Help with 2Talk extension dialling

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52 posts

Master Geek

Reply # 744137 13-Jan-2013 17:50
That seems like serious overkill, a base station for each handset/extension. I have found that the Gigaset handsets have a built-in 'internal dial' feature, that allows each handset to phone the other handsets that are tethered to the same base station; it's still not working to have them internally dial handsets on the other base station, however. And it's too late to change from Siemens to snom, I'm afraid!

chevrolux

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Reply # 744138 13-Jan-2013 18:00
Yea I have to disagree on that. Like I mentioned earlier I like to keep it to 3 or 4 handsets to one base as you are limited with only two active SIP calls. The internal call routing allows one registration to be used exclusively by one handset so it isn't really an issue having one base. In saying all that though, for more than maybe 10 handsets we spec Verse gear. Really really good system. @hads - you might know this one... even when dialling the single digit extension it comes back with 'busy'. I remember on the older siemens phones there was an option to disallow internal calls. Are you aware of any settings on the later models?

jlg84

52 posts

Master Geek

Reply # 744371 14-Jan-2013 09:33
I am making slow progress with this situation. I have now got it so that one of the handsets on my C610IP base station (coincidentally, it is the handset that came with that base) can extension dial to one of the A510H handsets that is tethered to my A510IP base station. However, the A510H handset cannot extension dial back to that handset (it rings on the outgoing handset, but not on the incoming one). I have checked and re-checked that the C610 handset is set to the correct numbers, has the extension entered correctly, etc and all seems to be right. Furthermore, I have noticed that when I try to dial from the other handsets on the A510IP base, I often get "no connection" as the error message on the handset when extension dialing other handsets in the house.

sbiddle

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Reply # 744376 14-Jan-2013 09:42
What type of router are you using? If it has a SIP ALG capability do you have this disabled? *Need help configuring your Linksys ATA or IP Phones for New Zealand? Check my blog post

hads

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Reply # 744377 14-Jan-2013 09:55
The Gigaset base units are set up for six handsets and SIP registrations and there's no issue using all of those if you want to. The "extension" calls aren't any different to other calls, the only difference is that because of the way they are routed at 2Talk they will come back onto your network. It should show up the same as calling one of your DID numbers directly. I'd guess at the router possibly interfering somehow which is what Steve is getting at in the post above. nicegear.co.nz - VoIP, DVB and other Linux compatible hardware

jlg84

52 posts

Master Geek

Reply # 744388 14-Jan-2013 10:12
Thanks, Hads (you guys are the ones I have been buying my equipment from, so thanks for that!). We have a rather complicated set up because of the size of our facility. We don't have a fancy router--all we have is a Netgear wired modem, two Digitus ethernet hubs, and a new ethernet switch on the way from Nicegear hopefully today. Is that perhaps the cause of this trouble? What should we be using?

chevrolux

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Reply # 744395 14-Jan-2013 10:31
@hads - I realise the extension dialing via 2Talk is just done through the SIP registration but isn't there also the ability to make calls internally by just dialing the handset number. ie Office (handset 1) should be able to ring Kitchen (handset 2) by just dialing 2. I know that the older ones you could turn off internal dialing but not sure if I have seen the setting in the VoIP capable handsets. Originally the OP had configured the registrations using the 'One Number' feature in 2Talk so the handsests didn't actually have unique registrations. But when you look at all the registrations now they are all active and yet when you ring the 2Talk extension it comes back unavailable. Seems really weird. The OP is going to try out an Asterisk install I think so maybe the situation will be improved then.

jlg84

52 posts

Master Geek

Reply # 744399 14-Jan-2013 10:42
@chevrolux After numerous tries installing the Asterisk system on an old laptop (purchased in China in 2009 and determined to be a counterfeit Sony, or "Phony") I have given up. The installation was going alright until it told me that it could not find the CentOS install files on the CDROM even though I copied the .iso file intact onto the disk. Tried it three times, each time getting the same error, and finally gave up.

ajobbins

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Reply # 744420 14-Jan-2013 11:20
Did you say you are using the 2talk 'One Number' feature? This will probably cause you issues because I don't think you can assign a unique extension in 2talk for the other registrations to that number. If you can set up each handset with it's own SIP reg, I would use free 028 numbers from 2talk for each of the handsets that you want to ring at the same time, and use Follow Me to make them ring. Then you can set a unique extension number in 2talk for each registration (handset). Other option would be to get a Raspberry Pi and run Asterisk on there. That will give you greater call control and flexibility for next to no cost. Twitter: ajobbins

chevrolux

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Reply # 744438 14-Jan-2013 11:36
He was using One Number but not anymore. I believe that was the original issue. A Raspberry Pi would be a great idea. I run one at home but have found it can be a bit hit and miss with the amount of simultaneous calls you can make. Like sometimes it will do 3 or 4 calls but then other times even the second call will go a bit unstable with bad speech.

ajobbins

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Reply # 744444 14-Jan-2013 11:42
chevrolux: He was using One Number but not anymore. I believe that was the original issue. A Raspberry Pi would be a great idea. I run one at home but have found it can be a bit hit and miss with the amount of simultaneous calls you can make. Like sometimes it will do 3 or 4 calls but then other times even the second call will go a bit unstable with bad speech. . Is it doing any codec transcoding? I've tested with 8 concurrent calls in a conference bridge using G711 and the Pi was sitting on about 15% CPU usage. As soon as you added a call that needed transcoding in there, it couldn't cope. Twitter: ajobbins

sbiddle

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Reply # 744466 14-Jan-2013 12:20
jlg84: @chevrolux After numerous tries installing the Asterisk system on an old laptop (purchased in China in 2009 and determined to be a counterfeit Sony, or "Phony") I have given up. The installation was going alright until it told me that it could not find the CentOS install files on the CDROM even though I copied the .iso file intact onto the disk. Tried it three times, each time getting the same error, and finally gave up. Assume you're using an n all in one Asterisk distribution you don't copy the .ISO to a CD, you burn it to a CD to create a disc image. *Need help configuring your Linksys ATA or IP Phones for New Zealand? Check my blog post

kornflake

101 posts

Master Geek

Reply # 744568 14-Jan-2013 14:38
as long as you have a CIR of 80/90k on the connection (per call leg) quality should be no issue (G711)

jlg84

52 posts

Master Geek

Reply # 744570 14-Jan-2013 14:41
I misspoke before; I did burn the ISO to a CD drive, and that was the CD that kept giving the message that it could not find the CentOS installation files.

jlg84

52 posts

Master Geek

Reply # 744576 14-Jan-2013 14:44
Regarding simultaneous calls, I do not envisage that we would ever have more than three calls at once (and that would be extraordinarily rare) so do not let that sort of issue dissuade anyone from making recommendations that would not work in a heavy-call load environment. We're a small lodge, with a total of five guest rooms, and few guests would use our phones anyway.

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