similar to: ZAP Digit Timeout

I am have same issue with PRI and overlap dialling is not enabled. Stuart -----Original Message----- From: "Peter Svensson"<psvasterisk@psv.nu> Sent: 01/02/05 16:55:52 To: "Asterisk Users Mailing List - Non-Commercial Discussion"<asterisk-users@lists.digium.com> Subject: Re: [Asterisk-Users] Zap channel occasionally misses dialing thefirst digit

Hi, I've noticed that a small but meaningful quota of calls from my Alcatel PBX to Asterisk are failing. This does not always happen and it is not easily reproducible but on high traffic I do get a large number of cases. Example: Alcatel PBX extension 7085 calls Asterisk PBX extension 6145 over a PRI E1 link. I see this in the Asterisk log: Dec 14 14:10:31 VERBOSE[11378] logger.c: --

Hello, This seems to be a weird one. I'm at work now and will get some more-verbose logs later when I get home if nobody has any ideas about what's happening here. I've got a tdm card with 1 FXO and 1 FXS. Asterisk is in the 1.2.x line, so is zaptel. astlinux to be specific. I can get the versions at home later if it might help. It's running on a silent epia 5000 board

....I THINK. When dialing 1+10 digits, I occasionally get a telco message "You must first dial a 1....". When I look at the console, the number is being sent to the ZAP channel properly. We're talking about a couple of POTS lines on a TDM400P. I'm thinking that it may be starting the dial too early after coming off-hook because I can just redial and have it work (or not)

brett@websmyths.com wrote: > Also - an outside chance - make sure Tip and Ring > are correct. You could be getting ground loops - depends on the noise. > I am having noise and slip errors between my TE110P and a legacy PBX T1 card. Could this be the same symptom? The connection is made using a 15 pin serial on the T1 Card side to RJ48 on the TE110P side. I can't determine what the

I need to select a line from the Zap group channel using the SIP Phone (not FXO and not FXS ports). ignorepat does not work? Also, what is the method to let the second dial tone has another tone frequency? Regards Bilal ---------------- No, ignorepat is for FXS ports (FXS ports use FXO signaling). Also, ignorepat does not apply to SIP phones, because SIP phones provide their own dialtone,

Good day, I am trying to setup an outbound dial plan which will time out if no answer. Using a X100P with the following dial command : exten => 101,3,Dial(Zap/1/3036972357,5) ; try the desk line - fail to step 104 It dials out successfully, but never times out. I have a basic Zapata config : group = 1 context = RedRockWeb language = en signalling = fxs_ks usecallerid = yes hidecallerid

Hello, this is an example extensions.conf. [default] exten => 500,1,Answer exten => 8,1,SetGlobalVar(firstdigit=8) exten => 8,2,Goto(process,s,1) exten => 9,1,SetGlobalVar(firstdigit=9) exten => 9,2,Goto(process,s,1) I call extension 500 and send dtmf digit 9. This is printed to the CLI: -- Executing Answer("Zap/20-1", "") in new stack -- Accepting

I have one POTS line going into a TDM400P. Here in Atlanta, we have 10 digit local dialing. I launch a call "Zap/1/7705551212" and it goes thru just fine. The next time I try it, without any modifications, I get a Bell recording telling me that I must dial the area code and seven digit number when placing a local call. It's like Asterisk may be starting the dial before the line

Hi I am trying to develop a night divert. Caller dials in after hours on Zap and it gets divert to a mobile number via a second Zap. The call bridges but will not hangup the channels when the parties finish. Is there something I am missing or an dial option that I should be using. I am using latest CVS. [night] exten => s,1,Answer exten => s,2,Wait,1 exten =>

I'm using a T100p to interface to a channel bank and from there to analog PSTN lines. Because of my particular setup I have to do post-connect inband DTMF dialing - which takes up to 5 seconds for a 10 digit number, assuming 0.5/sec per digit (ie. using "zap/g1/31|5|D(6045551212)". Even with an 'outside transfer' voice prompt before commencing dialing my users are getting

Hi guys, I am trying to use DISA. The scenario is - I call my home number (where X100P seats) from mobile phone, enter the password, enter international number and get connected via voiptel. It works perfectly when I call extension setup with DISA from X-PRO SIP phone, but when I dial into Zap, It seems that it does not detect DTMF tones. Here is a log and config files Please help

Hi! For demonstration purposes I try to bridge an incoming video call from a 3G mobile handset to another 3G mobile handset using asterisk as "switch". On the incoming call leg I see all expected bearer capabilities (Digital, 64k Transparent, G.7xx 384k video) but on the outgoing call leg the bearer capability G.7xx 384k video get lost and therefore the call is dropped from the mobile

> Watch the output of 'pri debug span 1' on the Asterisk server while > placing the call - bug #4468 (http://bugs.digium.com/view.php?id=4468) > might be relevant. Yes, this is exactly what is happening. Thanks a lot. I am thinking about adding a special case for the IBM 800 number since it is the only one my company is complaining about. Currently I have this in my dialplan:

I'm having problems using the following. [sip] exten => _*4.,1,macro(test,${EXTEN:2},${CALLERIDNUM}) [macro-test] exten => s,1,Answer exten => s,3,Flash exten => s,3,Dial(SIP/${ARG2},30,t) exten => s,4,Dial(SIP/${ARG1},30,t) exten => s,t,Hangup exten => s,i,Hangup exten => s,h,Hangup I know I must be missing something simple, but here is the output from

I have four Polycom 330 phones connected to an asterisk system. There are other VoIP phones connected too. All of the extensions are four digits beginning with 11. From any of the phones, except the Polycom, picking up the handset to call extension 1103 for example works fine. With the Polycom 330, as I press the second 1 of 1103 it stops taking input and gives me an error. I tried

Hey all, Working slowly on getting the myriad number of parts to my fax system plan together, and one of the pieces I want to nail is how to go about, for the outbound context (fax-out) pre-pending a digit onto a number? I.e., for all my testing right now, I've been dialing '91XXXXXXXXXX', as the asterisk server doing faxing junctions into my old Rolm CBX switch, and so I need the

Hi, I'm having a problem with the timeout option when dialing a ZAP channel. The goal is to ring a number for 15 seconds, if no one picks up, go to voicemail. The dial command is: exten => s,1,Dial(ZAP/1/6135551111,15) exten => s,2,VoiceMail(u1) exten => s,102,VoiceMail(b1) The call will continue to ring beyond 15 seconds. What's interesting is that a SIP channels does not have

Hi, sorry for this noop question, but does anybody know how to cut the first digit of a variable? example: 044612345 should be after cut operation: 44612345 My try in the extension.conf: exten => _0[0-9].,2,Cut(mynum=EXTEN,/ ,1) exten => _0[0-9].,3,Dial(Zap/g1/${mynum},90,T) but it didn't work, my problem is the delemiter, I have no delemiter, the default is "-" but how to

For make outgoing call, i setup 0. However 0 is write in the cdr dst field. Is there a way to remove it when asterisk send it to cdr_mysql ? exten => _0X.,1,Dial,SIP/${EXTEN:1}@mygateway I just want have in cdr dst = ${EXTEN:1} This don't work : exten => _0X.,1,SetVar(EXTEN=${EXTEN:1}) exten => _0X.,2,Dial,SIP/${EXTEN}@mygateway Use another variable still record ${EXTEN} --