Koozali.org: home of the SME Server
Obsolete Releases => SME VoIP (Asterisk, SAIL etc) => Topic started by: PWDasterisk on October 08, 2008, 05:14:26 PM
-
Version: sail-2.2.1-664
Zaptel Hardware: TDM13B
1) Previous post mentions this subject:
Routes - Pull down - no Zap Group for FXO trunks
"This is kind of a bug/documentation (don't know which) error. The AGI will always use the zap group on an alias outbound dial, irrespective of which zap line you actually choose."
> Has the issue to have Zap Group for FXO trunks to be explicitly listed in the pull down been addressed yet?
2) Previous zaptel releases disabled the ZAP_CHECK_HOOKSTATE by default due to an improper detection state upon initialization - this meant that a dead FXO trunk could be selected for outbound group routing resulting in an incomplete call. Enabling this in the source code and then recompiling created another issue which Digium has now addressed: http://bugs.digium.com/view.php?id=8290
> Is there some way to install the current zaptel release without breaking the RPM? This would involve uncommenting the line /* #define ZAP_CHECK_HOOKSTATE */ in the chan_zap.c source code and then recompiling.
In a production environment hookstate detection to flag and bypass dead trunks in outbound dialing is standard. I assume that since SARC/SAIL is in commercial use this problem has been discussed. Has anybody addressed this issue yet?
-
Good questions...
Routes - Pull down - no Zap Group for FXO trunks
Pull down was always there, it just didn't appear when you created the route so you had to re-edit the new route to reveal it. Anyhow, it was fixed sometime in the mid 620 releases if I remember correctly. It's fine now.
Also, the second refrence you give refers to aliases, not routes. That also was addressed some time back when the Alias dialer inherited the Route mechanism.
ZAP_CHECK_HOOKSTATE
As you know, we get our rpms from ATrpms so we are dependent upon them for release levels. However, SAIL doesn't care how you installed Asterisk. So, you can patch the ATrpms SRPMs and rebuild them if you wish, or you can hand-crank your own Asterisk/zaptel compiles (you'll need to install a dev kit onto SME, but that's not hard). Either way is fine as far as we're concerned but it may cause you an extra maintenance overhead up the road.
In a production environment hookstate detection to flag and bypass dead trunks in outbound dialing is standard. I assume that since SARC/SAIL is in commercial use this problem has been discussed. Has anybody addressed this issue yet?
Yo might be surprised but, no, we've never discussed it. 80% of our commercial sales are to E/1 (ISDN30e) users and most of the remainder are either ISDN2e or VoIP pureplay. We avoid analogue circuits like the plague, because they are rare here anyway, the sound quality is always poor (compared to ISDN) and you can spend a small lifetime on a customer site balancing the different volume levels and impedances on an AUX group (the UK term for what I think is referred to as a "Rotary" in the United States).
Kind Regards
S