Outgoing call work sometimes

Asterisk Asterisk Support
1

Hi everybody,

I use asterisk and as i am au “noob” here is my version :
pabx*CLI> show version
Asterisk 1.4.21.2~dfsg-3 built by root @ asterisk-cdmaker on a i686 running Linux on 2009-01-29 14:18:02 UTC

I have problem with outgoing calls which sometimes work and sometimes fail (more fail).
When the fail, it seems like the phone is ringing but not.

I don’t know where to start debugging. Here is the diag on outgoing call :

– Executing [06xxxxxxxx@internal:1] Macro(“SIP/631-08797ec0”, “dialdefault|06xxxxxxxx”) in new stack
– Executing [s@macro-dialdefault:1] SetCallerPres(“SIP/631-08797ec0”, “allowed”) in new stack
– Executing [s@macro-dialdefault:2] GotoIf(“SIP/631-08797ec0”, “1?100”) in new stack
– Goto (macro-dialdefault,s,100)
– Executing [s@macro-dialdefault:100] Set(“SIP/631-08797ec0”, “CALLERID(num)=320746630”) in new stack
– Executing [s@macro-dialdefault:101] misdn_check_l2l1(“SIP/631-08797ec0”, “g:extern|2”) in new stack
– Executing [s@macro-dialdefault:102] Dial(“SIP/631-08797ec0”, “mISDN/g:extern/06xxxxxxxx/e256|60|r”) in new stack
P[ 1] channel with stid:0 for one second still in use!
– Called g:extern/06xxxxxxxx/e256
Really destroying SIP dialog ‘50f3e0c40fd395423aec26e17b97a47a@192.168.120.1’ Method: OPTIONS
– mISDN/1-u322 is proceeding passing it to SIP/631-08797ec0
– mISDN/1-u322 is making progress passing it to SIP/631-08797ec0
Really destroying SIP dialog ‘677f615e5aa25d1f52055d6141b69663@192.168.120.1’ Method: OPTIONS
Really destroying SIP dialog ‘22548f590d1d542639847d5c1a9c278a@192.168.120.1’ Method: OPTIONS
Really destroying SIP dialog ‘5ab7c73d1008bd5a4da90ff0574ba23d@192.168.120.1’ Method: OPTIONS
Really destroying SIP dialog ‘28fe6cf453f407e0479bf46f033f67fc@192.168.120.1’ Method: OPTIONS
== Spawn extension (macro-dialdefault, s, 102) exited non-zero on 'SIP/631-08797ec0’
Really destroying SIP dialog ‘00179538-138c005c-56765e59-7673e1ee@192.168.120.35’ Method: ACK

thank you
Regards

2

Not too many folks still using mISDN. You could try swapping to DAHDI instead - but you’d probably also want to look at a newer version of Asterisk…1.4’s been EOL for a while now.

3

Hi,

The fact is I don’t know how to update quickly and how to swap.
do I need to buy new material ?

4

Try adding a “w” to your trunk to delay dialing??

That works good for pots lines…not sure about misdn…