I did an installation using a Dialogic 8 port FXO/FXS DMG1008.
Quite a lot to configure but it worked i would say flawlessly with
sipxecs 3.8.
If i remember correctly and i think i do, grouping ports was easy.
*Vänliga Hälsningar/Best Regards*
/Ola Samuelson
/
2009-12-18 14:46, Eric Varsanyi skrev:
Yeah, I wish I could have stuck with the Audiocodes 4FXO/4FXS I
originally bought.
I discovered after I tried to get it working that this is the only
member of the mediapack family that sipXecs doesn't support. I was
willing to figure out its rather complex configuration manually but I
ran across a note on the Wiki saying sipXecs won't allow you to use
the ports other than as a single group -- since I have several
different classes of FXO (LD, local, doorphones) that was a show
stopper and I returned it.
It didn't help they didn't want to give me the current firmware for a
brand new unit w/o a service contract, I think they're not interested
in smaller deployments.
I decided to the try SPA3102's because there are tons of people using
it with Asterix and some chance it worked at all. Since the units are
1 port there's no restriction with sipXecs on creating a gateway per
port which I couldn't do with the Audiocodes. I'd love to switch to
something less lame, the damn thing still ignores CPC call supervision
for no apparent reason.
Cost isn't really a factor, I just wanted something that was 'known to
work' (at least in some sense of the word). Maybe I should just get 6
of the smallest Audiocodes gateways (?) if they are really the best in
class?
-Eric
On Dec 18, 2009, at 1:48 AM, Picher, Michael wrote:
I cringe when I have to work on those gateways…
Hey, but they’re cheap!
Hmm… “You get what you…”
*From:* sipx-users-boun...@list.sipfoundry.org
<mailto:sipx-users-boun...@list.sipfoundry.org>
[mailto:sipx-users-boun...@list.sipfoundry.org] *On Behalf Of *Eric
Varsanyi
*Sent:* Thursday, December 17, 2009 4:08 PM
*To:* sipx-users@list.sipfoundry.org
<mailto:sipx-users@list.sipfoundry.org>
*Subject:* [sipx-users] DTMF between SPA-3102 and Polycom 650s/335s
I've had a heck of a time getting calls through sipXecs between a
Linksys 3102 gateway (FXO side) and Polycom 650 and Polycom 335
phones. I thought I would send this in in case someone else has
issues like this and is googling around.
Some data points:
- The IVR system in sipXecs only responds to RTP telephony events,
it does not (nor should it IMO) attempt to decode DTMF from the
encoded audio stream
- RFC 2833 allows the ID to be used to send 'telephony events'
(DTMF tones amongst some others) in the RTP stream to be dynamically
negotiated based on a Mime-Type field of 'telephony-event'
- The default setup of the Polycoms (3.2.2.0477) by sipXecs
4.2-17527 (and whatever form of 4.0.4 came off the ISO installer)
leaves the polycom set to negotiate telephony events on dynamic ID 127
- A call originated by the SPA3102 (inbound call) negotiates
dynamic ID 101 and all is well
- A call originated by the Polycom negotiates id 127
- The SPA3102 apparently ignores the negotiated dynamic ID and
expects telephone events on RTP ID 101 only, thus outbound DTMF from
the Polycom has no effect
Some quality time with wireshark indicated the outbound calls from
the Polycoms were indeed asking for telephony events and the 3102 was
acknowledging the negotiation with the an ACK accepting ID 127 -- yet
the actual RTP DTMF packets were being sent by the polycoms and
ignored by the 3102.
A working configuration (for me) that allows the IVR to work from
inside polycoms, and calls from the 3102's FXO -- and allows the
polycoms to send DTMF on outbound calls:
Linksys SPA 3102 5.1.10(GW)
sipXecs SVN checkout 4.2-17527 (though had problems with 4.0.4 as well)
Polycom 650's and Polycom 335's 3.2.2.0477 (though same issues at the
previous release which is supported on 4.0.4)
Linksys PSTN Line configuration tab:
DTMF Process INFO: yes
DTMF Process AVT: yes
DTMF Tx Method: Auto
DTMF Tx Mode: Strict
DTMF Tx Strict Hold Off Time: 180 (likely this one doesn't matter,
I just haven't reset it after experimenting)
Symmetric RTP: Yes
(also beware that sipXecs will choke on inbound calls with north
american style caller ID from the PSTN because the 3102 doesn't quote
the callerid string, I have a hack/patch for sipXecs for this if
anyone is interested)
Device/Phone config for Polycoms:
In the DTMF section:
viaRtp - checked
rfc2833Control - checked
rfc2833Payload - 101 (NOT default of 127)
If you don't check rfc2833Control calls to SPA3102 will work fine but
calls to the IVR will not negotiate DTMF and you won't be able to get
your voicemail.
-Eric Varsanyi
PS: The 3102's have sure been a bear to get working. DTMF, CPC
supervision (which still mysteriously doesn't work, falling back on
busy tone recognition at this point), and the caller-ID formatting
being broken have burned hours. I returned a new Audiocodes gateway
when I discovered it was the one sipXecs unsupported model in the MP
line (4FXO/4FXS) and to get current firmware (for a brand new unit)
you need to buy a support contract -- I sure wish I had kept it, it
might have been a pain to configure but its probably not basically
broken the many ways the 3102's are.
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