Kristan

Another one I can't find Whereabouts is this? I can't see a setting on the extension for it? Also, how does this fit in with dialog permissions? We previously implemented this "how to decide who can pick up a call" with rather complicated dialog permissions strings to prevent certain people picking calls outside their team. What would be really nice is what the Mitels do, and make one of the softkeys into a "pickup" button when a phone in their group is ringing. I'm not quite sure SIP allows for that sort of thing though

Aha, that works - do the "Agents" & "Calls" stats refresh automatically or "push" AJAX style or is it a case of pressing F5?

Probably just me being blind, but in the release notes for 3.0 it says : I've my extension as a member of the agent group, but I can't see where in my user login that the stats are displayed? ta

Just installed it now, I'll let you know any problems

I don't mind - I just need to get this box working for the customer as soon as possible!

Ok, after some playing around, I'm now seeing this : [3] 2008/01/01 00:21:49: PSTN: Channel 0 going to TALKING [7] 2008/01/01 00:21:49: b0e94e45@pbx#880541485: RTP pass-through mode [7] 2008/01/01 00:21:49: 3c2aac33c832-xeq3o76mhyze@snom300-000413258E11#6f76d7b172: RTP pass-through mode [5] 2008/01/01 00:21:50: PSTN: Busy Tone detected on 0 (version: 2.4.2)n [5] 2008/01/01 00:21:52: Last message repeated 3 times However the call shows as "connected" in the calls screen and as far as the phone is concerned, it's also connected - yet the call hasn't been connected and is busy. What should the correct sequence be? I'm also seeing this at the end of the call : [3] 2008/01/01 00:21:53: PSTN: Channel 0: Hangup [5] 2008/01/01 00:21:53: PSTN: Channel 0 goes onhook [5] 2008/01/01 00:21:53: PSTN: enable_callerid 0 [3] 2008/01/01 00:21:53: PSTN: Channel 0 going to GO_ONHOOK [5] 2008/01/01 00:21:53: PSTN: Busy Tone detected on 0 (version: 2.4.2)n [3] 2008/01/01 00:21:54: PSTN: Channel 0 going to IDLE I guess it shouldn't be detecting a busy tone then? See, this is why I prefer ISDN.....

Ok there are a load more settings now... But I'm not really sure what I'm looking at to be honest. The settings from the sipura (which work fine for our carrier btw) don't seem to translate to anything on the PSTN settings page, so I'm a bit lost. And there also seems to be lots of settings on the sipura vs what's on the cs410. Are these important? I'm definitely no PSTN expert, so I suspect I may need a bit of help getting this running...

What's the latest build? Most recent I can see on the website is 2933? I've grabbed 2987 now though, I'll have a little play and let you know how I get on.

Aha, I do indeed have a file in the recordings folder, and it contains the conference. But clicking the speaker still just gives me a blank page?

Tried setting up a conference with the recording enabled, and I get a little speaker icon next to the conference once I've finished, but when I click on it I just get a blank page? Is the recording maybe not available until after the end time? Do you need the call recording license to record conferences, or does it work regardless? ta

Guys, I could do with a response to this - I'm happy to try and get it working myself, but have no idea where to start as none of the settings are exposed I need to change as far as I can tell. I've got a customer waiting for the cs410, and I'd like to get it in as soon as possible.

If you're not getting any audio in any direction, it sounds more network/firewall related than anything to do with PBXnSIP. Does redhat enterprise have some fairly tight ipchains rules by default that perhaps need changing, specifically with regards to RTP? If you do an ngrep on the server do you see the RTP?

Just adding a bit more detail if helpful, this is what the log shows when dialling out (I've taken the DTMF digits out as it's my phone number ). From what I can see, as soon as the last DTMF is sent it goes straight to TALKING - shouldn't it go to RINGING or something first? [5] 2008/01/01 00:07:31: PSTN: Country Code set to 64 [5] 2008/01/01 00:07:31: PSTN: Tone Detection set to 0 [3] 2008/01/01 00:07:31: PSTN: Channel 0 going to DIALLING [6] 2008/01/01 00:07:31: Sending RTP for d68e4385@pbx#1344858385 to 1.1.1.2:2050 [7] 2008/01/01 00:07:32: PSTN: DTMF: x [7] 2008/01/01 00:07:32: PSTN: DTMF: x [7] 2008/01/01 00:07:32: PSTN: DTMF: x [7] 2008/01/01 00:07:33: PSTN: DTMF: x [7] 2008/01/01 00:07:33: PSTN: DTMF: x [7] 2008/01/01 00:07:33: PSTN: DTMF: x [7] 2008/01/01 00:07:34: Call d68e4385@pbx#1344858385: Clear last INVITE [7] 2008/01/01 00:07:34: Set packet length to 20 [9] 2008/01/01 00:07:34: Resolve 55: url sip:127.0.0.1:5062 [9] 2008/01/01 00:07:34: Resolve 55: udp 127.0.0.1 5062 [7] 2008/01/01 00:07:34: Determine pass-through mode after receiving response [3] 2008/01/01 00:07:34: PSTN: Channel 0 going to TALKING

Trying out a CS410 in the UK and generally works ok, except it doesn't seem to detect ringing correctly (thinks it's connected) and the same for busy (just connects regardless). I've tried the "enable busy detection" and also disabled the reverse polarity as it seems the UK doesn't use this setting. All the values needed are here : http://www.provu.co.uk/pdf/sipura/sipura_u...al_settings.pdf I assume one of you boffins can translate into what settings I need to put into PBXnSIP? Ta!

Hi All, We have a customer where after hours, we redirect their calls to an external number. This is done as a hunt group with a night service flag, which when set, redirects to another hunt group, with it's final stage set to the external number. This is fine, except however the PBX uses the incoming number as the "from", and our SIP gateways just turn round and say no, as the car appears that it's not coming from one of our valid users. In olden PBXnSIP days, there was the explicit identity which I could set, but that's now been replaced by Trunk DID, and from what I can see on the wiki, the PBX's rules will always use the incoming number as the from. Is there any way I can override this and set it explicity? Thanks

Both identical, at exactly the same time, with exactly the same content : Call: 07624xxxxxx You missed the following call: From: 07624xxxxxx (click 07624xxxxxx to call back) To: Kristan McDonald (1189) Time: 2008 07 23 10:26:21 This email was sent because your account settings have "send missed call" turned on. Please note that some Email clients may suppress the click-to-dial link. Do not reply to this Email. It was sent automatically.

Well I deleted my extension and re-created it... And now it works fine! Very strange... One other oddity - I always seem to get two emails for missed calls, not one. It's done this for as long as I can remember but I'm not sure why?

Hmm, that's very strange.. especially as normal inbound calls aren't affected, it's only calls to VM's - and asterisk wouldn't know any different? The re-invite is allowed so the PBX can pass through T.38. I'll have a deeper dig...

Yep, sent into support@pbxnsip.com on friday at 16:58 BST. I can resend if you want?

Did you get anywhere with this? We're still seeing the issue here...

SDP looks fine - setup is Vega ISDN->asterisk->PBXnSIP. It's been working fine until (as far as I can tell) we installed the v3 build, though that could just be co-incidence. Also all normal incoming calls are fine, it's just when it tries to send it to VM. I can PM you the pcap if you want? It's only 160k

A strange one (aren't all mine? ) When calling extension from an external number, 9 times out of 10 instead of the caller getting my greeting and going to my VM, they just get silence and can't leave a message. 10th time, it works! The wireshark trace shows the PBX isn't sending any audio after the initial connection, but is receiving it fine - any ideas?? PBX is running 3.0.0.2976 Ta!

Solved

Hi All, I've been fighting with a Polycom 650 this today with no joy. The video here shows what's happening : http://www.flickr.com/photos/kristanm/2657889771 Basically those are speed dial entries assigned to buttons on the left via the phonebook, and when a call is placed... they all disappear! Some of the buttons do still dial, but the one normally used as a 2nd line key seems to revert back to that behaviour. My contacts are Polycom are a bit stumped and are suggesting it might be a PBXnSIP interoperability issue, but I can't really see how? As a side note, I have an expansion panel on this phone, and if I use the left hand buttons as monitoring the status of other extensions and force the speed dials onto the expansion panel, they work fine and persist through calls. Very strange. Polycoms are on the latest firmware, PBX version 2.1.10.2474. I've got full pcap files, polycom boot log and app logs and config files and will make them available shortly. Anyone seen this before or have any ideas what could cause this behaviour?? Thanks, Kristan

Had a VERY strange problem with a cs410 which may affect windows/linux installs also, but I thought I would share to perhaps save people some of the pain I've gone through over the past few days (bear with me, it's long!). We needed to put some wireless phones on a building site, so got polycom 8002's - quite rugged and wireless. Put a 3com 7760 wireless access point in with a 12db antenna and hooked it all up to a cs410 through a switch. Phones worked, registered etc. but after 1-2 seconds the audio would die, and often the access point would seem to have a fit and would need a reboot. This would happen when using X-lite on my laptop over the wireless, but not when plugged straight into the switch. Further testing revealed audio was going out, but not back. The PBX was sending it, but the AP didn't look like it was broadcasting it. Strange thing was, if I turned off the WMM (wireless Qos apparently) setting on the access point it would all work ok, but the polycoms required this to before they would associate with an AP, so turning it off wasn't an option. I assumed the AP must have been dodgy, so took it home and rigged it up to get the phones to register to our office PBX. Everything worked flawlessly. By now, I was really confused. I'd ruled out the cs410, as that worked fine wired, I'd ruled out the polycoms as I'd had them working fine in the office, and I'd ruled out the access point as it worked at home. So there wasn't much else to try. However... I was struck by a thought that the difference between the cs410 and our PBX is operating system, and I'm fairly certain I didn't do the reg hack to enable dscp on our windows box whereas it's enabled as standard on linux. So this evening I've changed the <tos_rtp> section of the pbx.xml file to 0, and lo and behold, it all starts to work. So it looks like the access point isn't handling the DSCP header correctly or can't allocate enough bandwidth and giving up somewhere along the way. Just setting it to a slightly lower priority (104) also seems to work. So all in all, it seems 3com 7760 access points don't do QoS properly, which is a shame because they're been good in every other respect. Someone with more experience with wireless Qos may be able to shed some light on this and get it working without hacking PBXnSIP, but it's all new to me - I didn't even know there was such a thing up until a few days ago!