asterisk users - Jun 2006 - Zaptel dialing too fast?

I have a situation when I dial out my Zaptel I am getting a recording that I
need to add a 1 or a 0 and the area code with this number. I have tried
appending this and the number going out the zap is 1NXXNXXXXXX so it is
going out with 1 and the area code. Someone has suggested that maybe the
zaptel is dialing too fast. My question is how can I add a pause before
dialing to test this out. I am using freePBX 2.1.0, is there a way to do
this in there or will this be a manual hack.

 

 

Here is the tail of the full log when making a call:

 

Jun 15 19:33:51 VERBOSE[25301] logger.c:     -- Executing
Dial("SIP/103-5595", "ZAP/g0/1NXXNXXXXXX |120|r") in new
stack

Jun 15 19:33:51 DEBUG[25301] chan_zap.c: Dialing '1NXXNXXXXXX '

Jun 15 19:33:51 DEBUG[25301] chan_zap.c: Deferring dialing...

Jun 15 19:33:51 DEBUG[2248] channel.c: Avoiding initial deadlock for
'Zap/1-1'

Jun 15 19:33:51 VERBOSE[25301] logger.c:     -- Called g0/1NXXNXXXXXX

Jun 15 19:33:52 DEBUG[25301] chan_zap.c: Exception on 11, channel 1

Jun 15 19:33:52 DEBUG[25301] chan_zap.c: Got event Hook Transition
Complete(12) on channel 1 (index 0)

Jun 15 19:33:54 DEBUG[2282] chan_sip.c: Stopping retransmission on
'632c53f81b496147556ba1f05f0988e5@ xxx.xxx.xxx.xxx' of Request 102:
Match
Found 

Jun 15 19:33:54 DEBUG[25301] chan_zap.c: Exception on 11, channel 1

Jun 15 19:33:54 DEBUG[25301] chan_zap.c: Got event Dial Complete(9) on
channel 1 (index 0)

Jun 15 19:33:54 DEBUG[25301] chan_zap.c: Enabled echo cancellation on
channel 1

Jun 15 19:33:54 DEBUG[25301] chan_zap.c: Engaged echo training on channel 1

Jun 15 19:33:56 DEBUG[25301] chan_zap.c: Exception on 11, channel 1

Jun 15 19:33:56 DEBUG[25301] chan_zap.c: Got event Dial Complete(9) on
channel 1 (index 0)

Jun 15 19:33:56 DEBUG[25301] chan_zap.c: Echo cancellation already on

Jun 15 19:33:56 VERBOSE[25301] logger.c:     -- Zap/1-1 answered
SIP/103-5595

Jun 15 19:33:56 DEBUG[2282] chan_sip.c: Stopping retransmission on
'02730A97-85A3-4FD3-B6EC
<mailto:02730A97-85A3-4FD3-B6EC-E20279757D61@xxx.xxx.xxx.xxx>
-E20279757D61@xxx.xxx.xxx.xxx' of Response 5200: Match Found

Jun 15 19:33:58 DEBUG[2282] chan_sip.c: Auto destroying call '
<mailto:dff8be2d-6f08052b-a937e392@xxx.xxx.xxx.xxx>
dff8be2d-6f08052b-a937e392@xxx.xxx.xxx.xxx'

Jun 15 19:34:11 NOTICE[25301] rtp.c: Comfort noise support incomplete in
Asterisk (RFC 3389). Please turn off on client if possible. Client IP:
xxx.xxx.xxx.xxx

Jun 15 19:34:16 DEBUG[2282] chan_sip.c: Stopping retransmission on
'079943eb0215cd9a5d9aec6c4fd96dfb@ xxx.xxx.xxx.xxx' of Request 102:
Match
Found

Jun 15 19:34:16 DEBUG[2300] chan_iax2.c: Peer lastms 71, historicms 71,
maxms 2000

Jun 15 19:34:16 DEBUG[2282] chan_sip.c: Stopping retransmission on
'3e4904875c7aaa750cacd89a7d94e891@ xxx.xxx.xxx.xxx' of Request 102:
Match
Found

Jun 15 19:34:17 DEBUG[2282] chan_sip.c: Stopping retransmission on
'6bb8c02a2549ac2104f44f9145e5fba9@ xxx.xxx.xxx.xxx' of Request 102:
Match
Found

Jun 15 19:34:18 DEBUG[2282] chan_sip.c: Stopping retransmission on
'5069f5f5785e9c6770ff6f815117646d@ xxx.xxx.xxx.xxx' of Request 102:
Match
Found

Jun 15 19:34:18 DEBUG[2282] chan_sip.c: Stopping retransmission on
'1ba2bc38733dfd693cde414e200a9546@ xxx.xxx.xxx.xxx' of Request 102:
Match
Found

Jun 15 19:34:18 DEBUG[2282] chan_sip.c: Stopping retransmission on '
<mailto:0ccc55236435810960f805fd7124feb3@xxx.xxx.xxx.xxx>
0ccc55236435810960f805fd7124feb3@xxx.xxx.xxx.xxx' of Request 102: Match
Found

Jun 15 19:34:23 DEBUG[25301] channel.c: Didn't get a frame from channel:
SIP/103-5595

Jun 15 19:34:23 DEBUG[25301] channel.c: Bridge stops bridging channels
SIP/103-5595 and Zap/1-1

Jun 15 19:34:23 DEBUG[25301] chan_zap.c: Hangup: channel: 1 index = 0,
normal = 11, callwait = -1, thirdcall = -1

Jun 15 19:34:23 DEBUG[25301] chan_zap.c: disabled echo cancellation on
channel 1

Jun 15 19:34:23 DEBUG[25301] chan_zap.c: Set option TDD MODE, value: OFF(0)
on Zap/1-1

Jun 15 19:34:23 DEBUG[25301] chan_zap.c: Updated conferencing on 1, with 0
conference users

Jun 15 19:34:23 VERBOSE[25301] logger.c:     -- Hungup 'Zap/1-1'

Jun 15 19:34:23 DEBUG[25301] app_dial.c: Exiting with DIALSTATUS=ANSWER.

Thanks,

 

Curt

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Curt Shaffer wrote:
>
> going out with 1 and the area code. Someone has suggested that maybe 
> the zaptel is dialing too fast. My question is how can I add a pause 
> before dialing to test this out. I am using freePBX 2.1.0, is there a 
> way to do this in
>
>   
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Dial

Doug


-- 
 
Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety."

Curt Shaffer wrote:
> I have a situation when I dial out my Zaptel I am getting a recording 
> that I need to add a 1 or a 0 and the area code with this number. I have 
> tried appending this and the number going out the zap is 1NXXNXXXXXX so 
> it is going out with 1 and the area code. Someone has suggested that 
> maybe the zaptel is dialing too fast. My question is how can I add a 
> pause before dialing to test this out. I am using freePBX 2.1.0, is 
> there a way to do this in there or will this be a manual hack.
I'm not a Freepbx user, but your description of the problem sounds like 
you need to add a "w" to the dial command. Its a fairly common problem
with some central office switches in their equipment is not ready to 
accept dtmf tones as quick as asterisk begins sending them.

Your Dial statement should look something like this:
  exten => _9XXXXXXX,1,Dial(Zap/4/w${EXTEN})
where the lower case "w" adds about a 1/4 second delay before dialing.

If that doesn't work, add more w's (like /www${EXTEN}