You can rule out the caller-id header brokenness by setting the SPA to not provide caller ID (or to provide a static caller ID) and seeing if it works.
If the corrupted caller-id headers are the issue you're having, you can look on the JIRA for a patch I made to work around this (http://track.sipfoundry.org/browse/XX-7307). I did run into some other standards issues surrounding DTMF and missing Contact headers (see http://track.sipfoundry.org/browse/XTRN-999 ) . The patch I wrote for the caller-id thing is hooked into a place where it could probably fix up the other issues I ran into, but I have no interest in arguing with folks about religion so I gave up. If you want to debug this in the hope its a configuration issue rather than one of the known-broken-no-hope-of-a-fix-with-sipxecs issues that you will ultimately run into, the tools at your disposal are: - packet capture of all traffic to/from the SPA while its failing - merged DEBUG logging from SIPX itself - turn on syslog at highest level from the SPA itself, sometimes this prints interesting information, esp WRT DTMF debugging Just to be clear: I spent a couple of months futzing with this POS and ran into two cases of it being dead broken vs. the standards. Since the sipxecs people were not at all interested in interoperability with a "broken" device I ditched it and got a working device (a Patton Smartnode) which I've had no problems at all with. There are other softswitches out there where the vendor did put in fixes to work around the lack of Linksys support and they work fine for folks using those, sipxecs (commercial or community) is just not one of them. -Eric On May 31, 2010, at 10:48 AM, Dave C. Lake wrote: > Hi Eric, > > I have tried using the config you gave me and seem to be getting most of it > working. I do seem to have problems with incoming calls. I remember you had > mentioned something about the caller ID header. I figure this may be the > problem. > > I so far can only get calls to hit the auto attendant. Once this is done it > doesn’t connect to any of other extensions if I dial them I get disconnected. > > Can you help me out on this? > > Regards > > Dave > > > <image003.jpg> > > Innovative Business Solutions > P.O. Box GR5102 > Unit 11, Bois D' Orange Commercial Complex, > Gros Islet, St. Lucia > Phone: (758) 4504427 > Fax: (758) 4582759 > Mobile: (758) 7248839 > Email: [email protected] > Web: www.ibsstlucia.com > > From: Eric Varsanyi [mailto:[email protected]] > Sent: Wednesday, May 05, 2010 12:00 PM > To: Dave C. Lake > Cc: [email protected] users > Subject: Re: SPA-3000 as PSTN Gateway > > On the 3102 you have to arrange for it to call a given 'user' via the sipx > proxy. > - Under PSTN Line, PSTN Caller Default DP ==> 2 > - Under PSTN Line dial plan 2 ==> S0<:201> , which roughly means 'wait 0 > seconds for dialing to end, then substitute nothing (the empty string before > the :, which is what you get from the PSTN) with '201' the SIP username > - Set up the sip proxy as your sipx installation > > On the sipx side: > - Create a user '201' (if you're manually setting up your telephones click > on advanced settings and get the password for that user) > - On the telephone you want to ring on inbound calls set it up to register > with sipxecs as user 201 > - Alternately you can set up a hunt group (from Features / Hunt Groups) > with an extension number and program the 3102 to call into that number, then > you can redirect inbound calls to other extensions using the hunt group rules > (a bit more flexible) > > I know I sound like a broken record, but the 3102 is just dead broken and may > not work right with some phones or sipxecs in some scenarios. What you're > describing though isn't one of them, I think all you need to do is change > S0<:0> to S0<:201> and register a phone as user 201. You can use tcpdump or > wireshark to look at the INVITE message from the 3102 and see what its really > putting out, there should be a 'To:' header with 4504427 > > Fax: (758) 4582759 > > Mobile: (758) 7248839 > > Email: [email protected] > > Web: www.ibsstlucia.com > > > > -----Original Message----- > > From: Eric Varsanyi [mailto:[email protected]] > > Sent: Tuesday, May 04, 2010 11:41 AM > > To: [email protected] > > Subject: Re: SPA-3000 as PSTN Gateway > > > > Dump of a "working" config below. > > > > The Spa3102 has many problems that no config will fix, it doesn't > > implement the SIP standards properly and Cisco has abandoned it > > support wise (the newest firmware is still broken and is quite old). > > > > The FXS side works more or less, though DTMF may not work depending on > > your ITSP. The FXO side kind of works as long as you don't allow it to > > import caller ID from calls (it corrupts the From: header, though I > > wrote a patch to sipxecs to work around the corruption), there was > > also a problem with REFER that did something bad to the AA. > > > > -Eric > > > > > > On May 4, 2010, at 10:32 AM, [email protected] wrote: > > > >> Hi Eric, > >> > >> I saw your post in the forum where you had setup an SPA -3102 as a > >> PSTN > > gateway. would you happen to have the configuration for this. I am > > attempting to do something similar. > >> > >> Thanks > >> > >> Dave Lake > > > > > > > > >
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