Topic: zaptel-kdml where can i find it for sme 7.4? & ring duration problem

[rrkelly]

where can i find the current zapptel-kdml for sme 7.4

also i have ring duration problem that i can not seem to find the cause of -- no matter what i set the ring time in seconds to in global's or a ring grp (alias) the best i can get is 3 rings then there is a slight bump in the ring tone the phone quits ringing the calling party still hears a ring but the phone itself does not ring 
asterisk cli does not show a hangup when the phone quits ringing.
i'm using an asstra 57i for the operator phone but have tried with 9133i also
thanks
rob

[SARK devs]

asterisk kmdls live at ATrpms.  Installation rules are the same as always...

yum install zaptel-kmdl-`uname -r` --enablerepo=atrpms

As to your second problem; I'm afraid you've got us beaten.  we've never heard of such a thing. I think you'll need to do some more experimentation.  First tho', go back to basics...

When did the problem start
What changed

See if you can remove the problem by regressing back to the way your system was when you didn't have the problem.

Run a SIP trace to ensure that nothing unusual is being sent to the phone.


Kind Regards

S

[Franco]

asterisk kmdls live at ATrpms.  Installation rules are the same as always...

yum install zaptel-kmdl-`uname -r` --enablerepo=atrpms

Hi,
I had to return my kernel to 2.6.9-67.0.7.ELsmp since there's no kmdls for 2.6.9-78.0.8.ELsmp.

[rrkelly]

zaptel-kdml for the sme 7.4  kernel has disappeared from atrpms  not a trace anywhere -- it was there last week gone today.
 
Got a copy of the sark beta; updated to sail 713, and will try to reproduce the ring problem.
We did a sip trace and lots of scrolling not really sure if it is sail, we have an audio codes feeding a grandstream GXW then into sail and on out to asstra's so it could be any of the aforementioned.
The phone quits after 3 rings but asterisk does not drop the call till some time later and the caller still gets the ring signal but not as long as i think it should there is no IVR (they don't want one).
thanks
rob
 

[SARK devs]

we have harvested the .78 kmdl rpms...

http://sarkpbx.com/sail/kmdls

Kind Regards

S

[SARK devs]

we have an audio codes feeding a grandstream GXW then into sail and on out to asstra's so it could be any of the aforementioned.

Good to see you kept it nice and simple  .

The audiocodes is analogue or digital upstream?

Best

S




 

[rrkelly]

 this is what happens when politico's/good old boys  trump common sense. The local phone company sold 20 sip lines the only problem was they did not know what sip line was. i would have
flushed them 3 month ago if there was any way possible. you would not believe the fun we have had to get this "Rube Goldstien" working. thanks for your help and if you would like the grandstream gxw info i would be glad to write it up -- GS has another bug fix coming then they should be reasonable good and not to bad to set up
thanks again
rob   

[SARK devs]

I'm still confused...

how does it go?

SIP -> Asterisk ->GS ->audio codes ->Telco

Yes?

[rrkelly]

that would be it sail/sip to gwx4XXX then fsx audio codes then mgcp to their switch which is more than capable of giving me a plain old sip trunk if
1 they could figure out how to program it  and  2 had someone that actually understood what a sip trunk was 
but they do have a really good, well connected salesman
please also note that vitelity was working from day one and was cheaper than
than the unmentionable phone company. patronage at its finest
rob

[SARK devs]

Just my 2c, but wouldn't it be easier (and a lot cleaner) to do...

sail/sip -> T1 -> Telco


Best

S

[rrkelly]

most definitely yes but only 1 T1 and 4 different locations and intermittent hi useage on the T1 also they can only provide their service at 2 of the 4 locations  They where suppose to provide/program sips in Nov. still waiting, it was suppose to be temporary. The one office with the best bandwidth naturally only has 2 phones and most of the time no one is there. The phone company claims they have MGCP support and i could make them provide a fat pipe

(it is next to the Mayo Clinic Campus which why they have the bandwidth)

and trunk the other offices to it.  -- does that sound reasonable? or should i
ask for something else and still try to trunk to the good bandwidth or wait and hope they get a clue? i open to any suggestions

rob

[SARK devs]

I'm still not 100% sure that I understand.  When I said T1 - I meant T1/TDM (not T1 data), presumably you can have as many T1 TDM circuits as you like into the SAIL box...

So - I guess it's then just a question of putting the SAIL box somewhere where there is good, cheap ADSL/SDSL (or T1) uplift; - perhaps in the next State (or even offshore) if you wanted to. Then you can hand the SIP traffic over to the telco using conventional T1/TDM circuits

In the end, SIP trunks are just IP addresses on the net..     


S