Telasip trunk rings on and on, bv doesn't

[djw]

I have a couple of sip trunks on an aah 2.0 box.

Calling outbound on the bv trunk, the outside called party stops ringing almost instantly when I hang up my extension.

On the Telasip trunk, the outside called party continues to ring (forever?) long after my extension has hung up.

Any ideas? The config on the trunks are for all practical purposes identical.

[AlexanderBell]

You can find setups for both providers that work on Nerd Vittles.

[djw]

Thanks for the reply, Ward, but I checked and it's already set-up just like you've outlined in Nerdvittles.

Still, Telasip won't hang up or tear down the connection when I hang up, bv does. Odd.

[edpeluffo]

I am having the exact problem with telasip. I believe it started today. Yesterday was a different problem with touch tones.

[AlexanderBell]

Originally Posted by djw

Thanks for the reply, Ward, but I checked and it's already set-up just like you've outlined in Nerdvittles.

Still, Telasip won't hang up or tear down the connection when I hang up, bv does. Odd.

Your original post said the BV and TelaSIP setups were virtually identical. Above you say you've set them up as outlined on Nerd Vittles. The setups really aren't identical so perhaps you could post your dialplans for the two providers. Something's obviously not quite right.

P.S. You have told TelaSIP that you're using an Asterisk server, haven't you?

[greyhound4334]

Interesting comments. Let me add a couple.

- I also use telasip, and also started experiencing "weirdness" with touchtones a couple of days or so ago. Tore my hair out yesterday trying to figure it out. My problem: using call files to dial out from asterisk, asterisk no longer received DTMF tones from my cell phone. I eventually fixed it by adding

exten => s,n,SIPDtmfMode(inband)

to my dial out context.

- AlexanderBell: what did you mean by "telling telasip about using an asterisk server"? I've "mentioned" it, and had a brief dialog with them about using IAX2 (Gene recommends against it), but is there something else that they need to know about to provision the account correctly? Their procedures are so informal, there's no "official" mechanism to really ask OR tell them anything about your account, except using e-mail.

[AlexanderBell]

Email works. Just say "I plan to use my SIP account with Asterisk." You also can set up dtmfmode=inband in your trunk setup (see below). For some this does indeed seem to make the problems go away. I personally haven't had to use it. You'd need to ask Gene what's different about your account and mine. The following has always done the trick for me, and we use this trunk all day every day:

context=telasip-in
dtmfmode=rfc2833
fromuser=alexbell
host=gw3.telasip.com
insecure=very
secret=mypasswd
type=peer
username=alexbell

You shouldn't need the fromuser line unless you have multiple accounts with TelaSIP as we do.

[greyhound4334]

I used the sipdtmfmode command in my dial plan because DTMF was working fine everywhere else. I just wanted to "tune" it for the outbound call initiated from a call file to my cell phone, where the problem turned up.

What's odd is that this problem started just a few days ago. Prior to that, the outbound calling to the cell phone was working fine. It's possible that it's something I changed, but I searched diligently and couldn't find anything. My ears perked up when I heard someone else mention similar problems starting up with telasip recently.

I don't mind e-mail for exchanging info. But telasip would certainly benefit from a bit more "formality" w.r.t. account setup, etc. They have NO way of querying your status or your call history. They're quite terse, bordering on rude, on e-mail. I understand they're a small shop, and I will say first and foremost: the price, call quality, and availability has been excellent. It's just that their informality can be irksome. Cas in point: as a newbie, I *thought* I wanted IAX for my connection to telasip. I wrote and asked them how I go about setting it up. Gene responded with something terse along the lines of "we don't recommend it". Didn't really answer my question.

I persisted and got something along the lines of "we'll have to provision it" along with some more details on why they didn't recommend it. I now wondered whether, indeed, my outstanding request to switch to IAX had been implicitly cancelled. Convinced by Gene's eventual explanation, and upon further consideration, I wished to cancel my request. E-mail to Gene received another somewhat ambiguous reply. Since this was weeks ago, I stopped worrying about it.

Were this an isolated case, I'd not even mention it. But sadly, I've now had occassion to deal with them on technical matters several times, and it seems that it is more representative of a pattern of communication with them, than of a single incident.

Finally, regarding your suggestion to tell them you're running asterisk. Again, I've "mentioned it to them", but I have absolutely NO CLUE what the results of that telling were. Should my account have been provisioned specifically? I have no way of knowing. Could that be the source of some obscure problems I've been having? Or might it bite me in the backside sometime later on? Your suggestion to inform them sounds born of experience and knowledge, so I'm really curious what they do differently, and how I can verify that it's been done.

Sorry for the long post. Don't take it as being critical of telasip's product, which I'm not. It's only the "loose" communications model that I find irksome.

[datarax]

Telasip had problems with the touch tones or dtmf tones due to a problem with their carrier. It is now cleared up as of today at least in my case. Yes, when make a call from telasip and hang up before the other party answers it does continue to ring. An ethereal trace indicates telasip's server does ack back an ok so his * server is getting the cancel but unfortunately either he isn't then passing it on to his carrier or his carrier is dropping the ball. I use Brekeke's ondo pbx/sipserver with telasip and I had the dtmf problem that went away today and still have the call continue to ring when I hang up which doesn't bother me. I wouldn't have known it if i didn't read it here. The dtmf was mentioned to me a couple days ago by Gene.

There are some real advantages to sip over aix especially for telasip. One of the biggest is bandwidth. By using sip the voice packets(udp)don't have to run thru his server. They can go from you directly to his carrier. Only the sip packets traverse between you and him which is minimal. The sip packets have the call control info like ringing, busy answered and cancel. This allows him to process many more calls and elimnating the extra hops required to pass everything thru him. I do believe he has plenty of bandwidth to handle it but it just makes more sense for the udp packets to go direct to his carrier.

Gene is a smart guy and he realized a couple of years ago that there were some real benefits to * but aix was not one of them for a service provider.

For you, if all your extensions are on the same wan/lan then sip might not make a big difference, but if your extensions are spread out amoung several locations then you want each extension's udp packets to go from the extension location direct to the carrier without having to come back to * . In Ondo we turn rtp relay to off to get this behavior. I don't know what you do in *.

So, yes your dtmf hair pulling could have been avoided by telasip sending out a mass email notification of the problem. All I know is that Gene is doing the utmost best he can. At some point he was sshing into a customers machine to work out problems. He gets volumes of calls from people setting up *. * is an unsupported product because it is freeware and it's unfair that the service provider should be responsible for it because in effect that's what is happening. Sip is Sip. It works the same on a sipura spa-2000 as it does on *. If you have a problem with a sip provider, try to get it to work on an ata. If it works on the ata, then fix * yourself. Those dtmf tones didn't work on a sipura either. I predict you will stay with Gene. The reason is you will never find anyone else that will do more to help you than him. It's because his willingness to help that makes him less available.

[djw]

Originally Posted by datarax

Yes, when make a call from telasip and hang up before the other party answers it does continue to ring. An ethereal trace indicates telasip's server does ack back an ok so his * server is getting the cancel but unfortunately either he isn't then passing it on to his carrier or his carrier is dropping the ball.

Thank you, datarax, for checking on this ring-on (i.e. ignored cancel) with Telasip. I did not think my config was a problem since everthing else works fine with Broadvoice and Telasip except the ring-on forever with Telasip only.

Can anyone else confirm this is an issue (or not) for them?

...Fingers crossed the thread doesn't get hijacked into a Gene discussion...