search for: 5551234

Hi All, I am new to Asterisk. Now I got one question on the identifier parameter of the Dial() command. I saw as below: exten => 20,1,Dia(Zap/3/5551234). Would you please let me know the meaning of "5551234"? Thanks, Songtao -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090907/d527e36f/attachment.htm

....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)

Hello For some reason, when dialing out through a call file and the remote line is busy, Asterisk doesn't jump to the "failed" extension in the context used by the call file: ====== call file Channel: Zap/1/5551234 Context: callbacktest Extension: start Priority: 1 MaxRetries: 1 ====== extension.conf [callbacktest] exten => start,1,NoOp(Status is ${DIALSTATUS}) exten => start,n,Wait(10) exten => start,n,Hangup exten => failed,1,NoOp(Reason call file failed is ${REASON}) ====== CLI ip04*CLI>...

Just throwing this out here, hopefully someone can tell me why. *CLI> show version Asterisk CVS-HEAD-11/17/04-10:16:38 built by root@wanderer on a i686 running Linux Zap/g1 is pri_cpe to Bell Canada 5551234 is a normal POTS line I have busied out (handset offhook) exten => 1234,1,Dial(Zap/g1/5551234,,g) exten => 1234,n,NoOp(HANGUPCAUSE is ${HANGUPCAUSE} and DIALSTATUS is ${DIALSTATUS}) Why, if 5551234 is busy, is DIALSTATUS set to CHANUNAVAIL? Should it not be BUSY? And now with IAX2 (I am...

This wav produces a bit of audible static when encoded at the highest bitrate vorbis will allow me to encode at (30.7kbps avg.): http://staff.xmms.org/zinx/misc/5551234.wav.gz An encoded version, with the bitrate set to approximate 128kbps (I think it output to 29 some odd kbps): http://staff.xmms.org/zinx/misc/5551234.ogg This is with the latest CVS tree as of Sat May 12 10:06:17 UTC 2001 -- Zinx Verituse --- >8 ---- List archives: http://www.xiph.o...

...m trying to change the Origination Number on my outgoing PRI, and running into a weird problem. If I make a call from a SIP extension off asterisk using the following context: [from-sip] exten => 800,1,Answer exten => 800,2,SetCallerID(6132718xxxx) exten => 800,3,Dial(Dial(${TRUNK-TELCO}/5551234) I am able to change the Origination Number! > Protocol Discriminator: Q.931 (8) len=39 > Call Ref: len= 2 (reference 175/0xAF) (Originator) > Message type: SETUP (5) > [04 03 80 90 a2] > Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) >...

...ew seconds, hang up, and ;check ChanIsAvail() able to detect when line idle again exten => 8888,1,Answer() exten => 8888,n,Playback(/var/tmp/manolo_camp-morning_coffee) exten => 8888,n,Hangup ========== CLI ;keep line engaged for a few seconds, and hang up from remote end originate Zap/1/5551234 extension 8888 at internal ========== extensions.conf ;call from XLite to check line status ;Loop until Zap/1 is available exten => 1111,1,Set(INDEX=0) exten => 1111,n,While(1) exten => 1111,n,ChanIsAvail(Zap/1) exten => 1111,n,GotoIf($["${AVAILORIGCHAN}" != "" | $...

Hi I have a simple question but I cannot find the answer. I have a line with 2 different phone numbers I want to redirect each phone number called to a different IP phone Example Someone calls 5551234 and the call is redirected to IP phone 192.168.0.2 Someone calls 5551235 and the call is redirected to IP phone 192.168.0.3 Thanks Patrick

...this for some time now. For incoming calls, I'd like to send my users a "localized" caller id number. By "localized," I mean one with out the 1+areacode for local calls and only 10 digits (minus the leading 1) for long distance calls. For example: I get a call from 15055551234. Since I live in the 505, I should see: 5551234 on my caller id. However, if I get a call from 18035556789, I should see: 8035556789 on my caller id. The problem is that I also want to preserve the original calling number in my CDR(src) field. But everytime I change the CID number, it changes...

Hi, When a VOIP user dials an external number, the calls are routed through our SIP provider. Is there a simple way to check whether the DDI exists locally before dialling out to the sip provider? Something like GotoIfExists(5551234 at incoming_calls) Currently, I'm paying for all calls, regardless of whether they exist locally. All DDIs exist in the incoming_calls context. Thanks Dan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachmen...

Good morning, I have a Bell Canada PRI here (switchtype=national) and I am trying to perform a call-forward-unconditional on one of the DIDs. The idea is that when DID 5551234 receives a call, Asterisk redirects it back out the same PRI to some external number. This is simple enough to do with something along these lines: [PRI] exten => 5551234,1,Set(CALLERID(RDNIS)=${EXTEN}) exten => 5551234,n,Dial(Zap/g1/5556789) This is a brute-force approach but there are t...

I am having trouble with a DID on a PRI. If there is a call to that DID (let say 5551234) , the next calls get a busy signal. How to I go about sending the call to the next available channel ? Thanks! G. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090715/8a5ba5f7/attachment.htm

...et up call forwarding to a mobile on extension 201, and then place a call from extension 202, the call gets diverted. I answer the call and talk for 30 seconds, then I hang up. The CDR shows two calls:- 2010-08-27 13:38:24 - 202 -> 201 - Answered - Billsec is 30 2010-08-27 13:38:24 - 202 -> 5551234 - Answered - Billsec is 0 5551234 is the mobile number. The second CDR entry should read 30 seconds, and the first should read 0 (or 30) Since it isn't behaving like I want, is there any way to disable the feature that allows a SIP phone to perform call forwarding? Thanks Dan --------------...

...some call queueing, with the slightly unusual caveat that the destination for the calls is not a phone or group of phones connected to my local asterisk box, but an "external" PSTN number. Can I setup a queue in asterisk and make the queue "member" an external address like SIP/5551234@my.pstn.gateway? There will be a smaller number of PSTN lines available at the far end destination than there are inbound calls queueing, so after X number of calls, attempts to call that "agent" will receive a busy response back until a call in progress is finished and a line becomes av...

...If($[${LEN(${NBR2CALL})} != 10]?nbr2call) exten => s,n,Playback(phone:${NBR2CALL},say) exten => s,n(end),Wait(2) exten => s,n,Hangup() ============== I notice that it sometimes works fine, but sometimes, Asterisk hangs up while I'm still typing: ============== CLI> originate Zap/1/5551234 extension s at test Executing [s at test:3] Read("Zap/1-1", "NBR2CALL|please-type-number|||2|30") in new stack -- <Zap/1-1> Playing 'please-type-number' (language 'fr') -- User disconnected == Spawn extension (test, s, 3) exited non-zero on 'Zap/1-1...

> At any rate, if I can figure out the right way to set the caller ID explicitly, and assuming they honor it if I do, then none of this will matter. Ok, so just do exten => s,n,Set(CALLERID(all)=?Greg? <5551234>) https://wiki.asterisk.org/wiki/display/AST/Function_CALLERID -- Cheers, Matt Riddell _______________________________________________ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/pabx_on_disk.php (PABX on a Disk) http://www.venturevoip.com/exchange.ph...

..."capi debug" trace where i SEND >UUS1 from a normal ISDN Phone TO an asterisk: > > Controller/PLCI/NCCI = 0x101 > CIPValue = 0x10 > CalledPartyNumber = <c1>5559999 > CallingPartyNumber = <01 81>5551234 > CalledPartySubaddress = default > CallingPartySubaddress = default > BC = <80 90 a3> > LLC = default > HLC = <91 81> > AdditionalInfo > BChannelinformat...

Hello, When I dial out there is a long delay in dialing. Is this normal? Thanks, David

...y. I had considered, of course, that it was a problem at their end. For the call, I see this on the CLI: *********************************************************** Verbosity is at least 6 -- Starting simple switch on 'Zap/2-1' -- Executing Dial("Zap/2-1", "Zap/g4/w5551234") in new stack -- Called g4/w5551234 -- Zap/4-1 answered Zap/2-1 -- Attempting native bridge of Zap/2-1 and Zap/4-1 -- Hungup 'Zap/4-1' == Spawn extension (localphone, 5551234, 1) exited non-zero on 'Zap/2-1' -- Hungup 'Zap/2-1' *******************...

I have been fighting with the ability to set the caller ID when I make outbound calls via a PRI line as well as via my SIP provider. The more I play around the less I understand. There is a setting in chan_dahdi.conf that seems to say do not pay attention to the CALLERPRES value and just allow the ID to be set. This setting is usecallingpres. If this is set to yes then the value of CALLERPRES