search for: waitfordigits

Hello I have an interesting situaltion and not sure if I am doing something wrong or it is a BUG. I Installed Rhino Channel on T1 line and connected Analog Phone on Rhino's Zap Channels. If i pickup analog phone and hangup without dialing any number , I am getting extra ring after hangup and if i dial any digit than there is no ring on Analog phone after hangup. Log's looks like this

...until I hang up. Here my extensions.conf: [incoming] exten => 20,1,Answer() exten => 20,2,Goto(menu,s,1) exten => 20,3,Hangup() exten => 492774XXXX,1,Answer() exten => 492774XXXX,2,Goto(menu,s,1) exten => 492774XXXX,3,Hangup() [outgoing] exten => _0.,1,WaitforDigits(5000) exten => _0.,2,Dial(SIP/${EXTEN}@sip.1und1.de <mailto:SIP/$%7bEXTEN%7d@sip.1und1.de> ) /// Normal Way --> outgoing call to SIP-Gateway (TIMEOUT) exten => _0.,3,Dial(misdn/1/${EXTEN}) exten => _999.,1,WaitforDigits(5000) exten => _999.,2,Dial(misdn/1/${EXTEN:3})...

I have a Cisco 7970 that I am tying to get working. The phone powers up and registers but nothing else. I am using the development build. Any thoughts? The message I get when the phone powers on and registers. *CLI> skinny debug Skinny Debugging Enabled -- Starting Skinny session from 192.168.101.6 Recieved AlarmMessage Device SEP000F23FCA366 is attempting to register --

In Asterisk 1.8, I used the following line in extensions.conf to allow me to pass "*82" in front of a dialed number, to disable the callerid block that's normally on that POTS line: ; disable callerid block exten => _*82.,1,Dial(${POTS}/${EXTEN}) But this seems to have stopped working when I upgraded to Asterisk 11.7. I get the following debug output, with a "no

Hello I have an interesting situaltion and not sure if I am doing something wrong or it is a BUG. I Installed Rhino Channel on T1 line and connected Analog Phone on Rhino's Zap Channels. When I pickup analog phone and hangup without dialing any number , I am getting extra ring after hangup and if I dial any digit than there is no ring on Analog phone after hangup. Log's looks like this

hi, i've a wirded problem, i try to dial out, using this dialplan [default] exten => _*7.,1,Macro(anrufextern-sip,${EXTEN:2}) [macro-anrufextern-sip] exten => s,1,SetCallerID(SIP-ID) exten => s,n,Dial(SIP/${ARG1}sip-out) exten => s,n,Hangup() when i use my analog telephone, everything is okay: - Starting simple switch on 'Zap/3-1' -- Executing

Hello I have an interesting situaltion and not sure if I am doing something wrong or it is a BUG. I Installed Rhino Channel on T1 line and connected Analog Phone on Rhino's Zap Channels. If i pickup analog phone and hangup without dialing any number , I am getting extra ring after hangup and if i dial any digit than there is no ring on Analog phone after hangup. Log's looks like this

Hi, I need help finding why my system is dropping calls.. I enabled debugging on my box in the hope it would lead me to the answer as to why my system is dropping calls but unfortunately nothing is jumping out at me.. I have attached the portion of the messages file for two calls that were dropped.. (numbers names and ip's have been changes to protect the innocent) Can someone give me a

Greetings everyone, I'm having some issues connecting a Asterisk box to a old ISDN PBX. Everything works fine but the undetermined digits rules. For instance, if I have _00X. and I start dialing for instance 0035..., Asterisk just get the 4 first numbers and starts dialing 0035. I've tried adding: stop_tone_after_first_digit = yes append_digits2exten = yes Although I

Hi, I'm having problems using a TDM400P Card. I put my SIM card in a Nokia Premicell and connected it to a TDM400P card but when I make calls to the number, it hangs up automatically. The card also can't call out. Any ideas? I've searched the archives without much success. I appreciate all your help. Details: I'm using Asterisk 1.2.17 on Fedora Core release 5 (Bordeaux). On an

Howdy, 1 x TDM400P card with 1 x fxo module. 1 x BT Pots line. Location - UK Calls work fine outbound but i'm unable to pickup the inbound calls. Asterisk debug: Asterisk -vvvvvvvvvvcg *CLI> -- Starting simple switch on 'Zap/1-1' -- Executing Wait("Zap/1-1", "1") in new stack -- Executing Answer("Zap/1-1", "") in new stack

Can anyone recommend a good voip provider? Thanks

Hi all Does Digium make a quad BRI card? I can't see anything of the sort on their page but I thought they might call it something else in the States. Failing that, can anyone recommend a make/model that would handle 4 BRI ports? -- Cheers Wayne

...; as well, since chan_misdn has no chance to distinguish if the L1 is down ; because of a lost Link or because the Provider shut it down... ; ; default: yes ; pmp_l1_check=yes ; ; Send Setup_Acknowledge on incoming calls anyway (instead of PROCEEDING), ; this requests additional Infos, so we can waitfordigits ; without much issues. This works only for PTP Ports ; ; default value: no ; need_more_infos=no ; set the method to use for channel selection: ; standard - always choose the first free channel with the lowest number ; round_robin - use the round robin algorithm to select a channel. use th...

Problem: *Call* *waiting* comes in, I press flash to answer it, and the first caller gets disconnected after 3 seconds. This is all ZAP - no VOIP. System: Analog stations and trunks running on a pair of TDM400's. It does NOT have * call* *waiting* from the phone company, and I have enabled it in all my conf files. The trunks are set to FXSKS and the stations are FXOKS. I am not using *call*