Topic: Asterisk / Sail - Delay before operation extension rings

[jonroberts]

I've got my SAIL server up and running (& very happy I am about it too    - Cheers Jeff aka Selintra).  However, I've got a couple of small issues I'd still like to resolve.

When I call in as an external caller, I can hear the phone ring 3 or 4 times before my Internal Operator extension starts to ring.  This seems to be the case for both incoming analogue and incoming SIP (although its maybe a little quicker for incoming SIP).  Its also the same whether the Operator extension is an IP phone or an analogue phone.

I don't have fax detection enabled on any trunk.  Any idea what's causing this delay & whether I can do anything to affect it?

Also, the analogue phone I'm using as my Internal extension supports Called ID display.  This works when connected directly to the exchange line, but not when operating as an Internal handset.

Its not a massive problem, but can Caller ID display be activated on internal analogue handsets or just on IP phones?

Cheers
Jon

[JonB]

Jon,

You can adjust the ring delay in the global settings.

Jon

[jonroberts]

Jon,

Thanks - I've changed that, but it doesn't seem to make a difference.  I think the Internal Ring Delay in Global Settings is the length of time (in seconds) an Internal Extensions rings before going to voicemail.  (& having just set it to zero & tested it, I'm pretty sure thats the case   )

Having just tested this a little further, I've realised that - even with the analogue phone connected directly to the exchange line (ie No asterisk involved) - the external caller still hears one or two rings before the Internal Phone rings.  So at worst the delay due to Asterisk is only a couple of seconds, so I'm not too worried about it, although it would still be nice to find out what's causing it.  

Jon

[SARK devs]

Hi Guys

We're running 2.1.14-311 on our production switch and it has the "late take" stuff which Sam did for someone on here.  If I dial in using a VoIP number, the operator extension rings at pretty much exactly the same time as the caller hears the first ring (I tested it from my mobile).

Analogue is about 2 seconds slower and that's because SARK is waiting for the caller-id spill which, in the UK, arrives either just before or just after the first ring (can't remember which) on an analogue line.    I have a feeling it arrives much earlier on the Bellcore system but I may be getting confused in my old age.

If you have FAX detect enabled on the TDM line, it is slower still because we have to hang around in the arrivals context (mainmenu) while we try to detect a FAX and this depends upon what you have the faxdetect delay set to.

Kind Regards

jeff@selintra

[SARK devs]

Its not a massive problem, but can Caller ID display be activated on internal analogue handsets or just on IP phones?

Sorry Jon - missed this.  Short answer is "Yes,  it can be done".  Longer answer is that you have to "push" the CLID to the analogue handset.  It isn't hard to do but we've just never bothered because all of our clients have SIP phones so it hasn't been necessary.

Call me if you have a neartime need for it.

Kind Regards

jeff@selintra.com

[del]

Hi Selintra,

Also, the analogue phone I'm using as my Internal extension supports Called ID display. This works when connected directly to the exchange line, but not when operating as an Internal handset.

Thats funny my cheapo handset from Walmart shows caller ID from incoming calls and from internal extensions, shows calling extension number and the user name as set in sail panel.
Could this be because it is connected to a Sipura 2000? By the way I have now figured out how to set up my "unlocked" Vonage PAP2, compared the settings with the SIpura 2000 (which just setup automatically using DHCP   ) so if anyone wants to use an unlocked Vonage PAP2 let me know.  Thanks again for the help.

Del

[SARK devs]

Could this be because it is connected to a Sipura 2000

Yes - the Sipura will do it in the US and maybe also for BT here in the UK (other countries I'm not sure about ).  

Nice to hear you got the PAP2 working Del, e-mail us your settings if you get a moment and we'll look at setting up a device definition for it.

Kind Regards

jeff

[jonroberts]

Jeff - I'm also using 2.1.14-311, so I assume its the same 'late-take' issue.  Is this now a permanent feature or will it be a customisable option in some later release?

If I start to miss the call-id on my Analogue phone, I'll look to getting myself an ATA - its the next thing for me to start playing with anyway ( - sorry, I meant analyse and evaluate   )

[SARK devs]

Hi Jon,

"late take" isn't an issue - it's supposed to be an enhancement   !
IT means that it terminates the call as near to the final extension as possible.  In theory, because of the SARK mechanics, this should minimise unnecessary ring delay (and it does seem to work well on VoIP).  However, analogue is a bit slower.

Hope this helps

Best

jeff@selintra.com

[del]

Hi Selintra,

I will do some screenshots/notes and email them to you. I can't remember what some of the original settings were but I have made notes as to which settings I changed. Basically I installed the Sipura 2000 first, made sure it worked then connected the PAP2 and opened both admin pages and just compared each and every setting I made them exactly the same. The server set the PAP2 login details correctly and I assume a few of others  I think that most of the wrong stuff was left over from Vonage. I will email everything soon.

Regards,
Del

[jonroberts]

"late take" isn't an issue - it's supposed to be an enhancement   !

Whoops! Never meant to imply otherwise    

Its not really a major problem, it just adds a couple of rings to the time the external caller waits before the phone is answered (or maybe it doesn't, it just makes it seem like it does), which probably gives me 1 or 2 rings less to answer before they decide I'm not here & hang up.

I'll just make sure I answer the phone a little quicker in future - not normally a problem except when I forget where I put it down!

[SARK devs]

Jon,

This is inbound analogue delay?  Yes?

I'll ask Sam to have a look to see if he can take anything out of the path.  

Best

Jeff

[Tib]

jonroberts

Your not running a sipura3000 by any chance are you?

I had a similar problem and it was one of the setting inside the sipura that transfered the call to the asterisk box ... after I changed the setting to a lower value all I got was one ring then straight away the asterisk (sail) will take over.

I'll check out the setting I changed ... at wotk atm.

Regards,

Tib

[jonroberts]

This is inbound analogue delay?

Yes, its the TDM11 card.  On testing, if the inbound call is forwarded to an IP Phone, it rings about 1 ring quicker than if its forwarded onto the analogue phone connected via the TDM card.

Inbound calls via Gradwell are better; there's about a full ring before the internal handsets ring, but nothing much.  Again, this delay is slightly longer if the calls are forwarded to the analogue phone (by forwarded to, I mean set the 'Open Inbound Route" to)

Your not running a sipura3000 by any chance are you?

Tib - No, I'm not running a sipura3000.  My IP Phone is a Grandstream GX2000 & the analogue phone is connected to the TDM card - although yesterday I realised that as my Router is a Vigor2600v, it has two inbuilt SIP ports that I can use to act as ATAs to SAIL - just another one of those things that I really should have realised about 4 weeks ago!