Re: [Asterisk-Users] Unable to accept incoming PSTN calls
you have all these includes in your (messy) dial plan yet you didnt post the files that you use in include. --- Johnny Stork [EMAIL PROTECTED] wrote: I am new to Asterisk and the protocol/language complex world of VoIp and PBX. But I have a dedicated machine running [EMAIL PROTECTED] 2.8, a single TDM400P with one FXS module card connected to a standard analog phone. The second card is an X100P connected to my analog PSTN phone line. I also have Grandsteam IP phone plugged into the network and a couple of x-lite SIP softphones. I can make outgoing calls on the Grandstream or any registered SIP sofware phone from any computer. I can also get a dial tone from the analog phone connected to the ZAP X100P port. But when incoming callas come in, none of the phones ring. No VoIP trunks, just the single ZAP trunk from the X100P. Below are my configurations and a tail of /var/log/asterisk/full when making a call from an outside line. There is much more in the extensions.conf file but I was not sure how much to include and noticed in another post that only a couple sections were included. Also, when making an outside PSTN call comes in the other non-asterisk-connected phones in the house ring fine, but none of the asterisk-connected extensions/phones? sip.conf file: [general] bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) disallow=all allow=ulaw allow=alaw context = from-sip-external ; Send unknown SIP callers to this context callerid = Unknown #include sip_nat.conf #include sip_custom.conf #include sip_additional.conf #include additional_a2billing_sip.conf extensions.conf: zapata.conf file: ; ; Zapata telephony interface ; ; Configuration file [trunkgroups] [channels] language=en context=from-pstn signalling=fxs_ks rxwink=300; Atlas seems to use long (250ms) winks ; ; Whether or not to do distinctive ring detection on FXO lines ; ;usedistinctiveringdetection=yes usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no echotraining=800 rxgain=0.0 txgain=0.0 group=0 callgroup=1 pickupgroup=1 immediate=no ;faxdetect=both faxdetect=incoming ;faxdetect=outgoing ;faxdetect=no ;Include genzaptelconf configs #include zapata-auto.conf group=1 ;Include AMP configs #include zapata_additional.conf extensions.conf file: ; include extension contexts generated from AMP #include extensions_additional.conf ; Customizations to this dialplan should be made in extensions_custom.conf ; See extensions_custom.conf.sample for an example #include extensions_custom.conf [from-trunk] ; just an alias since VoIP shouldn't be called PSTN include = from-pstn [from-pstn] include = from-pstn-custom ; create this context in extensions_custom.conf to include customizations include = ext-did ;exten = fax,1,Goto(ext-fax,in_fax,1) exten = _.,1,Wait(1) exten = _.,2,Goto(from-pstn,s,1) var/log/asterisk/full (when recieving a call from pstn): Apr 26 18:43:33 VERBOSE[2696] logger.c: -- Remote UNIX connection Apr 26 18:43:52 VERBOSE[25804] logger.c: -- Remote UNIX connection disconnected Apr 26 18:44:57 VERBOSE[25810] logger.c: -- Starting simple switch on 'Zap/1-1' Apr 26 18:44:59 VERBOSE[25810] logger.c: -- Executing Wait(Zap/1-1, 1) in new stack Apr 26 18:45:00 VERBOSE[25810] logger.c: -- Executing Goto(Zap/1-1, from-pstn|s|1) in new stack Apr 26 18:45:00 VERBOSE[25810] logger.c: -- Goto (from-pstn,s,1) Apr 26 18:45:00 VERBOSE[25810] logger.c: -- Executing Wait(Zap/1-1, 1) in new stack Apr 26 18:45:00 DEBUG[2775] manager.c: Manager received command 'Command' Apr 26 18:45:00 DEBUG[2775] manager.c: Manager received command 'Command' Apr 26 18:45:01 VERBOSE[25810] logger.c: -- Executing Goto(Zap/1-1, from-pstn|s|1) in new stack Apr 26 18:45:01 VERBOSE[25810] logger.c: -- Goto (from-pstn,s,1) Apr 26 18:45:01 VERBOSE[25810] logger.c: -- Executing Wait(Zap/1-1, 1) in new stack Apr 26 18:45:01 DEBUG[25810] chan_zap.c: Exception on 17, channel 1 Apr 26 18:45:01 DEBUG[25810] chan_zap.c: Got event Ring Begin(1Cool on channel 1 (index 0) Apr 26 18:45:02 VERBOSE[25810] logger.c: -- Executing Goto(Zap/1-1, from-pstn|s|1) in new stack Apr 26 18:45:02 VERBOSE[25810] logger.c: -- Goto (from-pstn,s,1) Apr 26 18:45:02 VERBOSE[25810] logger.c: -- Executing Wait(Zap/1-1, 1) in new stack Apr 26 18:45:03 VERBOSE[25810] logger.c: -- Executing Goto(Zap/1-1, from-pstn|s|1) in new stack Apr 26 18:45:03 VERBOSE[25810] logger.c: -- Goto (from-pstn,s,1) Apr 26 18:45:03 VERBOSE[25810] logger.c: -- Executing Wait(Zap/1-1, 1) in new stack Apr 26 18:45:03 DEBUG[25810] chan_zap.c: Exception on 17, channel 1 Apr 26 18:45:03 DEBUG[25810] chan_zap.c: Got event
Re: [Asterisk-Users] Unable to accept incoming PSTN calls
[from-pstn] include = from-pstn-custom ; create this context in extensions_custom.conf to include customizations include = ext-did ;exten = fax,1,Goto(ext-fax,in_fax,1) exten = _.,1,Wait(1) exten = _.,2,Goto(from-pstn,s,1) Here is what is happening : Your ZAP channels are in the context from-pstn Since there is no s extension defined, it goes to _. (which match anything) So, like seen in the log, Asterisk wait a second, then execute Goto(from-pstr,s,1) which brings it back to _.,1. It just loop there until the caller hangup Since you're using [EMAIL PROTECTED], you have to go into AMP (or FreePBX) and click on Setup - Incoming Calls and define something to do with incoming calls hth ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Unable to accept incoming PSTN calls
Since I am using [EMAIL PROTECTED] 2.8 which now uses freePBX, there does not seem to be a menu area/settings for Incoming Calls? If you have a similiar setup, or know what the settings should be, could you possibly post them? If I were to create a dial group to ring all extensions, could that be used in place of s? Thanks kindly -Original Message- From: Time Bandit [mailto:[EMAIL PROTECTED] Sent: Thursday, April 27, 2006 6:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Unable to accept incoming PSTN calls [from-pstn] include = from-pstn-custom ; create this context in extensions_custom.conf to include customizations include = ext-did ;exten = fax,1,Goto(ext-fax,in_fax,1) exten = _.,1,Wait(1) exten = _.,2,Goto(from-pstn,s,1) Here is what is happening : Your ZAP channels are in the context from-pstn Since there is no s extension defined, it goes to _. (which match anything) So, like seen in the log, Asterisk wait a second, then execute Goto(from-pstr,s,1) which brings it back to _.,1. It just loop there until the caller hangup Since you're using [EMAIL PROTECTED], you have to go into AMP (or FreePBX) and click on Setup - Incoming Calls and define something to do with incoming calls hth ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Unable to accept incoming PSTN calls
For instance, I have tried the 2 below, but still it does not ring an existing extension, although the logs show it trying [from-pstn] include = from-pstn-custom ; create this context in extensions_custom.conf to include customizations include = ext-did ;exten = fax,1,Goto(ext-fax,in_fax,1) exten = _.,1,Wait(1) exten = _.,2,Goto(from-pstn,SIP/100,1) or [from-pstn] include = from-pstn-custom ; create this context in extensions_custom.conf to include customizations include = ext-did ;exten = fax,1,Goto(ext-fax,in_fax,1) exten = _.,1,Wait(1) exten = _.,2,Goto(from-pstn,100,1) -Original Message- From: Johnny Stork Sent: Thursday, April 27, 2006 7:11 AM To: asterisk-users Subject: RE: [Asterisk-Users] Unable to accept incoming PSTN calls Since I am using [EMAIL PROTECTED] 2.8 which now uses freePBX, there does not seem to be a menu area/settings for Incoming Calls? If you have a similiar setup, or know what the settings should be, could you possibly post them? If I were to create a dial group to ring all extensions, could that be used in place of s? Thanks kindly -Original Message- From: Time Bandit [mailto:[EMAIL PROTECTED] Sent: Thursday, April 27, 2006 6:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Unable to accept incoming PSTN calls [from-pstn] include = from-pstn-custom ; create this context in extensions_custom.conf to include customizations include = ext-did ;exten = fax,1,Goto(ext-fax,in_fax,1) exten = _.,1,Wait(1) exten = _.,2,Goto(from-pstn,s,1) Here is what is happening : Your ZAP channels are in the context from-pstn Since there is no s extension defined, it goes to _. (which match anything) So, like seen in the log, Asterisk wait a second, then execute Goto(from-pstr,s,1) which brings it back to _.,1. It just loop there until the caller hangup Since you're using [EMAIL PROTECTED], you have to go into AMP (or FreePBX) and click on Setup - Incoming Calls and define something to do with incoming calls hth ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to accept incoming PSTN calls
Johnny,You need to setup an Inbound Route that matches all DIDs and all CIDs. In FreePBX, click on Inbound Routes, create a new route with blank CID and DID, and point it where you want it to go. It should work after that. AlexOn 4/27/06, Johnny Stork [EMAIL PROTECTED] wrote: Since I am using [EMAIL PROTECTED] 2.8 which now uses freePBX, there does not seem to be a menu area/settings for Incoming Calls?If you have a similiar setup, or know what the settings should be, could you possibly post them? If I were to create a dial group to ring all extensions, could that be used in place of s?Thanks kindly -Original Message- From: Time Bandit [mailto:[EMAIL PROTECTED] ] Sent: Thursday, April 27, 2006 6:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Unable to accept incoming PSTN calls [from-pstn] include = from-pstn-custom ; create this context in extensions_custom.conf to include customizations include = ext-did ;exten = fax,1,Goto(ext-fax,in_fax,1) exten = _.,1,Wait(1) exten = _.,2,Goto(from-pstn,s,1) Here is what is happening : Your ZAP channels are in the context from-pstn Since there is no s extension defined, it goes to _. (which match anything) So, like seen in the log, Asterisk wait a second, then execute Goto(from-pstr,s,1) which brings it back to _.,1. It just loop there until the caller hangup Since you're using [EMAIL PROTECTED], you have to go into AMP (or FreePBX) and click on Setup - Incoming Calls and define something to do with incoming calls hth ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Robar[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to accept incoming PSTN calls
[from-pstn] include = from-pstn-custom ; create this context in extensions_custom.conf to include customizations include = ext-did ;exten = fax,1,Goto(ext-fax,in_fax,1) exten = _.,1,Wait(1) exten = _.,2,Goto(from-pstn,100,1) Try somethin like [from-pstn] include = from-pstn-custom ; create this context in extensions_custom.conf to include customizations include = ext-did exten = _.,1,Wait(1) exten = _.,2,Goto(from-pstn,s,1) exten = s,1,Answer exten = s,2,Dial(SIP/100,20) hth ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Unable to accept incoming PSTN calls
I actually tried
that before but it didnt seem to work. I tried once again and still nothing
rings, whether I set the destination to a single extension, or a ring group. But
the suggestion from another user below did work, but wont go to voicemail yet
when its not answered.
[from-pstn]
include = from-pstn-custom ; create this context in
extensions_custom.conf to include customizations
include = ext-did
exten = _.,1,Wait(1)
exten = _.,2,Goto(from-pstn,s,1)
exten = s,1,Answer
exten = s,2,Dial(SIP/100,20)
-Original Message-From: Alex Robar
[mailto:[EMAIL PROTECTED]Sent: Thursday, April 27, 2006
7:32 AMTo: Asterisk Users Mailing List - Non-Commercial
DiscussionSubject: Re: [Asterisk-Users] Unable to accept incoming
PSTN calls
Johnny,You need to setup an Inbound Route that matches all DIDs and
all CIDs. In FreePBX, click on Inbound Routes, create a new route with blank
CID and DID, and point it where you want it to go. It should work after that.
Alex
On 4/27/06, Johnny
Stork [EMAIL PROTECTED]
wrote:
Since
I am using [EMAIL PROTECTED] 2.8 which now uses freePBX, there does not seem to be a menu
area/settings for "Incoming Calls"?If you have a similiar setup, or
know what the settings should be, could you possibly post them? If I were to
create a dial group to ring all extensions, could that be used in place
of "s"?Thanks kindly -Original Message-
From: Time Bandit [mailto:[EMAIL PROTECTED] ]
Sent: Thursday, April 27, 2006 6:19 AM To: Asterisk Users Mailing
List - Non-Commercial Discussion Subject: Re: [Asterisk-Users]
Unable to accept incoming PSTN calls
[from-pstn] include = from-pstn-custom ; create this
context in extensions_custom.conf to include customizations
include = ext-did ;exten =
fax,1,Goto(ext-fax,in_fax,1) exten = _.,1,Wait(1)
exten = _.,2,Goto(from-pstn,s,1) Here is what is
happening : Your ZAP channels are in the context
"from-pstn" Since there is no "s" extension defined, it goes to "_."
(which match anything) So, like seen in the log,
Asterisk wait a second, then execute "Goto(from-pstr,s,1)" which
brings it back to "_.,1". It just loop there until the caller hangup
Since you're using [EMAIL PROTECTED], you have to go into AMP (or
FreePBX) and click on Setup - Incoming Calls and define
something to do with incoming calls hth
___ --Bandwidth and
Colocation provided by Easynews.com
-- Asterisk-Users mailing list To UNSUBSCRIBE or
update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth
and Colocation provided by Easynews.com
--Asterisk-Users mailing list To UNSUBSCRIBE or update options
visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Alex Robar[EMAIL PROTECTED]
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to accept incoming PSTN calls
I actually tried that before but it didnt seem to work. I tried once again and still nothing rings, whether I set the destination to a single extension, or a ring group. But the suggestion from another user below did work, but wont go to voicemail yet when its not answered. [from-pstn] include = from-pstn-custom ; create this context in extensions_custom.conf to include customizations include = ext-did exten = _.,1,Wait(1) exten = _.,2,Goto(from-pstn,s,1) exten = s,1,Answer exten = s,2,Dial(SIP/100,20) add this exten = s,3,Voicemail(u100) hth ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
