[asterisk-ss7] SAM over 16 digit´s

Domjan Attila adomjan at tvnet.hu
Wed Jul 1 02:10:47 CDT 2009


Hi, working only incomming SAM
what timer is the problem? I configured those:

mtp3_timer.q707_t1 = 4000 
mtp3_timer.q707_t2 = 30000

mtp3_timer.t1 = 500
mtp3_timer.t2 = 700
mtp3_timer.t3 = 500
mtp3_timer.t4 = 500
mtp3_timer.t5 = 500
mtp3_timer.t6 = 500

mtp3_timer.t10 = 60000
mtp3_timer.t12 = 800
mtp3_timer.t13 = 800 
mtp3_timer.t14 = 2000 
mtp3_timer.t19 = 67000
mtp3_timer.t21 = 63000
mtp3_timer.t22 = 300000   
mtp3_timer.t23 = 300000

isup_timer.t1 = 30000 
isup_timer.t2 = 180000
isup_timer.t5 = 300000
isup_timer.t6 = 90000
isup_timer.t7 = 20000

isup_timer.t8 = 10000

isup_timer.t12 = 15000 
isup_timer.t13 = 300000
isup_timer.t14 = 15000 
isup_timer.t15 = 300000
isup_timer.t16 = 15000 
isup_timer.t17 = 300000
isup_timer.t18 = 15000 
isup_timer.t19 = 300000
isup_timer.t20 = 15000
isup_timer.t21 = 300000

isup_timer.t22 = 15000
isup_timer.t23 = 300000

isup_timer.t27 = 240000

isup_timer.t33 = 12000 
isup_timer.t35 = 15000 
isup_timer.digittimeout = 5000

On Tue, 2009-06-30 at 20:49 -0300, Ing. Juan G. Dominguez wrote:
> Attila:
>  
> I install your recommend lib_ss7 soft for solution my problem with
> timers and the send SAM over 16 digits. Your lib_ss7 working with
> SAM ? What is the variable?
> Regards..
>         ----- Original Message ----- 
>         From: Martin Kuca 
>         To: asterisk-ss7 at lists.digium.com 
>         Sent: Wednesday, April 29, 2009 2:23 PM
>         Subject: Re: [asterisk-ss7] chan-ss7 1.1, dial with music on
>         hold - silence
>         
>         
>         there is a bug in app_dial in asterisk 1.4.24
>         problem solved.
>         
>         
>         On 4/29/2009 5:09 PM, Martin Kuca wrote: 
>         > Hello, 
>         > i'm using chan_ss7 1.1, dahdi 2.1.0.4+2.1.0.2 and asterisk
>         > 1.4.24. 
>         > Incoming call from ss7 side. 
>         > When i'm using param m (music on hold in Dial), there is no
>         > sound, only silence. 
>         > I had chan_ss7 0.9, asterisk 1.2 and zaptel there was no
>         > problem before. 
>         > example: 
>         > exten => 333,1,progress() 
>         > exten => 333,2,noop 
>         > exten => 333,3,dial(SIP/444 at proxy,,m()) 
>         > exten => 333,4,hangup 
>         > 
>         > If i do not use param m, it is ok - i hear ringing. 
>         > 
>         > Somebody can help please? 
>         > 
>         > BR, 
>         > martin 
>         > 
>         > 
>         > 
>         > 
>         > ____________________________________________________________
>         > 
>         > _______________________________________________
>         > --Bandwidth and Colocation Provided by http://www.api-digital.com--
>         > 
>         > asterisk-ss7 mailing list
>         > To UNSUBSCRIBE or update options visit:
>         >    http://lists.digium.com/mailman/listinfo/asterisk-ss7
>         
>         
>         
>         
>         ______________________________________________________________
>         
>         _______________________________________________
>         --Bandwidth and Colocation Provided by
>         http://www.api-digital.com--
>         
>         asterisk-ss7 mailing list
>         To UNSUBSCRIBE or update options visit:
>            http://lists.digium.com/mailman/listinfo/asterisk-ss7
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
> 
> asterisk-ss7 mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-ss7
-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: application/pgp-signature
Size: 197 bytes
Desc: This is a digitally signed message part
Url : http://lists.digium.com/pipermail/asterisk-ss7/attachments/20090701/785cd48e/attachment.pgp 


More information about the asterisk-ss7 mailing list