Topic: Router or SARK for DHCP?

[del]

Hi Dave,

What I was going to do was create a route and choose line 1 for the Primary Path, line 2 for the Secondary Path and line three for the Tertiary Path, that way it should (I believe) use line 1 if available then line 2 if line 1 is in use and line 3 if line 1 and line 2 are in use. What do you think?

Del

AFAIK you can set up three identical dial plans for the individual ports instead a single plan for the DAHDI group.

[David Harper]

Sounds exactly right. It's been a while since I did one of these so please excuse my latenight semi-ignorance...

[del]

Sounds exactly right. It's been a while since I did one of these so please excuse my latenight semi-ignorance...

I know the feeling  My problem is I don't do this often enough, mainly due to the fact that SME/SARK just runs and therefore I don't mess unless I have to  This is how I have my current server set up, except I am using VoIP lines at the moment, this will be my first attempt at analogue lines and there as been so many changes since my last install (Asterisk 1.2 and sail-2.2.1-617) Thanks for your input.

Del

[del]

Hi Dave,

What I was going to do was create a route and choose line 1 for the Primary Path, line 2 for the Secondary Path and line three for the Tertiary Path, that way it should (I believe) use line 1 if available then line 2 if line 1 is in use and line 3 if line 1 and line 2 are in use. What do you think?

Del

OK, I've 3 trunks into my TDM404EF card, added 4 Snom 300 handsets and 1 X-lite ext. to SAIL and all internal calls work OK, I can dial in to all 3 trunks but I can't make outgoing calls on the trunks, it rings a couple of times and then AT&T says that I need to put a 0 or a 1 plus the area code to dial long distance, even if it's a local call. I have set up a route as follows:

Active Yes

Prmary Path DAHD2-1

Secondary Path DAHDI3-1

Tertiary Path DAHDI1-1

Quartenary Path None

Route Dial Plans _011X. _1NXXXXXXXXX _NXXXXXXXXX _NXXXXXX

Altwernate (Blank)

Authorization required? No

Description all_calls

Cluster default

I went to the Asterisk CLi and dialed out and it actually worked, then I tried again and it doesn't work anymore

Call that worked:

Verbosity was 0 and is now 3
    -- Executing [13212351867@internal:1] AGI("SIP/5003-086c9d18", "selintra|OutCluster|13212351867") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
    -- AGI Script selintra completed, returning 0
    -- Executing [13212351867@default:1] AGI("SIP/5003-086c9d18", "selintra|OutRoute|all_calls") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
    -- AGI Script Executing Application: (Dial) Options: (DAHDI/2/13212351867||)
    -- Called 2/13212351867
    -- DAHDI/2-1 answered SIP/5003-086c9d18
    -- Executing [h@default:1] Hangup("SIP/5003-086c9d18", "") in new stack
  == Spawn h extension (default, h, 1) exited non-zero on 'SIP/5003-086c9d18'
    -- Hungup 'DAHDI/2-1'
  == Spawn extension (default, 13212351867, 1) exited non-zero on 'SIP/5003-086c                                                                             9d18'

Call that failed

Verbosity is at least 3
    -- Executing [13212351867@internal:1] AGI("SIP/5003-086d0e10", "selintra|OutCluster|13212351867") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
    -- AGI Script selintra completed, returning 0
    -- Executing [13212351867@default:1] AGI("SIP/5003-086d0e10", "selintra|OutRoute|all_calls") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
    -- AGI Script Executing Application: (Dial) Options: (DAHDI/2/13212351867||)
    -- Called 2/13212351867
    -- DAHDI/2-1 answered SIP/5003-086d0e10
    -- Executing [h@default:1] Hangup("SIP/5003-086d0e10", "") in new stack
  == Spawn h extension (default, h, 1) exited non-zero on 'SIP/5003-086d0e10'
    -- Hungup 'DAHDI/2-1'
  == Spawn extension (default, 13212351867, 1) exited non-zero on 'SIP/5003-086d0e10'

Can anyone see anything wrong with it?

Thanks,
Del

[del]

I have been told by the Digium dealer that I need to add www to the dialing prefix to ad a short pause but I can't find where to add it, they use FreePBX and I can see where it goes there so can any one point me in the right direction?

Thanks,
Del

[SARK devs]

And you want a delay because....

?

S

[del]

Apparently the extensions are dialing before AT&T's line is available and therefore it's missing the first digit dialed