel PSTN ->
softphone but not the other way around, I'm all ears :-)

Thank you.

Re: SIP gateway <-> IP phone?


[...]
. Changing that setting solved that issue
Quoted text here. Click to load it

Uhm, I confess I haven't studied the log with the care it would require. To
save me time, can you please detail all the IP addresses of softphone, Axon
IP PBX, Sipura 3102 (is it 192.168.0.233 ?), router, any third party (e.g.,
provider proxies) in the piture...

In particular, which piece of equipment has the routable IP adress
"83.157.189.2" that appears in the log at one point? Have you perhaps
enabled STUN support on the SPA-3102, and, as a consequence, is the latter
telling the softphone to send its RTP data to the external IP address of the
router rather than the internal IP address of the SPA-3102? This would be
fine for peers on the Internet, but if the softphone is on the same LAN,
it'll never work...

From what I can see, there are a couple of things I don't like very much:

- The IP address 127.0.0.1 (i.e. localhost) being used in several places of
the SIP/SDP dialogue. This is an address that only works for interprocess
communications on the same host.
- The absence of a trailing ";rport" parameter on the SIP "Via:" headers,
which means "no support for http://www.ietf.org/rfc/rfc3581.txt " and
therefore "likely problems with NAT traversal".

Also, I can't see and SIP message referring to an extension 101... And why
do you do an "FXS to FXO" forwarding? I thought you were trying to receive a
call on the FXO interface and ring a softphone, without involving the FXS
interface at all...

Cheers --

Enzo


Re: SIP gateway <-> IP phone?


On Tue, 22 Aug 2006 16:41:03 +0800, "Enzo Michelangeli"
Quoted text here. Click to load it

Sorry. I wish I understood what the logs say, and just keep the
relevant part instead of just dumping this massive piece of stuff...

Quoted text here. Click to load it

The softphone and the Axon PBX are both running on the same W2003
Server test host with IP = 192.168.0.233. The Linksys is connected to
the same switch with IP = 192.168.0.253, with no software firewall
between the two (I checked in Control Panel, and W2003 doesn't seem to
have a software firewall like XP). I call into the Linksys from a
regular analog landline, the Linksys sends an SIP message to Axon,
which looks up the extension, makes it ring, I pick up the call : I
can hear the remote PSTN caller, but he can't hear me.

FWIW:
- Skype works fine, so I guess the sound card is full-duplex (picked
up that bit from the FAQ on SJphone's site
- I guess the same issue with a GrandStream BudgeTone 100 (model 101)
- SJPhone is actually worse than X-Lite 3, as the former doesn't do
sound either way.

Quoted text here. Click to load it

It's the dynamic IP assigner to the ADSL modem.

Quoted text here. Click to load it

I barely know what STUN does, so haven't touched this. By default,
Voice > SIP > NAT Support Parameters > STUN Enable = No, and STUN
Server is empty.

Quoted text here. Click to load it

I do, but the only way I could get the Linksys to notify the Axon PBX
of an incoming call is by doing this:
- PSTN Line > PSTN Ring Thru Line 1 = yes
- User 1 > Call Forward Settings > Cfwd All Dest = fxo (where "fxo" is
the account used by the Linksys to register with Axon in the "External
line" section)

But maybe this is totally not the way to set it up, in which case,
what else can I try? Incidently, since I only want to ring a single
remote phone, I guess I could remove the Axon PBX from the equation,
and just tell the Linksys to hit the phone's static IP through the Net
directly?

Like you, I suspect that it's sending its RTP data elsewhere, or maybe
the issue is due to the fact that I don't use the FXS and expect data
to flow between the FXO and an SIP phone? I doubt it's codec (they use
G711u by default) or RTP ports (no firewall).

Is there an IRC channel somewhere where I could ask the experts?

Thank you.

Site Timeline