TDM01B & A&H 2.5 outgoing dialing issues

[adroitboy]

I am having a hard time getting my new TDM01B (one FXO port) working with my current setup.

Currently I have A@H 2.5 installed. I did not have the card installed when I first installed A@H. To fix this, I installed the card, ran rebuild_zaptel, rebooted, ran genzaptelconf, and rebooted again. My issue is that I cannot seem to get calls to go out over zap reliably. I dial 9xxx-xxxx which uses an outbound route with the zap channel only. It seems like it starts to work, but then just sits there (I have the asterisk -r info below)

If it works, there is a shorter delay, and then it starts. When it does not work - which is most of the time - it appears that it is going to dial, but never does. After a long delay (15-30 seconds) the Qwest operator (from the POTS system) gives the message:

"were sorry, your call did not go through. Please try your call again later"

Interstingly, when -- Zap/4-1 answered SIP/201-576c come up, the phone that I am talking on goes from near silence to having some static on it. It's like it picks up the line (hence the static), but not enough to get dial tone?

I actually just listened in. What happens is that it dials xxx-966 and then at least a 3-4 second pause and then it dials the last (seventh) digit. It also does the same behavior when I dial xxx-xxx-966 <pause4> 6. However this is too long of a pause for the telco and it gives me the "were sorry, your call did not go through. Please try your call again later" message.

Why the delay on this last digit?

In AMP
=========
ZAP Identifier (Trunk name) is set to g0 (that's a zero)

Flash Operator Panel
==================
ID's the line as Zap4. It does show off the hook (red)when it is trying to dial.

I don't appear to have any IRQ conflicts

[root@asterisk1 ~]# cat /proc/interrupts
           CPU0      
  0:    4601013          XT-PIC  timer
  1:          8          XT-PIC  i8042
  2:          0          XT-PIC  cascade
  5:     375172          XT-PIC  eth0
  8:          1          XT-PIC  rtc
  9:          0          XT-PIC  Ensoniq AudioPCI
10:    4515234          XT-PIC  wctdm
11:          0          XT-PIC  uhci_hcd
14:      21183          XT-PIC  ide0
15:      40366          XT-PIC  ide1
NMI:          0
ERR:          0

asterisk CLI output

    -- Executing Macro("SIP/401-5347", "dialout-trunk|1|633xxxx|") in new stack
    -- Executing GotoIf("SIP/401-5347", "1?3:2)") in new stack
    -- Goto (macro-dialout-trunk,s,3)
    -- Executing Macro("SIP/401-5347", "user-callerid") in new stack
    -- Executing DBget("SIP/401-5347", "AMPUSER=DEVICE/401/user") in new stack
    -- DBget: varname=AMPUSER, family=DEVICE, key=401/user
    -- DBget: set variable AMPUSER to 401
    -- Executing DBget("SIP/401-5347", "AMPUSERCIDNAME=AMPUSER/401/cidname") in new stack
    -- DBget: varname=AMPUSERCIDNAME, family=AMPUSER, key=401/cidname
    -- DBget: set variable AMPUSERCIDNAME to Aaron & Tricia Corded
    -- Executing GotoIf("SIP/401-5347", "0?5") in new stack
    -- Executing SetCallerID("SIP/401-5347", ""Aaron &amp; Tricia Corded" <401>") in new stack
    -- Executing NoOp("SIP/401-5347", "Using CallerID "Aaron &amp; Tricia Corded" <401>") in new stack
    -- Executing Macro("SIP/401-5347", "record-enable|401|OUT") in new stack
    -- Executing GotoIf("SIP/401-5347", "0 > 0?2:4") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing AGI("SIP/401-5347", "recordingcheck|20060306-141123|1141683083.48") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
  recordingcheck|20060306-141123|1141683083.48: Outbound recording not enabled
    -- AGI Script recordingcheck completed, returning 0
    -- Executing NoOp("SIP/401-5347", "No recording needed") in new stack
    -- Executing Macro("SIP/401-5347", "outbound-callerid|1") in new stack
    -- Executing DBget("SIP/401-5347", "USEROUTCID=AMPUSER/401/outboundcid") in new stack
    -- DBget: varname=USEROUTCID, family=AMPUSER, key=401/outboundcid
    -- DBget: set variable USEROUTCID to TriciaAaron <360633xxxx>
    -- Executing GotoIf("SIP/401-5347", "1?4") in new stack
    -- Goto (macro-outbound-callerid,s,4)
    -- Executing GotoIf("SIP/401-5347", "0?6") in new stack
    -- Executing SetCallerID("SIP/401-5347", "TriciaAaron <360633xxxx>") in new stack
    -- Executing NoOp("SIP/401-5347", "CallerID set to "TriciaAaron" <360633xxxx>") in new stack
    -- Executing SetGroup("SIP/401-5347", "OUT_1") in new stack
    -- Executing CheckGroup("SIP/401-5347", "1") in new stack
    -- Executing SetVar("SIP/401-5347", "DIAL_NUMBER=633xxxx") in new stack
    -- Executing SetVar("SIP/401-5347", "DIAL_TRUNK=1") in new stack
    -- Executing AGI("SIP/401-5347", "fixlocalprefix") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
    -- AGI Script fixlocalprefix completed, returning 0
    -- Executing SetVar("SIP/401-5347", "OUTNUM=633xxxx") in new stack
    -- Executing Cut("SIP/401-5347", "custom=OUT_1|:|1") in new stack
    -- Executing GotoIf("SIP/401-5347", "0?16") in new stack
    -- Executing Dial("SIP/401-5347", "ZAP/g0/633xxxx") in new stack
    -- Called g0/633xxxx
    -- Zap/4-1 answered SIP/401-5347
    -- Hungup 'Zap/4-1'
  == Spawn extension (macro-dialout-trunk, s, 14) exited non-zero on 'SIP/401-5347' in macro 'dialout-trunk'
  == Spawn extension (from-internal, 9633xxxx, 1) exited non-zero on 'SIP/401-5347'
    -- Executing Macro("SIP/401-5347", "hangupcall") in new stack
    -- Executing ResetCDR("SIP/401-5347", "w") in new stack
    -- Executing NoCDR("SIP/401-5347", "") in new stack
    -- Executing Wait("SIP/401-5347", "5") in new stack
  == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/401-5347' in macro 'hangupcall'
  == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/401-5347'
asterisk1*CLI>

My zapata.conf

usecallerid=yes
hidecallerid=no
callwaiting=no
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=800
rxgain=0.0
txgain=0.0
group=0
callgroup=1
pickupgroup=1
immediate=no
                                                                         
;faxdetect=both
faxdetect=incoming
;faxdetect=outgoing
;faxdetect=no

;Include genzaptelconf configs
#include zapata-auto.conf

;Include AMP configs
#include zapata_additional.conf

zapata-auto.conf

; Autogenerated by /usr/local/sbin/genzaptelconf -- do not hand edit
; Zaptel Channels Configurations (zapata.conf)
;
; This is not intended to be a complete zapata.conf. Rather, it is intended
; to be #include-d by /etc/zapata.conf that will include the global settings
;
callerid=asreceived

; Span 1: WCTDM/0 "Wildcard TDM400P REV I Board 1"
; channel 1, WCTDM, inactive.
; channel 2, WCTDM, inactive.
; channel 3, WCTDM, inactive.
signalling=fxs_ks
; Note: this is a trunk. Create a ZAP trunk in AMP for Channel 4
context=from-pstn
group=0
channel => 4


; Span 2: ZTDUMMY/1 "ZTDUMMY/1 1"

[adroitboy]

I actually just listened in. What happens is that it dials xxx-966 and then at least a 3-4 second pause and then it dials the last (seventh) digit. It also does the same behavior when I dial xxx-xxx-966 <pause4> 6. However this is too long of a pause for the telco and it gives me the "were sorry, your call did not go through. Please try your call again later" message.

Why the delay on this last digit?

Sorry to respond to myself, but someone must have an idea!?! I would really love to not have to re-install A@H again! So the magic question is - Why the long delay to dial the last digit on the ZAP channel?

[dswartz]

not sure if this is what bit me, but same deal (dunno about the delay on the last digit), but i went from being able to call reliably, to almost never working. googled around and came up with the theory that asterisk was dialing too soon and ticking off the telco. i changed my dial plan (actually, i think it was the prefix?) to put a 'ww' at the front of all dial numbers going out zap port (this causes a brief delay). problem hasn't recurred...

[adroitboy]

Originally Posted by dswartz

not sure if this is what bit me, but same deal (dunno about the delay on the last digit), but i went from being able to call reliably, to almost never working. googled around and came up with the theory that asterisk was dialing too soon and ticking off the telco. i changed my dial plan (actually, i think it was the prefix?) to put a 'ww' at the front of all dial numbers going out zap port (this causes a brief delay). problem hasn't recurred...

Thanks for this tidbit. I'll try to do this and see what happens. This might be part of the problem as well...who knows. It is really strange thogh when it dials all of the digits and then pauses for a few seconds. I suppose it could be picking up early as well.

[adroitboy]

dswartz,

Thanks for the suggestion. It looks like adding a ww+. to my outgoing dial rules for the zap trunk made it work. So perhaps it was not the delay in the last digit, but rather the card dialing too quickly after picking up the line.

Anyway, it now works. Thanks!