Inbound call answered by 's'

General Help
1

I’ve been able to configure asterisk for OUTbound calls (using freepbx), and used it to make all types of outbound calls using PSTN line terminating in FXO port 1 of digium TDM400 card. The called party hears fine. The caller hears his voice repeat.

The bigger problem, however, is that my incoming calls on same PSTN line are answered by extension ‘s’ as shown by freepbx reporting tool. I’ve defined 1 inbound route defined with DID/CID set to blank, zap channel set to 1, and core-destination set to <101> which is Xlite SIP extension capable of making outbound calls using ZAP channel 1.

The caller hears ‘goodbye’ after about 1 second and call disconnects.

Some of the log during that inbound call is:
Dec 5 13:33:15 VERBOSE[5852] logger.c: – Starting simple switch on ‘Zap/1-1’
Dec 5 13:33:15 NOTICE[5852] chan_zap.c: Got event 18 (Ring Begin)…
Dec 5 13:33:16 NOTICE[5852] chan_zap.c: Got event 2 (Ring/Answered)…
Dec 5 13:33:20 NOTICE[5852] chan_zap.c: Got event 18 (Ring Begin)…
Dec 5 13:33:20 VERBOSE[5852] logger.c: – Executing Playback(“Zap/1-1”, “vm-goodbye”) in new stack
Dec 5 13:33:20 DEBUG[5852] chan_zap.c: Took Zap/1-1 off hook
Dec 5 13:33:20 DEBUG[5852] chan_zap.c: Enabled echo cancellation on channel 1
ec 5 13:33:20 DEBUG[5852] cdr_addon_mysql.c: cdr_mysql: inserting a CDR record.
Dec 5 13:33:20 DEBUG[5852] cdr_addon_mysql.c: cdr_mysql: SQL command as follows: INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid) VALUES (‘2006-12-05 13:33:20’,’’,’’,‘s’,‘default’, ‘Zap/1-1’,’’,‘ResetCDR’,‘w’,0,0,‘ANSWERED’,3,’’,‘1165307595.27’)
Dec 5 13:33:20 VERBOSE[5852] logger.c: – Executing NoCDR(“Zap/1-1”, “”) in new stack
Dec 5 13:33:20 WARNING[5852] cdr.c: CDR on channel ‘Zap/1-1’ not posted
Dec 5 13:33:20 WARNING[5852] cdr.c: CDR on channel ‘Zap/1-1’ lacks end
Dec 5 13:33:20 VERBOSE[5852] logger.c: – Executing Wait(“Zap/1-1”, “5”) in new stack
Dec 5 13:33:26 VERBOSE[5852] logger.c: – Executing Hangup(“Zap/1-1”, “”) in new stack
Dec 5 13:33:26 VERBOSE[5852] logger.c: == Spawn extension (macro-hangupcall, s, 4) exited non-zero on ‘Zap/1-1’ in macro 'hangupcall’
Dec 5 13:33:26 VERBOSE[5852] logger.c: == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'Zap/1-1’
Dec 5 13:33:26 DEBUG[5852] chan_zap.c: Hangup: channel: 1 index = 0, normal = 16, callwait = -1, thirdcall = -1
Dec 5 13:33:26 DEBUG[5852] chan_zap.c: disabled echo cancellation on channel 1
Dec 5 13:33:26 DEBUG[5852] chan_zap.c: Set option TDD MODE, value: OFF(0) on Zap/1-1
Dec 5 13:33:26 DEBUG[5852] chan_zap.c: Updated conferencing on 1, with 0 conference users
Dec 5 13:33:26 VERBOSE[5852] logger.c: – Hungup ‘Zap/1-1’

2

Ok just so u know there is no issue with Freepbx and inbound routes

It works …YOURS may not but that is on YOUR END…

So how about we go about seeing what is wrong with your setup…

What happens when u dail 7777 ???

Your 1 st post said exten 101 and then u say 111 which is it??

And do u have those extens setup and have vm setup???

just go in and create a BLANK in bound route going to a exten with voice mail nothing else.

3

dialing 7777 prompts for voicemail with default greeting.

A blank inbound rule with call forwarded to extension with voicemail is exactly the current setup which isnt working. SetDestination->Core has been tried with both ext-111 and voicemail-box-111.

When a call is made to zap channel, say zap-1 - the output from that call is pasted in original message. the caller hears goodbye and call hangs up. the zap-channel on FOP is temporarily red/busy.

To give idea of current setup - all types of outbound calls are working. VoIP calls and VM to all configured ext is working. Similarly outbound/domestic calls using the same zap-1 are working too.

Could there be something to look in zaptel/zapata files related to this? I haven’t touched those files at all as the default asterisk+freepbx has worked flawlessly for outbound and asterisk-internal calls.

4

post them here

5

I’ve explained the problem and my configuration in quite detail. The caller hears a Goodbye message and call hangs up.

I cannot, and dont need to put a DID. I simply want all calls on ZAP channel to go extension 111.

And DID is not possible - the FXO ports are connected to an analog PBX. PSTN lines terminate at the PBX. PBX doesn’t support callerID hence the FXO channels are oblivious to the fact as to who is calling them.

6

I’ve been trying to find a solution but the basic freepbx configuration seems ok based on all the reading. I am attaching relevant setcion from extension_additional.conf in /etc/asterisk:

[macro-from-zaptel-1]
include => macro-from-zaptel-1-custom
exten => s,1,Noop(Entering macro-from-zaptel-1 with DID = ${DID})
exten => s,n,Set(FROM_DID=s)
exten => s,n,Goto(ext-local,111,1)
; end of [macro-from-zaptel-1]

[ext-did]
include => ext-did-custom
exten => s,1,Noop(No DID or CID Match)
exten => s,n,Answer
exten => s,n,Wait(2)
exten => s,n,Playback(ss-noservice)
exten => s,n,SayAlpha(${FROM_DID})
exten => _[#X].,1,Set(FROM_DID=${EXTEN})
exten => _[#X].,n,Noop(Received an unknown call with DID set to ${EXTEN})
exten => _[*#X].,n,Goto(ext-did,s,1)

7

I continue to have this problem with default installation of freepbx and asterisk!

I could get rid of the freepbx and go back to reinstalling * and create ext and routes by hand in config files.

But since every thing else (creating extension, outbound plans, voice mail) has worked flawlessly with freepbx, I am surprised a simple inbound route isn’t working and no one has had this problem?

Please throw a bone! or 2.

8

I have configured more than 4-5 inbound routes with freepbx and all of them worked properly . What problem are you getting ?
Make sure to put ur phone number ( DID ) in did section of incoming route ( wthout countrycode 1 ) also in trunks sections register  like username:[email protected]/didnumber
like
user:[email protected]/3333333333

On 08/12/06, farzal <[email protected] ([email protected])> wrote:[quote] I continue to have this problem with default installation of freepbx and asterisk!

I could get rid of the freepbx and go back to reinstalling * and create ext and routes by hand in config files.

But since every thing else (creating extension, outbound plans, voice mail) has worked flawlessly with freepbx, I am surprised a simple inbound route isn’t working and no one has had this problem?

Please throw a bone! or 2.

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