joeh

Is there an ETA that I can go back with?

We have a customer whose extension setup is to forward to an internal hunt group on 'no answer'. (PBXnSIP v2, latest stable with Snom 360s) They have a strange issue where; - External call comes in - Call is put on hold - Person dials other member of staff - They dont answer - Person presses "X" on the Snom to cancel internal call and retrieve previous call - The caller is no longer on hold and is now being passed around the hunt group Is this expected behaviour? I would expect the internal call to be transferred round while the other leg of the call (on hold) should remain? As a workaround I've asked them to increase the no-answer timeout. Is it recommended to call-forward to a hunt group?

Further to this - I have done further testing using SysInternal's FileMon. Looking at the output, I can see PBXCtrl.exe reading in the various XML files from the adrbook directory, presumably into memory. Calls come in, and do not resolve against this list. I have confirmed the domain index (2) matches the address book entries, but still no joy. Is there any kind of extended debug or something to explain why this isn't working ?

We have had similar feedback from our customers. They like the fact they can see the status (as per Mitels etc), but find the fact they can see 'who' someone is calling, is a little off-putting.

I believe the Mitels have a function whereby you can specify a DHCP option which tells the phone to tag its frames with VLAN X. I know the 802.1Q VLAN is an option with Snoms, can this also be done using DHCP, or does it require a combination of DHCP and provisioning?

Extensions Is the 10 extension limit a 'hard' limit on the system or can existing extension licenses be purchased. I understand there's obviously resource constraints, but say we sell a 410 to a small office, who then grows to 11 or maybe 12 extensions - is the only option for them to upgrade to the CS425? FXO Ports I know when I've tried early versions of the Asterisk analog cards, they didn't work well in the UK - due to differences with impedence and callerID. We generally use external ISDN gateways for PSTN Connectivity, but it would be good to use the FXO ports if need be. Do you know if the hardware\libraries etc support the UK telephone network? (If they don't, is this planned?)

Same domains, no IP conflicts. I 'think' (and this has happened before) - it's conflicting call forwarding on both the System and the Snom phones. e.g. System has redirect after 10s to extension 100, while phone says goto extension 200 after 10s. I'm not sure - it baffled me... Logging is on full, so next time it happens I should have a SIP trace.

We have a customer who has reported a problem with their system. When a call comes into a central huntgroup, a member of staff answers the call "Hello Company XX....", the caller then replies "Hello.." but is cut short, and the caller is then immediately connected to someone else - who answers "Hello Company X". They claim they could reproduce this behavior. The two members of staff who answer the call (and answer it again) are members of the same hunt group, and have individual extensions. I have enabled full logging on the PBX and will try and get them to reproduce these problems next week, backed up by logfiles. Any ideas? It's like some kind of race condition or something? (Its V2 btw)

Could you not emulate the SMTP server\agent model whereby a seperate process or thread polls a directory for queued messages. That way, the PBXnSIP process simply drops an EML or whatever in a spool directory - while a seperate process sends them (you could use the native IIS SMTP server on Windows) - that way, if there's files in the directory, they're not being sent.

ah right - I interpreted this as some kind of software loopback, whereby you can tell Windows to take its 'Input' (or recording as its known) source from the WAV output, thus emulating a the cable that you suggest. No worries, I'll do it using the cable instead, as you suggested.

Hi Bill I had the exact same use case a while back, but couldn't figure out a way of doing it. Some key systems call it "DDI Tagging". Joe

How? Am I missing something, or is this MP3 Player\PC\Sound-card dependant?

There are a couple of types of VPN, generally IPSec or PPTP. You could also count the various options and derivatives (L2TP, Tunnel, Transport) but these shouldn't concern you. You want a vanilla IPSec LAN-LAN (routed) VPN using IKE Pre-shared keys; which most devices support (including the RV042) Some general things to watch out for; - If you are using IPSec make sure the two sites are on different IP Subnets otherwise you'll have problems - Make sure each PBX has a route to the remote site, I'm guessing these routers\firewalls will be the default gateways so this shouldn't be an issue - Go for 'Routed' and aviod any NAT options, otherwise you will have the same problems as before... Asides from that - you should be up and running without any real trouble, whether your VPN is IPSec or PPTP. Use LAN -> LAN IPSec and it should be fine.

Perhaps the agent groups base active membership on the registrations. Can you not use a static registration, e.g. "sip:0011111838383@mygateway.local" - that way the PBX sees a registration and forwards the call. The only thing I've been unsure of using this method is, if the gateway is a SIP Registration trunk, how to tell the registration to pass the appropriate credentials to the trunk.

stun.xten.com I think should work ok. I think its because the Snom 300 initiates the call and opens up the necessary pin-holes in the router/firewall - so symmetric RTP is working. When the remote office initiates the call, the sessions dont exist so are dropped. It's very hard to say though - it varies with different models of router and UA.

Without a SIP trace/ethereal dump it's very hard to see what's going on. Can you maybe run Ethereal on the PBX and on the softphone PC, configure them both to capture and initiate the (failing) call. The one way audio sounds like the softphone is replying to the INVITE saying "send you audio here", the PBX then sends the audio to a) The wrong IP The incorrect port or c) The router has restrictive NAT policies. STUN should discover the type of NAT for you.

Is that without any user interaction? e.g. System plays the message, then immediatelly calls the other extension.

This is from a message I posted to the list - I thought I'd take it to the forums (now i can access them) How about introducing some kind of 'poor mans' auto-attendant. A new kind of extension that basically, plays a message, then calls another extension - similar to the Playback application in Asterisk. That way - you can have arbitary messages in the extension logic, before redirecting calls, while changing hunt groups, prior to mailbox messages etc. Something like;

Yep that would be great, not that there's anything wrong with American accents

We're based in the UK and the 'Pound' key confuses the hell of out of people, because it's known as the 'Hash' key. We often find people who come across the prompt pause for a number of seconds looking for the ? key - then may hit the # by chance. Is there a possibility of having 'UK English' prompts with the one subtle change being changing the Pound to a Hash?

We have a customer who wishes to answer a call and divert it to a given destination. They want the ability to view the CDR logs and cross-reference the forwarded leg of the call to a particular account code. Rather than doing a custom system built on Asterisk I was wondering if it could be done with PBXnSIP. One idea I considered was using a prefix code and some dial-plan manipulation. e.g. The number she wants to call is 0111 111111 and wants it billed to client 894. She would answer the call and divert it to **894*0111111111 - the dial plan would strip the **894* and the call would go out as normal. My question is, would the CDR log the original URI or the amended URI after the dial-plan manipulation\normalization. If this isn't possible - is there another way of doing this, or is their account-code like functionality. Suggestions appreciated.

Quick question - whats the easiest way of allowing external access to voicemail (via the PSTN network). With a range of DDI numbers, I'm sure one can be routed to an auto-attendant which in turn can prompt someone for their mailbox extension, then the Pin? I know with Asterisk there is a general menu (please enter your extension, please enter your pin?) is there anything out of the box like this, or will it be a matter of doing our own prompts to enable such a feature?

I've done tests using G.711U with different combinations of jitter and echo variables - with the same results. Faxes to national fax numbers work ok 50-75% of the time, international ones fail most, if not all of the time. Plugging the same fax into a vanilla analogue line works fine. This is when both VoIP endpoints are on a uncongested 100Mbps LAN (the ATA and PSTN Gateway). Searching through Google suggests that faxing using G.711 may work sometimes, but isn't ideal - even on a LAN. My understanding was that T.38 may improve things. That said, the problems I am experiencing could be due to the interaction between the ATA and the analogue fax (I have tried different ATA vendors).

Would a T.38 ATA (SPA2100) and T.38 gateway (Vegastream) with pass-through produce better results (this is all over a LAN) ? (Has anyone done this?)

We use non-T.38 ISDN gateways with our PBXnSIP deployments. We have a customer who has a fax number slap bang in the middle of a DDI range that is (now) routed to their PBXnSIP. I attached a Linksys ATA to that extension, fixed G.711u, disabled echo cancellation\jitter buffers etc. Faxes from 'good' lines receive and send ok, faxes from international numbers or poor quality lines tend to fail. Their old PBX basically had what I guess is a cut down TA - which they then plug their analogue fax into. They then route a DDI number to this interface. My question is - what is the best way of supporting faxes on PBXnSIP installations? Is it best to avoid it entirely, stick with T.38 Gateways and ATAs or suggest a seperate analogue line entirely? I would be interested to here what people use in the form of gateways and ATAs to support faxes.