Sir,
On Tue, May 10, 2011 at 6:15 PM, A J Stiles <asterisk_l...@earthshod.co.uk>wrote: > (Reformatted *again*. The proper place to post your reply is *below* the > message or section to which you are replying, so it reads like: question, > answer, question, answer. This makes things much easier for anyone with a > similar problem in future, trying to make sense of old messages in the > archives.) > > sorry sir I am new to asterisk, > On Tuesday 10 May 2011, mahesh katta wrote: > > On Tue, May 10, 2011 at 4:49 PM, A J Stiles > > > > <asterisk_l...@earthshod.co.uk>wrote: > > > "Not working" can mean a lot of things. > > > > > > So, let's start at the beginning. Have you ever actually managed to > get > > > an outgoing call to work *at all* -- i.e., successfully placed a call > > > onto a trunk, made an external phone ring and been able to speak to the > > > other party, > > > even if the caller ID that shew up was incorrect ? > > > > > > Assuming so, please show the dialplan section that worked for that. > > > > > > > > > Otherwise, you need to go back to first principles and work out exactly > > > what > > > you are doing wrong. We need to start from a point where you are able > to > > > place external calls *before* we can discuss how to set the desired > > > caller ID > > > to appear on the receiving end. > > > > > > Obligatory car analogy: There's no point arguing over whether > motorways > > > are > > > quicker than A-roads, if you can't even get the engine to start. > > > > > > > > > Next, please describe how to determine, based on the internal > extension, > > > what > > > caller ID number should be presented to the outside world. > > > > > A.j Sir, > > > > I am using vicidial server asterisk box. it has asterisk 1.27v , > > > > In sip configuration is extensiosn like below > > > > [5001] > > username=5001 > > secret=1234 > > mailbox=5001 > > type=friend^M > > host=dynamic^M > > canreinvite=no^M > > qualify=yes^M > > nat=yes^M > > context=default > > > > 5001 to 5099 > > > > and PRI pilot no. is 4457900 in dubai , telcom give a DID's 4457900 to > > 4457999, > > > > when i am dialing from pstn on DID's(inbound) that will connecting (i was > > configured with sip means when i dial the 4457901 from outside i will get > a > > call on 5001 extension) > > O.K. So what are you seeing in the console when you place an outgoing > call? > There should be two NoOp() lines per call; one with the original extension > number, and another with what we are trying to set the caller ID to. > > If it helps, change the lines for steps 1 and 3 as follows: > > exten => _90XXXXXXXXX,1,NoOp(Int exten:${CALLERID(num)}) > exten => _90XXXXXXXXX,3,NoOp(Ext ident:${outgoing_ident}) > > so it is more obvious what they are supposed to represent. Run > asterisk -vvvvvvvvvr in a console, reload the dialplan, capture some output > as you make calls from various extensions, and paste it here. The > important > question is: Do the "Ext ident" numbers look like what they are supposed > to? > And if not, then how do they differ from what they are supposed to look > like? > > NoOp("SIP/5001-b792a1a8", "Int exten:044578999") in new stack -- Executing Set("SIP/5001-b792a1a8", "outgoing_ident=044578999") in new stack -- Executing NoOp("SIP/5001-b792a1a8", "Ext ident:044578999") in new stack -- Executing Set("SIP/5001-b792a1a8", "CALLERID(name)=044578999") in new stack -- Executing AGI("SIP/5001-b792a1a8", "agi://127.0.0.1:4577/call_log") in new stack -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 -- Executing Set("SIP/5001-b792a1a8", "CALLERID(num)=044578999") in new stack -- Executing MixMonitor("SIP/5001-b792a1a8", "/var/spool/asterisk/astrec/20110510-172343-044578999-90559566768-1305033823.129.gsm|av(0)V(0)") in new stack -- Executing Dial("SIP/5001-b792a1a8", "Zap/g0/0559566768||tTo") in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g0/0559566768 == Begin MixMonitor Recording SIP/5001-b792a1a8 -- Zap/1-1 is proceeding passing it to SIP/5001-b792a1a8 -- Zap/1-1 is making progress passing it to SIP/5001-b792a1a8 -- Hungup 'Zap/1-1' == Spawn extension (default, 90559566768, 8) exited non-zero on 'SIP/5001-b792a1a8' -- Executing DeadAGI("SIP/5001-b792a1a8", "agi:// 127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0-----CANCEL----------") in new stack -- AGI Script agi:// 127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0-----CANCEL----------completed, returning 0 == End MixMonitor Recording SIP/5001-b792a1a8 > > > > here Problem when i dial (out bound) out of box from 5001 extensions the > > caller id showing pilot no. (4457900) . > > I need to display that DID's each extension. > > The usual cause of the caller ID not being set to what you want, is that > you > tried to set it to some number that is not allocated to you, and your telco > changed it to some default. (Otherwise, you could fake your outgoing > caller > ID, which nobody wants.) > > sir i complaint to telco for this they are said that is set on router. > > > exten => _90XXXXXXXXX,1,NoOp(${CALLERID(num)}) > > exten => _90XXXXXXXXX,2,Set(outgoing_ident=0445789${CALLERID(num):-2}) > > exten => _90XXXXXXXXX,3,NoOp(${outgoing_ident}) > > exten => _90XXXXXXXXX,4,Set(CALLERID(name)=${outgoing_ident}) > > exten => _90XXXXXXXXX,5,AGI(agi://127.0.0.1:4577/call_log) > > exten => _90XXXXXXXXX,6,Set(CALLERID(num)=${outgoing_ident}) > > exten => > > > _90XXXXXXXXX,7,MixMonitor(/var/spool/asterisk/astrec/${TIMESTAMP}-${CALLERI > >DNUM}-${EXTEN}-${UNIQUEID}.gsm|av(0)V(0)) exten => > > _90XXXXXXXXX,8,Dial(${TRUNK}/${EXTEN:1},,tTo) > > exten => _90XXXXXXXXX,9,Hangup > > Right, I'm confused now. This dialplan segment is expecting for you to > dial > 90, followed by nine digits. But the numbers as which you want to > identify -- 4457901 to 4457999 -- are only seven digits long. > > sir we are dialing out of world when we dial out of world 4457901 to 4457999 will show them. these are DID's which is telco provided to us. > I'm guessing there probably is an STD code consisting of 0 followed by two > digits to indicate the town, and then a 7-digit local number within the > town. > In which case, I seem to have made a mistake at step 2, somehow letting an > extraneous figure 8 get in. Try: > > exten => _90XXXXXXXXX,2,Set(outgoing_ident=44579${CALLERID(num):-2}) > > Executing NoOp("SIP/5001-b790fc10", "Int exten:044578999") in new stack -- Executing Set("SIP/5001-b790fc10", "outgoing_ident=44578999") in new stack -- Executing NoOp("SIP/5001-b790fc10", "Ext ident:44578999") in new stack -- Executing Set("SIP/5001-b790fc10", "CALLERID(name)=44578999") in new stack -- Executing AGI("SIP/5001-b790fc10", "agi://127.0.0.1:4577/call_log") in new stack -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 -- Executing Set("SIP/5001-b790fc10", "CALLERID(num)=44578999") in new stack -- Executing MixMonitor("SIP/5001-b790fc10", "/var/spool/asterisk/astrec/20110510-174142-44578999-90559566768-1305034902.131.gsm|av(0)V(0)") in new stack -- Executing Dial("SIP/5001-b790fc10", "Zap/g0/0559566768||tTo") in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g0/0559566768 == Begin MixMonitor Recording SIP/5001-b790fc10 -- Zap/1-1 is proceeding passing it to SIP/5001-b790fc10 -- Zap/1-1 is making progress passing it to SIP/5001-b790fc10 -- Hungup 'Zap/1-1' == Spawn extension (default, 90559566768, 8) exited non-zero on 'SIP/5001-b790fc10' -- Executing DeadAGI("SIP/5001-b790fc10", "agi:// 127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0-----CANCEL----------") in new stack -- AGI Script agi:// 127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0-----CANCEL----------completed, returning 0 == End MixMonitor Recording SIP/5001-b790fc10 > If that does not seem to work, try prefixing the "44579" with the STD code > for > your town, with and without the leading zero. And don't forget to reload > the > dialplan between edits :) > > > -- > AJS > > Answers come *after* questions. > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Best Regards, Mahesh Katta *BUZZ**WORKS* Business Services Private Limited BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI 201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E) Mumbai 400069 GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634 Web http://www.buzzworks.com
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users