NO JOY Extensions going offline, at two sites

carl0s

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Weird one.
I'm using Incredible PBX 13 ISO.

First site uses Cisco / Linksys spa508g, 8-line phones. The only thing I hate about these phones is that they're so difficult to Google for problems about - they're the same as the other SPA Sipura things, but other people might have SPA504G, or SPA502G, or SPA303. Googling "Cisco asterisk" returns results on all the 79xx actual-Cisco phones. If it was a Polycom, I would just googling "Polycom Soundpoint asterisk sip unreachable" for example.

Anyway, model-designation rant over:
They were OK with single line configured.
When 2 lines were configured to different extensions, I found that the first extension would become "unreachable", either through qualify, or with no IP address showing when doing sip show peers.

I removed the additional extension configuration on the handsets, and played around some more, NAT settings (causes sip info or options keepalives I think), register timeouts, etc, and it was OK in the end. Register timeouts didn't sort it (the problem still happened, but it fixed itself sooner each time). I think giving up on the multiple extensions sorted it. I tried different SIP ports for each extension, even though this isn't necessary because the phone knows what account the SIP packet is intended for.

Now I set up a new system today, with the same phones, but the addition of a DECT Gigaset N300IP (+a540h handset), and they all seem fine, except for the Gigaset N300IP, which within the first hour of testing, would be unavailable and the calling Cisco phone would show "service unavailable" intermittently when trying to call to the Gigagset internally. This was just today and I haven't looked into it more, but I suspect it's the same problem and I expect to be hearing about it tomorrow.

The only thing in common, I think, is that they both have the same Netgear FS-728 poe switch.

Any ideas what might be wrong?
 
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carl0s

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It's got to be either some quirk in the switch, or a bug in the version of Asterisk, right?

Hey I'm a "new member" who joined 6.5 years ago :D
 

carl0s

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Only other thing I have done at both sites is disable chan_pjsip. I did this because Asterisk segfaulted when I tried to set up my ip-authenticated (no user/pass) sip trunk via pjsip, and I got fed up of having to choose chan_sip every time I created an extension.

I realise I might have been able to configure the trunk successfully, but if my misconfiguration of a sip trunk in FreePBX causes a segfault in Asterisk then that tells me that chan_pjsip has potentially devastating bugs, and this was right at the first hurdle.
 

geopeterwc

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A shot in the dark, @carl0s: ... When configuring each of the extensions on the 8-line Cisco/Linksys phones, did you assign a different SIP port to each of the extensions ... and have a match in both the phone and the iPBX extension configuration?

I haven't used the Cisco/Linksys phone that you've referenced, but I've made the mistake with other multi-line phones and ended up scratching my head for hours trying to figure out what's wrong before stumbling over the configuration bit. Hope that your problem has a solution that's that simple!

/Pete./
 

carl0s

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Hey Pete, thanks for the ideas. I tried all of different ports on the phone side, the extension config side, etc.

That bug report is 100% it. The timing of the report, and fix, is perfect. That box is still on 13.6.0 and the new one is most likely (don't have external access to it just yet.. only went in today/yesterday) 13.6.0 as well. The bug report cites the problem being mostly prevalent with dual-extension phones, and DECT phones. Well, on the new install I only saw the problem when dialling to the DECT phone (they have 17x Cisco SPA508G desk phones, and 1x DECT Gigaset phone).

I will update Asterisk, and 99% (dare I say 100%? :) ) expect that to be the end of it! So much for running newer versions eh. My biggest site is still on Asterisk 1.4 and has been for years :D
 

carl0s

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I thought there was an "update-asterisk" script that downloaded & compiled Asterisk with whatever modules/options are required for the distro, installs to the expected locations, etc?

Or am I going to have to manually compile and install from source myself?
 

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