Capstone Works
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- Apr 1, 2014
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So my problem is this:
For SIP Inbound calls in context [from-trunk-sip-nexVortex] , callers SUCCESSFULLY hear the ringing before the call is picked up by an extension or ring group
For DAHDI (Grande T1 PRI) Inbound calls in context [from-digital], callers hear SILENCE while the extension or ring group rings, until the call is picked up by an person or voicemail.
In sip_custom.conf we've tried various combinations of
pedantic=no
prematuremedia=no
progressinband=yes
rtcachefriends=yes
rtsavesysname=yes
rtupdate=yes
rtautoclear=yes
ignoreregexpire=0
in sip_general_additional we've tried
<snip>
progressinband=yes
pedantic=no
<snip>
No luck with any of these sorts of settings - SO we moved on to DialPlan modification with SUCCESS, but not FreePBX tolerant success....
In summary, we were able to add exten => 8922,n,Playtones(ring) DIRECTLY to extensions_additional.conf, but we have not been successful in adding it to extensions_custom.conf in a way that would still let us MANAGE our extension definitions from within FreePBX
Any help would be appreciated!
Here goes the details:
------------------
Sanitized summary details, below.
<----------------------------------------------------->
Asterisk = ONLINE | Dahdi = ONLINE | MySQL = ONLINE x
SSH = ONLINE | Apache = ONLINE | Iptables = ONLINE x
Fail2ban = ONLINE | Internet = ONLINE | Ip6Tables = ONLINE x
Disk Free = ADEQUATE| Mem Free = ADEQUATE | NTPD = ONLINE x
SendMail = ONLINE | Samba = ONLINE | Webmin = ONLINE x
Ethernet0 = ONLINE | Ethernet1 = N/A | Wlan0 = N/A x
PIAF Installed Version = 2.0.6.5 under *HARDWARE* x
FreePBX Version = 2.11.0.38 x
Running Asterisk Version = 11.10.2 x
Asterisk Source Version = 11.8.1
Dahdi Source Version = 2.9.0
Libpri Source Version = 1.4.14
IP Address = x.x.x.x on eth0 x
Operating System = CentOS release 6.5 (Final) x
Kernel Version = 2.6.32-431.1.2.0.1.el6.x86_64 - 64 Bit x
Incredible Version = 11.8
CDR-MYSQL = OK | CDR-CUSTOM = OK | Flite Engine = OK x
G722 Codec = OK | G726 Codec = OK | Speex Codec = OK x
Resample Codec = OK | Alaw Codec = OK | Ulaw Codec = OK x
GSM Codec = OK | ILBC Codec = OK | Dahdi Codec = OK x
Jabber Connect = N/A | Gtalk Channel = N/A | Silk Codec = OK x
Siren7 Codec = OK | Siren14 Codec = OK | G729 Codec = N/A x
SCCP-B Codec = N/A | Motif Codec = OK | x
Asterisk Uptime = 10 minutes, 51 seconds x
System Uptime = 72 hours x
IAX2 Registrations = 0 IAX2 registrations. x
SIP Registrations = 0 SIP registrations. x
Parked Calls = 0 parked calls in total. x
Installed ISO Version = 20650 x
System Installed on = 2014-03-17T14:16-0500 x
Kickstart Method = Normal ks kickstart
<----------------------------------------------------->
In extensions.conf
; from-digital:
;
; Context to set for PRI's and equivalent
;
[from-digital]
exten => s,1,Playtones(ring)
include => from-digital-custom
include => from-pstn
;-------------------------------------------------------------------------------
<----------------------------------------------------->
In extensions_additional.conf
[from-trunk-sip-nexVortex]
include => from-trunk-sip-nexVortex-custom
exten => _.,1,Set(GROUP()=OUT_19)
exten => _.,n,Goto(from-trunk,${EXTEN},1)
;--== end of [from-trunk-sip-nexVortex] ==--;
[ext-did-0002]
include => ext-did-0002-custom <= Added this to try to add Playtopnes(ring) to this entire context (see below), but it does not work
exten => fax,1,Goto(${CUT(FAX_DEST,^,1)},${CUT(FAX_DEST,^,2)},${CUT(FAX_DEST,^,3)})
<snip>
exten => 8922,1,Set(__FROM_DID=${EXTEN})
exten => 8922,n,Gosub(app-blacklist-check,s,1())
exten => 8922,n,Set(CDR(did)=${FROM_DID})
exten => 8922,n,ExecIf($[ "${CALLERID(name)}" = "" ] ?Set(CALLERID(name)=${CALLERID(num)}))
exten => 8922,n,Set(CHANNEL(musicclass)=default)
exten => 8922,n,Set(__MOHCLASS=default)
exten => 8922,n,Set(__CALLINGPRES_SV=${CALLERPRES()})
exten => 8922,n,Set(CALLERPRES()=allowed_not_screened)
exten => 8922,n,Playtones(ring) <= Manually added this, it works, but FreePBX will of course overwrite
exten => 8922,n(dest-ext),Goto(from-did-direct,126,1)
<snip>
;--== end of [ext-did-0002] ==--;
[ext-did]
include => ext-did-custom
include => ext-did-0001
include => ext-did-0002
exten => foo,1,Noop(bar)
;--== end of [ext-did] ==--;
<----------------------------------------------------->
Calls come in via DHADI in context [from-digital]
[from-digital]
exten => s,1,Playtones(ring) <= Tried it here also, but this does not see to hav and effect
include => from-digital-custom
include => from-pstn
[from-pstn]
include => from-pstn-custom ; create this context in extensions_custom.conf to include customizations
include => ext-did
include => ext-did-post-custom
include => from-did-direct
include => ext-did-catchall ; THIS MUST COME AFTER ext-did
<----------------------------------------------------->
In extensions_custom.conf, tried adding these
[ext-did-0002-custom]
exten => s,1,Playtones(ring)
;# // END ext-did-0002-custom
[ext-did-custom]
exten => s,1,Playtones(ring)
;# // END ext-did-custom
Thanks again for taking a look at this - its driving me crazy!
For SIP Inbound calls in context [from-trunk-sip-nexVortex] , callers SUCCESSFULLY hear the ringing before the call is picked up by an extension or ring group
For DAHDI (Grande T1 PRI) Inbound calls in context [from-digital], callers hear SILENCE while the extension or ring group rings, until the call is picked up by an person or voicemail.
In sip_custom.conf we've tried various combinations of
pedantic=no
prematuremedia=no
progressinband=yes
rtcachefriends=yes
rtsavesysname=yes
rtupdate=yes
rtautoclear=yes
ignoreregexpire=0
in sip_general_additional we've tried
<snip>
progressinband=yes
pedantic=no
<snip>
No luck with any of these sorts of settings - SO we moved on to DialPlan modification with SUCCESS, but not FreePBX tolerant success....
In summary, we were able to add exten => 8922,n,Playtones(ring) DIRECTLY to extensions_additional.conf, but we have not been successful in adding it to extensions_custom.conf in a way that would still let us MANAGE our extension definitions from within FreePBX
Any help would be appreciated!
Here goes the details:
------------------
Sanitized summary details, below.
<----------------------------------------------------->
Asterisk = ONLINE | Dahdi = ONLINE | MySQL = ONLINE x
SSH = ONLINE | Apache = ONLINE | Iptables = ONLINE x
Fail2ban = ONLINE | Internet = ONLINE | Ip6Tables = ONLINE x
Disk Free = ADEQUATE| Mem Free = ADEQUATE | NTPD = ONLINE x
SendMail = ONLINE | Samba = ONLINE | Webmin = ONLINE x
Ethernet0 = ONLINE | Ethernet1 = N/A | Wlan0 = N/A x
PIAF Installed Version = 2.0.6.5 under *HARDWARE* x
FreePBX Version = 2.11.0.38 x
Running Asterisk Version = 11.10.2 x
Asterisk Source Version = 11.8.1
Dahdi Source Version = 2.9.0
Libpri Source Version = 1.4.14
IP Address = x.x.x.x on eth0 x
Operating System = CentOS release 6.5 (Final) x
Kernel Version = 2.6.32-431.1.2.0.1.el6.x86_64 - 64 Bit x
Incredible Version = 11.8
CDR-MYSQL = OK | CDR-CUSTOM = OK | Flite Engine = OK x
G722 Codec = OK | G726 Codec = OK | Speex Codec = OK x
Resample Codec = OK | Alaw Codec = OK | Ulaw Codec = OK x
GSM Codec = OK | ILBC Codec = OK | Dahdi Codec = OK x
Jabber Connect = N/A | Gtalk Channel = N/A | Silk Codec = OK x
Siren7 Codec = OK | Siren14 Codec = OK | G729 Codec = N/A x
SCCP-B Codec = N/A | Motif Codec = OK | x
Asterisk Uptime = 10 minutes, 51 seconds x
System Uptime = 72 hours x
IAX2 Registrations = 0 IAX2 registrations. x
SIP Registrations = 0 SIP registrations. x
Parked Calls = 0 parked calls in total. x
Installed ISO Version = 20650 x
System Installed on = 2014-03-17T14:16-0500 x
Kickstart Method = Normal ks kickstart
<----------------------------------------------------->
In extensions.conf
; from-digital:
;
; Context to set for PRI's and equivalent
;
[from-digital]
exten => s,1,Playtones(ring)
include => from-digital-custom
include => from-pstn
;-------------------------------------------------------------------------------
<----------------------------------------------------->
In extensions_additional.conf
[from-trunk-sip-nexVortex]
include => from-trunk-sip-nexVortex-custom
exten => _.,1,Set(GROUP()=OUT_19)
exten => _.,n,Goto(from-trunk,${EXTEN},1)
;--== end of [from-trunk-sip-nexVortex] ==--;
[ext-did-0002]
include => ext-did-0002-custom <= Added this to try to add Playtopnes(ring) to this entire context (see below), but it does not work
exten => fax,1,Goto(${CUT(FAX_DEST,^,1)},${CUT(FAX_DEST,^,2)},${CUT(FAX_DEST,^,3)})
<snip>
exten => 8922,1,Set(__FROM_DID=${EXTEN})
exten => 8922,n,Gosub(app-blacklist-check,s,1())
exten => 8922,n,Set(CDR(did)=${FROM_DID})
exten => 8922,n,ExecIf($[ "${CALLERID(name)}" = "" ] ?Set(CALLERID(name)=${CALLERID(num)}))
exten => 8922,n,Set(CHANNEL(musicclass)=default)
exten => 8922,n,Set(__MOHCLASS=default)
exten => 8922,n,Set(__CALLINGPRES_SV=${CALLERPRES()})
exten => 8922,n,Set(CALLERPRES()=allowed_not_screened)
exten => 8922,n,Playtones(ring) <= Manually added this, it works, but FreePBX will of course overwrite
exten => 8922,n(dest-ext),Goto(from-did-direct,126,1)
<snip>
;--== end of [ext-did-0002] ==--;
[ext-did]
include => ext-did-custom
include => ext-did-0001
include => ext-did-0002
exten => foo,1,Noop(bar)
;--== end of [ext-did] ==--;
<----------------------------------------------------->
Calls come in via DHADI in context [from-digital]
[from-digital]
exten => s,1,Playtones(ring) <= Tried it here also, but this does not see to hav and effect
include => from-digital-custom
include => from-pstn
[from-pstn]
include => from-pstn-custom ; create this context in extensions_custom.conf to include customizations
include => ext-did
include => ext-did-post-custom
include => from-did-direct
include => ext-did-catchall ; THIS MUST COME AFTER ext-did
<----------------------------------------------------->
In extensions_custom.conf, tried adding these
[ext-did-0002-custom]
exten => s,1,Playtones(ring)
;# // END ext-did-0002-custom
[ext-did-custom]
exten => s,1,Playtones(ring)
;# // END ext-did-custom
Thanks again for taking a look at this - its driving me crazy!