No Outbound call - Solved

[geb]

Hi All

This is my first real post

I'm stuck at the dialplan section of the owners manual
I need help to get past this section
The more I read up about this issue the more changes I make and the further away a solution appears to be.
I would prefer an integration between vtiger 5.4 and vicidial but nobody seams to be able to do it
I would like the better sms integration etc
I've done a cli at the bottom of this page on a range of numbers to see if I can get through


Current status

I Can make internal calls and
I can get external calls in with cid
but cant make external calls out

I am disappointed to report that I paid poundteam to do a 5.4 Vtiger Install integration but they gave up -
So now I have to go back to Vtiger 5.2 and going to have to do this myself- with your help of course
I would really prefer to get a hotshot to take this problem away from me in exchange for money

[geb]

Specs
Version: 2.4b0.5
Vicibox X.X from .iso
Vicidial 2.4-309a Build 110430 -1642 |
Kernel Version 2.6.18-238.9.1.el5.goPAE (SMP)
Distro Name
GoAutoDial CE 2.1/ GoAutoDial CE 2.0
Asterisk X.X.X | Single Server | Grandstream Hardware | No Extra Software After Installation |
Processors 2
Model Celeron(R) Dual-Core CPU T3500 @ 2.10GHz
CPU Speed 2.09 GHz
Asterisk 1.4.39.1-vici
======================================================================
PHONES WITHIN THIS SERVER:
EXTENSION NAME ACTIVE
8001 sip8001 Y
7001 7001 Y
8002 8002 Y
8003 8003 Y
8004 8004 Y
8005 8005 Y
======================================================================

SIP_generic
type=friend
host=dynamic
canreinvite=no
context=default

PHONES USING THIS CONF TEMPLATE:
EXTENSION NAME SERVER ACTIVE
8001 sip8001 xxx.xxx.x.35 Y

[geb]

7001 7001 Y
8002 8002 Y
8003 8003 Y
8004 8004 Y
8005 8005 Y
======================================================================

SIP_generic
type=friend
host=dynamic
canreinvite=no
context=default

PHONES USING THIS CONF TEMPLATE:
EXTENSION NAME SERVER ACTIVE
8001 sip8001 xxxxxxxxxxxx Y

======================================================================

CARRIERS WITHIN THIS SERVER:
CARRIER ID NAME REGISTRATION ACTIVE
PARAXIP TEST ParaXip CPD example N
SIPEXAMPLE TEST SIP carrier example register => testcarrier:test@xx.xx.xx.15:5060 N
IAXEXAMPLE TEST IAX carrier example register => testcarrier:test@xx.xx.xx.15:4569 N
GoAutoDial Goautodial SIP Account Template register => username:password@sip2.gxxxxxxxxxxxl.xxx:5060/username N
grandstream grandstream register => 1234:1234@xxx.xxx.x.160:5060/1234 Y

[geb]

Carrier

register => 1234:1234@xxx.xxx.x.160:5060/1234

[grandstream]
disallow=all
allow=g729
allow=gsm
allow=ulaw
type=friend
username=1234
secret=1234
host=xxx.xxx.x.160
dtmfmode=rfc2833
context=trunkinbound
qualify=yes
insecure=very
nat=yes



exten => _927XXXXXXXXXX,1,AGI(agi://xxx.xx.xx.1:4577/call_log)
exten => _927XXXXXXXXXX,2,Dial(SIP${EXTEN:3}@goautodial,,tTo)
exten => _927XXXXXXXXXX,3,Hangup

[geb]

======================================================================
extensions.conf


[general]
static=yes
writeprotect=no

[globals]
CONSOLE=Console/dsp ; Console interface for demo
;TRUNK=Zap/r1 ; Trunk interface
;TRUNKX=Zap/r2 ; 2nd trunk interface
TRUNKIAX=IAX2/ASTtest1:test@:4569 ; IAX trunk interface
TRUNKIAX1=IAX2/ASTtest1:test@:4569 ; IAX trunk interface
;TRUNKBINFONE=IAX2/1112223333:PASSWORD@xxxxxxxxxx ; IAX trunk interface
;SIPtrunk=SIP/1234:PASSWORD@xxxxxxxxxxx ; SIP trunk
TRUNKXYZ=SIP/gxw4108
#include extensions-vicidial.conf

[trunkinbound]
; DID call routing process
exten => _X.,1,AGI(agi-DID_route.agi)

; FastAGI for VICIDIAL/astGUIclient call logging
exten => h,1,DeadAGI(agi://xxx.xx.xx.1:4577/call_log--HVcaus ... EBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})


[default]
include => vicidial-auto
exten => _1234,1,AGI(agi-DID_route.agi)
exten => _9042XXXX.,1,AGI(agi://xxx.xx.xx.1:4577/call_log)
exten => _9042XXXX.,2,Dial(${TRUNKXYZ}/${EXTEN:4},,tTo)
exten => _9042XXXX.,3,Hangup

[incoming]
exten => s,1,NoOp(Call from MySipProvider)
exten => s,n,Dial(SIP/8002&SIP/8003&SIP/8004, 20)

[geb]

sip.conf

[general]
context=default ; Default context for incoming calls
;allowguest=no ; Allow or reject guest calls (default is yes)
allowoverlap=no ; Disable overlap dialing support. (Default is yes)
;allowtransfer=no ; Disable all transfers (unless enabled in peers or users)
;realm=mydomain.tld ; Realm for digest authentication
bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=xxx.xxx.xx.x ; IP address to bind to (xxx.xxx.xx.xbinds to all)
srvlookup=yes


[1234]
disallow=all
allow=all
username=1234
secret=1234
host=xxx.xxx.x.35
qualify=yes
dtmfmode=rfc2833
type=friend
context=trunkinbound
insecure=invite
insecure=very
canreinvite=no


[gw]
type=friend
disallow=all
allow=ulaw
allow=alaw
allow=g729
host=xxx.xxx.x.160
context=default
insecure=port
insecure=invite
insecure=very
dtmfmode=rfc2833

[gw]
type=peer
qualify=yes
insecure=port,invite
disallow=all
allow=ulaw
context=incoming

[gw]
type=peer
secret=1234
username=8001
fromuser=8001
fromdomain=xxx.xxx.x.35
host=xxx.xxx.x.35
outboundproxy=xxx.xxx.x.35
nat=yes
context=incoming
caninvite=no ; Appears to be required for outgoing audio
canreinvite=no ; Appears to be required for outgoing audio
disallow=all
allow=ulaw,alaw

allowexternaldomains=yes

[geb]

manager.conf

[general]
enabled = yes
port = 5038
bindaddr = 0.0.0.0

[admin]
secret =goautodial
deny=0.0.0.0/0.0.0.0
permit=xxx.xxx.2.9/255.255.255.0
read=system,call,log,verbose,command,agent,user
write=system,call,log,verbose,command,agent,user

[8001]
secret =goautodial
deny=0.0.0.0/0.0.0.0
permit=xxx.xxx.2.9/255.255.255.0
read=system,call,log,verbose,command,agent,user
write=system,call,log,verbose,command,agent,user

[cron]
secret = 1234
read = system,call,log,verbose,command,agent,user
write = system,call,log,verbose,command,agent,user

[updatecron]
secret = 1234
read = command
write = command

[listencron]
secret = 1234
read = system,call,log,verbose,command,agent,user
write = command

[sendcron]
secret = 1234
read = command
write = system,call,log,verbose,command,agent,user

[geb]

[Jul 16 16:20:59] VERBOSE[26820] logger.c: -- Reloading module 'chan_dahdi.so' (DAHDI Telephony w/PRI)
[Jul 16 16:20:59] ERROR[26820] chan_dahdi.c: Unable to load chan_dahdi.conf
[Jul 16 16:20:59] VERBOSE[26820] logger.c: -- Reloading module 'cdr_custom.so' (Customizable Comma Separated Values CDR Backend)
[Jul 16 16:20:59] VERBOSE[26820] logger.c: == Parsing '/etc/asterisk/cdr_custom.conf': [Jul 16 16:20:59] VERBOSE[26820] logger.c: Found
[Jul 16 16:20:59] VERBOSE[26820] logger.c: -- Reloading module 'app_amd.so' (Answering Machine Detection Application)
[Jul 16 16:20:59] VERBOSE[26820] logger.c: == Parsing '/etc/asterisk/amd.conf': [Jul 16 16:20:59] VERBOSE[26820] logger.c: Found
[Jul 16 16:20:59] VERBOSE[26820] logger.c: -- AMD defaults: initialSilence [2500] greeting [1500] afterGreetingSilence [800] totalAnalysisTime [5000] minimumWordLength [100] betweenWordsSilence [50] maximumNumberOfWords [3] silenceThreshold [256]
[Jul 16 16:20:59] VERBOSE[26820] logger.c: -- Reloading module 'codec_alaw.so' (A-law Coder/Decoder)
[Jul 16 16:20:59] VERBOSE[26820] logger.c: -- Reloading module 'pbx_dundi.so' (Distributed Universal Number Discovery (DUNDi))
[Jul 16 16:20:59] VERBOSE[26820] logger.c: == Parsing '/etc/asterisk/dundi.conf': [Jul 16 16:20:59] VERBOSE[26820] logger.c: Found
[Jul 16 16:20:59] DEBUG[26820] pbx_dundi.c: Seeding global EID 'b8:70:f4:87:d0:b8' from 'eth0' using 'siocgifhwaddr'
[Jul 16 16:20:59] VERBOSE[26820] logger.c: -- Reloading module 'res_crypto.so' (Cryptographic Digital Signatures)
[Jul 16 16:20:59] VERBOSE[26820] logger.c: -- Reloading module 'chan_agent.so' (Agent Proxy Channel)
[Jul 16 16:20:59] VERBOSE[26820] logger.c: == Parsing '/etc/asterisk/agents.conf': [Jul 16 16:20:59] VERBOSE[26820] logger.c: Found
[Jul 16 16:20:59] VERBOSE[26820] logger.c: -- Reloading module 'cdr_manager.so' (Asterisk Manager Interface CDR Backend)
[Jul 16 16:20:59] VERBOSE[26820] logger.c: == Parsing '/etc/asterisk/cdr_manager.conf': [Jul 16 16:20:59] VERBOSE[26820] logger.c: Found
[Jul 16 16:21:01] VERBOSE[26854] logger.c: == Parsing '/etc/asterisk/manager.conf': [Jul 16 16:21:01] VERBOSE[26854] logger.c: Found
[Jul 16 16:21:01] VERBOSE[26854] logger.c: == Manager 'sendcron' logged on from xxx.xx.xx.1
[Jul 16 16:21:01] VERBOSE[26855] logger.c: == Parsing '/etc/asterisk/manager.conf': [Jul 16 16:21:01] VERBOSE[26855] logger.c: Found
[Jul 16 16:21:01] VERBOSE[26855] logger.c: == Manager 'sendcron' logged on from xxx.xx.xx.1
[Jul 16 16:21:01] VERBOSE[26854] logger.c: == Manager 'sendcron' logged off from xxx.xx.xx.1
[Jul 16 16:21:02] VERBOSE[26820] logger.c: == Manager 'cron' logged off from xxx.xx.xx.1
[Jul 16 16:21:03] VERBOSE[26855] logger.c: == Manager 'sendcron' logged off from xxx.xx.xx.1
[Jul 16 16:21:06] VERBOSE[27259] logger.c: == Parsing '/etc/asterisk/manager.conf': [Jul 16 16:21:06] VERBOSE[27259] logger.c: Found
[Jul 16 16:21:06] VERBOSE[27259] logger.c: == Manager 'sendcron' logged on from xxx.xx.xx.1
[Jul 16 16:21:06] VERBOSE[27259] logger.c: == Manager 'sendcron' logged off from xxx.xx.xx.1
[Jul 16 16:21:37] NOTICE[3251] chan_sip.c: Call from '8001' to extension '9270218763355' rejected because extension not found.
[Jul 16 16:21:46] NOTICE[3251] chan_sip.c: Call from '8001' to extension '0218763355' rejected because extension not found.
[Jul 16 16:21:59] VERBOSE[27466] logger.c: -- Executing [8001@default:1] Dial("SIP/8001-0000001c", "SIP/8001|60|") in new stack
[Jul 16 16:21:59] VERBOSE[27466] logger.c: -- Called 8001
[Jul 16 16:22:01] VERBOSE[27466] logger.c: -- SIP/8001-0000001d is ringing
[Jul 16 16:22:01] VERBOSE[27499] logger.c: == Parsing '/etc/asterisk/manager.conf': [Jul 16 16:22:01] VERBOSE[27499] logger.c: Found
[Jul 16 16:22:01] VERBOSE[27499] logger.c: == Manager 'sendcron' logged on from xxx.xx.xx.1
[Jul 16 16:22:01] VERBOSE[27516] logger.c: == Parsing '/etc/asterisk/manager.conf': [Jul 16 16:22:01] VERBOSE[27516] logger.c: Found
[Jul 16 16:22:01] VERBOSE[27516] logger.c: == Manager 'sendcron' logged on from xxx.xx.xx.1
[Jul 16 16:22:01] VERBOSE[27516] logger.c: == Manager 'sendcron' logged off from xxx.xx.xx.1
[Jul 16 16:22:03] VERBOSE[27499] logger.c: == Manager 'sendcron' logged off from xxx.xx.xx.1
[Jul 16 16:22:06] VERBOSE[27948] logger.c: == Parsing '/etc/asterisk/manager.conf': [Jul 16 16:22:06] VERBOSE[27948] logger.c: Found
[Jul 16 16:22:06] VERBOSE[27948] logger.c: == Manager 'sendcron' logged on from xxx.xx.xx.1
[Jul 16 16:22:06] VERBOSE[27948] logger.c: == Manager 'sendcron' logged off from xxx.xx.xx.1
[Jul 16 16:22:10] VERBOSE[3251] logger.c: -- Got SIP response 486 "Busy Here" back from 192.168.2.5
[Jul 16 16:22:10] VERBOSE[27466] logger.c: -- SIP/8001-0000001d is busy
[Jul 16 16:22:10] VERBOSE[27466] logger.c: == Everyone is busy/congested at this time (1:1/0/0)
[Jul 16 16:22:10] VERBOSE[27466] logger.c: -- Executing [8001@default:2] Goto("SIP/8001-0000001c", "default|850266666666668001|1") in new stack
[Jul 16 16:22:10] VERBOSE[27466] logger.c: -- Goto (default,850266666666668001,1)
[Jul 16 16:22:10] VERBOSE[27466] logger.c: -- Executing [850266666666668001@default:1] Wait("SIP/8001-0000001c", "1") in new stack
[Jul 16 16:22:11] VERBOSE[27466] logger.c: -- Executing [850266666666668001@default:2] VoiceMail("SIP/8001-0000001c", "8001|u") in new stack
[Jul 16 16:22:11] VERBOSE[27466] logger.c: -- Playing 'vm-theperson' (language 'en')
[Jul 16 16:22:13] VERBOSE[27466] logger.c: -- Playing 'digits/8' (language 'en')
[Jul 16 16:22:13] VERBOSE[27466] logger.c: -- Playing 'digits/0' (language 'en')
[Jul 16 16:22:14] VERBOSE[27466] logger.c: == Spawn extension (default, 850266666666668001, 2) exited non-zero on 'SIP/8001-0000001c'
[Jul 16 16:22:14] VERBOSE[27466] logger.c: -- Executing [h@default:1] DeadAGI("SIP/8001-0000001c", "agi://xxx.xx.xx.1:4577/call_log--HVcauses--PRI-----NODEBUG-----17-----BUSY----------") in new stack
[Jul 16 16:22:14] VERBOSE[27466] logger.c: -- AGI Script agi://xxx.xx.xx.1:4577/call_log--HVcaus ... ---------- completed, returning 0
[Jul 16 16:22:30] NOTICE[3251] chan_sip.c: Call from '8001' to extension '910218763355' rejected because extension not found.
[Jul 16 16:22:53] NOTICE[3251] chan_sip.c: Call from '8001' to extension '9270218763499' rejected because extension not found.

[geb]

I've been circling this vicidial vtiger development for some time now and now I'm ready to commit
Hope I did not include too much stuff but its better than you asking me for more
Thanks in advance

[DomeDan]

Good post!

You have named the carrier [grandstream]
but in the dialplan you tell it to use @goautodial
fix that

and after "SIP" you need a slash: "SIP/${EXTEN:3}"

[geb]

Many thanks for the reply

Striker changed the configs as follows and it was able to make a outgoing call as well
However a short while later it stopped making outbound calls again so Im back to square one
incoming good with caller id but outgoing fails after two rings new cli at end of post
=====================================================================================

[geb]

THe carrier section now seems to have nothing enabled / active ie
I wonder if this could be the problem?

GoAutoDial Goautodial SIP Account Template 192-168-2-35 SIP register => username:passwordatsip2.goautodial-com:5060/username N MODIFY

grandstream grandstream 192-168-2-35 SIP N MODIFY ( i though this one should be active)

IAXEXAMPLE TEST IAX carrier example 192-168.2-35 IAX2 register => testcarrier:testatten-10-10-15:4569 N MODIFY

PARAXIP TEST ParaXip CPD example 192-168-2-35 SIP N MODIFY

SIPEXAMPLE TEST SIP carrier example 192-168-2-35 SIP register => testcarrier:testatten-10-10-15:5060 N MODIFY

==================================================================================
eturning 0

[geb]

=======================================================================================
this is config of the carrier grandstream ( which I though should have been active

[grandstream]
disallow=all
allow=all
type=peer
host=192-168-2-160
dtmfmode=rfc2833
context=trunkinbound

Here is the dial plan for that carrier grandstream

exten => _927XXXXXXXXXX,1,AGI(agi://127-0-0-1:4577/call_log)
exten => _927XXXXXXXXXX,2,Dial(${SIPGRAND}/${EXTEN:3},,tTor)
exten => _927XXXXXXXXXX,3,Hangup

exten => _9044XXXXX.,1,AGI(agi://127-0-0-1:4577/call_log)
exten => _9044XXXXX.,2,Dial(SIP/222/${EXTEN:4},,tTor)
exten => _9044XXXXX.,3,Hangup
qualify=yes
insecure=port
insecure=invite

================================================================================

[geb]

sip.conf is below

[general]
context=default ; Default context for incoming calls
;allowguest=no ; Allow or reject guest calls (default is yes)
allowoverlap=no ; Disable overlap dialing support. (Default is yes)
;allowtransfer=no ; Disable all transfers (unless enabled in peers or users)
;realm=mydomain.tld ; Realm for digest authentication
bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=xxx.xxx.xx.x ; IP address to bind to (xxx.xxx.xx.xbinds to all)
srvlookup=yes

[geb]

; Enable DNS SRV lookups on outbound calls
;domain=mydomain.tld ; Set default domain for this host
;pedantic=yes ; Enable checking of tags in headers,
;tos_sip=cs3 ; Sets TOS for SIP packets.
;tos_audio=ef ; Sets TOS for RTP audio packets.
;tos_video=af41 ; Sets TOS for RTP video packets.
;maxexpiry=3600 ; Maximum allowed time of incoming registrations
;minexpiry=60 ; Minimum length of registrations/subscriptions (default 60)
;defaultexpiry=120 ; Default length of incoming/outgoing registration
;t1min=100 ; Minimum roundtrip time for messages to monitored hosts
;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
;checkmwi=10 ; Default time between mailbox checks for peers
;buggymwi=no ; Cisco SIP firmware doesn't support the MWI RFC
;vmexten=voicemail ; dialplan extension to reach mailbox sets the
disallow=all ; First disallow all codecs
allow=ulaw ; Allow codecs in order of preference
allow=gsm
mohinterpret=default
mohsuggest=default
language=en ; Default language setting for all users/peers

[geb]

relaxdtmf=yes ; Relax dtmf handling
trustrpid = no ; If Remote-Party-ID should be trusted
sendrpid = yes ; If Remote-Party-ID should be sent
progressinband=no ; If we should generate in-band ringing always
;useragent=Asterisk PBX ; Allows you to change the user agent string
;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address
;usereqphone = no ; If yes, ";user=phone" is added to uri that contains
dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833
;compactheaders = yes ; send compact sip headers.
videosupport=no ; Turn on support for SIP video. You need to turn this on
;maxcallbitrate=384 ; Maximum bitrate for video calls (default 384 kb/s)
callevents=yes ; generate manager events when sip ua
;alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected,
;g726nonstandard = yes ; If the peer negotiates G726-32 audio, use AAL2 packing
;matchexterniplocally = yes ; Only substitute the externip or externhost setting if it matches
;regcontext=sipregistrations
;rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity
;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity
rtpkeepalive=60 ; Send keepalives in the RTP stream to keep NAT open
;sipdebug = yes ; Turn on SIP debugging by default, from
;recordhistory=yes ; Record SIP history by default
;dumphistory=yes ; Dump SIP history at end of SIP dialogue
;allowsubscribe=no ; Disable support for subscriptions. (Default is yes)
;subscribecontext = default ; Set a specific context for SUBSCRIBE requests
notifyringing = yes ; Notify subscriptions on RINGING state (default: no)
notifyhold = yes ; Notify subscriptions on HOLD state (default: no)
limitonpeers = yes ; Apply call limits on peers only. This will improve

[geb]

ocalnet=192-168-0-0/255.255.0.0; All RFC 1918 addresses are local networks
localnet=10.0.0.0/255.0.0.0 ; Also RFC1918
localnet=172-16-0-0/12 ; Another RFC1918 with CIDR notation
localnet=169-254-0-0/255.255.0.0 ;Zero conf local network
nat=yes ; Global NAT settings (Affects all peers and users)
canreinvite=no ; Asterisk by default tries to redirect the
jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP
jbmaxsize = 100 ; Max length of the jitterbuffer in milliseconds.
jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP
jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
qualify=yes ; By default, qualify all peers at 2000ms
limitonpeer = yes ; enable call limit on a per peer basis, different from limitonpeers

#include sip-vicidial.conf

[geb]

[222]
type=friend
host=dynamic
;username=222
secret=222
context=trunkinbound
disallow=all
allow=all
insecure=invite
insecure=port
insecure=very
dtmfmode=rfc2833
[gxw_trunk]
host=192-168-2-160
type=friend
context=trunkinbound
disallow=all
allow=all
insecure=invite
insecure=port
dtmfmode=rfc2833
fromdomain=192-168-2-160
=================================================================================

[geb]

The following is the cli prior to activating the grandstream carrier above

I cant understand where is ip comes from
" failed for '192.168.2.159' - No matching peer found"

== Manager 'sendcron' logged off from 127-0-0-1
-- Executing [90440614008290atdefault:1] AGI("SIP/8001-00000002", "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 [90440614008290atdefault:2] Dial("SIP/8001-00000002", "SIP/222/0 614008290||tTor") in new stack
-- Called 222/0614008290
-- Got SIP response 503 "Service Unavailable" back from 192-168-2-160
-- SIP/222-00000003 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [90440614008290atdefault:3] Hangup("SIP/8001-00000002", "") in n ew stack
== Spawn extension (default, 90440614008290, 3) exited non-zero on 'SIP/8001-0 0000002'
-- Executing [hatdefault:1] DeadAGI("SIP/8001-00000002", "agi://127-0-0-1:457 7/call_log--HVcauses--PRI-----NODEBUG-----34-----CONGESTION----------") in new s tack
-- AGI Script agi://127-0-0-1:4577/call_log--HVcauses ... EBUG-----3 4-----CONGESTION---------- completed, returning 0
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127-0-0-1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127-0-0-1
== Manager 'sendcron' logged off from 127-0-0-1
== Manager 'sendcron' logged off from 127-0-0-1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127-0-0-1
== Manager 'sendcron' logged off from 127-0-0-1
== Refreshing DNS lookups.
-- Executing [222attrunkinbound:1] Goto("SIP/gxw_trunk-00000004", "incoming|s |1") in new stack

[geb]

Striker changed the carrier name to 222 and after changing grandstream cid settings it worked - in and out

My config files are currently looking like they soiled themselves as there is so much un-commented stuff in there that it looks like the last failed mars mission- before the US metric conversion.
However getting back to earth

After another wonderful remote " session" and getting the outbound working a short while later outgoing stopped working again . I am becoming increasingly freaked out
I rebooted grandstream and goauatodial - still not working
As mentioned above the carrier was showing non active ????

Its very frustrating as quick reply posts are prevented " too spamy " by the board
By definition config files contain "too many outside urls"
I suppose its something I will have to accept
I get really worried about deploying onto this platform as I cant even get a simple dialout to stay working
Imagine 3000 agents sitting around while I tinker with these config files

Suddenly the Sap and MSDynamics does not look that expensive after all

[DomeDan]

You need to enable the carrier "Active = Y"


this looks a little weird:
[grandstream]
disallow=all
allow=all

try to set it to this instead:
disallow=all
allow=gsm
allow=ulaw
allow=alaw


what is this?:
exten => _9044XXXXX.,1,AGI(agi://127-0-0-1:4577/call_log)
exten => _9044XXXXX.,2,Dial(SIP/222/${EXTEN:4},,tTor)
exten => _9044XXXXX.,3,Hangup

and this stuff:
qualify=yes
insecure=port
insecure=invite
should not be in the dialplan they should be in the "account entry"

[geb]

sip.cong new

[general]
context=default ; Default context for incoming calls
;allowguest=no ; Allow or reject guest calls (default is yes)
allowoverlap=no ; Disable overlap dialing support. (Default is yes)
;allowtransfer=no ; Disable all transfers (unless enabled in peers or users)
;realm=mydomain.tld ; Realm for digest authentication
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)
srvlookup=yes
disallow=all ; First disallow all codecs
allow=ulaw ; Allow codecs in order of preference
allow=gsm

mohinterpret=default
mohsuggest=default
language=en ; Default language setting for all users/peers
relaxdtmf=yes ; Relax dtmf handling
trustrpid = no ; If Remote-Party-ID should be trusted
sendrpid = yes ; If Remote-Party-ID should be sent
progressinband=no

[222]
type=friend
host=dynamic
;username=222
secret=222
context=trunkinbound
disallow=all
allow=gsm
allow=ulaw
allow=alaw
qualify=yes
insecure=invite
insecure=port
insecure=invite
insecure=very
dtmfmode=rfc2833

[gxw_trunk]
host=192.168.2.160
type=friend
context=trunkinbound
disallow=all
allow=gsm
allow=ulaw
allow=alaw
qualify=yes
insecure=invite
insecure=port
dtmfmode=rfc2833
fromdomain=192.168.2.160

[geb]

carrier still enabled ie

[grandstream]
disallow=all
allow=gsm
allow=ulaw
allow=alaw
type=peer
host=192.168.2.160
dtmfmode=rfc2833
context=trunkinbound

The following three lines - should I delete them or ?

exten => _927XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _927XXXXXXXXXX,2,Dial(${SIPGRAND}/${EXTEN:3},,tTor)
exten => _927XXXXXXXXXX,3,Hangup

What should I do about the following three lines ?

exten => _9044XXXXX.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9044XXXXX.,2,Dial(SIP/222/${EXTEN:4},,tTor)
exten => _9044XXXXX.,3,Hangup

[geb]

i now tried both dial strings 927061etc and 9044061etc

cli follows



-- Registered extension context 'monitor_exit'
-- Added extension 'h' priority 1 to monitor_exit
-- Added extension '_X' priority 1 to monitor_exit
-- Added extension 'i' priority 1 to monitor_exit
-- Added extension '#' priority 1 to monitor_exit
-- Added extension 't' priority 1 to monitor_exit
-- Registered extension context 'incoming'
-- Added extension 's' priority 1 to incoming
-- Added extension 's' priority 2 to incoming
-- Reloading module 'app_voicemail.so' (Comedian Mail (Voicemail System))
== Parsing '/etc/asterisk/voicemail.conf': Found
-- Reloading module 'chan_mgcp.so' (Media Gateway Control Protocol (MGCP))
Reloading MGCP
== Parsing '/etc/asterisk/mgcp.conf': Found
== MGCP Listening on 0.0.0.0:2727
== Using TOS bits 0
-- Reloading module 'codec_dahdi.so' (Generic DAHDI Transcoder Codec Translator)
-- Reloading module 'codec_ulaw.so' (mu-Law Coder/Decoder)
-- Reloading module 'codec_g726.so' (ITU G.726-32kbps G726 Transcoder)
-- Reloading module 'res_indications.so' (Indications Resource)
-- Unregistered indication country 'at'
-- Unregistered indication country 'au'
-- Unregistered indication country 'br'
-- Unregistered indication country 'be'
-- Unregistered indication country 'ch'
-- Unregistered indication country 'cl'
-- Unregistered indication country 'cn'
-- Unregistered indication country 'cz'
-- Unregistered indication country 'de'
-- Unregistered indication country 'dk'
-- Unregistered indication country 'ee'
-- Unregistered indication country 'es'
-- Unregistered indication country 'fi'
-- Unregistered indication country 'fr'
-- Unregistered indication country 'gr'
-- Unregistered indication country 'hu'
-- Unregistered indication country 'it'
-- Unregistered indication country 'lt'
-- Unregistered indication country 'mx'
-- Unregistered indication country 'nl'
-- Unregistered indication country 'no'
-- Unregistered indication country 'nz'
-- Registered IAX2 to '127.0.0.1', who sees us as 127.0.0.1:36631 with no messages waiting

-- Unregistered indication country 'pl'
-- Unregistered indication country 'pt'
-- Unregistered indication country 'ru'
-- Unregistered indication country 'se'
-- Unregistered indication country 'sg'
-- Unregistered indication country 'uk'
[Jul 19 08:28:00] NOTICE[21646]: indications.c:502 ast_unregister_indication_country: Removed default indication country 'us'
-- Unregistered indication country 'us'
-- Unregistered indication country 'us-o'
-- Unregistered indication country 'tw'
-- Unregistered indication country 'za'
== Parsing '/etc/asterisk/indications.conf': Found
-- Registered indication country 'at'
-- Registered indication country 'au'
-- Registered indication country 'br'
-- Registered indication country 'be'
-- Registered indication country 'ch'
-- Registered indication country 'cl'
-- Registered indication country 'cn'
-- Registered indication country 'cz'
-- Registered indication country 'de'
-- Registered indication country 'dk'
-- Registered indication country 'ee'
-- Registered indication country 'es'
-- Registered indication country 'fi'
-- Registered indication country 'fr'
-- Registered indication country 'gr'
-- Registered indication country 'hu'
-- Registered indication country 'it'
-- Registered indication country 'lt'
-- Registered indication country 'mx'
-- Registered indication country 'nl'
-- Registered indication country 'no'
-- Registered indication country 'nz'
-- Registered indication country 'pl'
-- Registered indication country 'pt'
-- Registered indication country 'ru'
-- Registered indication country 'se'
-- Registered indication country 'sg'
-- Registered indication country 'uk'
-- Registered indication country 'us'
-- Registered indication country 'us-o'
-- Registered indication country 'tw'
-- Registered indication country 'za'
-- Setting default indication country to 'us'
-- Reloading module 'codec_adpcm.so' (Adaptive Differential PCM Coder/Decoder)
-- Reloading module 'pbx_ael.so' (Asterisk Extension Language Compiler)
[Jul 19 08:28:00] NOTICE[21646]: pbx_ael.c:4516 pbx_load_module: Starting AEL load process.
[Jul 19 08:28:00] NOTICE[21646]: pbx_ael.c:4523 pbx_load_module: AEL load process: calculated config file name '/etc/asterisk/extensions.ael'.
[Jul 19 08:28:00] NOTICE[21646]: pbx_ael.c:4531 pbx_load_module: AEL load process: parsed config file name '/etc/asterisk/extensions.ael'.
[Jul 19 08:28:00] NOTICE[21646]: pbx_ael.c:4534 pbx_load_module: AEL load process: checked config file name '/etc/asterisk/extensions.ael'.
-- Registered extension context 'macro-std-exten-ael'
-- Registered extension context 'ael-demo'
-- Added extension 's' priority 1 to macro-std-exten-ael
-- Added extension 's' priority 2 to macro-std-exten-ael
-- Added extension 's' priority 3 to macro-std-exten-ael
-- Added extension 's' priority 4 to macro-std-exten-ael
-- Added extension 's' priority 5 to macro-std-exten-ael
-- Added extension 's' priority 6 to macro-std-exten-ael
-- Added extension 'a' priority 1 to macro-std-exten-ael
-- Added extension 'a' priority 2 to macro-std-exten-ael
-- Added extension 'a' priority 3 to macro-std-exten-ael
-- Added extension '_sw-7-.' priority 10 to macro-std-exten-ael
-- Added extension '_sw-7-.' priority 11 to macro-std-exten-ael
-- Added extension 'sw-7-' priority 10 to macro-std-exten-ael
-- Added extension 'sw-7-BUSY' priority 10 to macro-std-exten-ael
-- Added extension 'sw-7-BUSY' priority 11 to macro-std-exten-ael
-- Added extension 's' priority 1 to ael-demo
-- Added extension 's' priority 2 to ael-demo
-- Added extension 's' priority 3 to ael-demo
-- Added extension 's' priority 4 to ael-demo
-- Added extension 's' priority 5 to ael-demo
-- Added extension 's' priority 6 to ael-demo
-- Added extension 's' priority 7 to ael-demo
-- Added extension 's' priority 8 to ael-demo
-- Added extension 's' priority 9 to ael-demo
-- Added extension 's' priority 10 to ael-demo
-- Added extension 's' priority 11 to ael-demo
-- Added extension 's' priority 12 to ael-demo
-- Added extension '2' priority 1 to ael-demo
-- Added extension '2' priority 2 to ael-demo
-- Added extension '3' priority 1 to ael-demo
-- Added extension '3' priority 2 to ael-demo
-- Added extension '500' priority 1 to ael-demo
-- Added extension '500' priority 2 to ael-demo
-- Added extension '500' priority 3 to ael-demo
-- Added extension '500' priority 4 to ael-demo
-- Added extension '600' priority 1 to ael-demo
-- Added extension '600' priority 2 to ael-demo
-- Added extension '600' priority 3 to ael-demo
-- Added extension '600' priority 4 to ael-demo
-- Added extension '_1234' priority 1 to ael-demo
-- Added extension '#' priority 1 to ael-demo
-- Added extension '#' priority 2 to ael-demo
-- Added extension 't' priority 1 to ael-demo
-- Added extension 'i' priority 1 to ael-demo
[Jul 19 08:28:00] NOTICE[21646]: pbx_ael.c:4540 pbx_load_module: AEL load process: compiled config file name '/etc/asterisk/extensions.ael'.
[Jul 19 08:28:00] NOTICE[21646]: pbx_ael.c:4543 pbx_load_module: AEL load process: merged config file name '/etc/asterisk/extensions.ael'.
[Jul 19 08:28:00] NOTICE[21646]: pbx_ael.c:4546 pbx_load_module: AEL load process: verified config file name '/etc/asterisk/extensions.ael'.
-- Reloading module 'app_queue.so' (True Call Queueing)
== Parsing '/etc/asterisk/queues.conf': Found
-- Reloading module 'chan_sip.so' (Session Initiation Protocol (SIP))
Reloading SIP
== Parsing '/etc/asterisk/sip.conf': Found
== Parsing '/etc/asterisk/sip-vicidial.conf': Found
== Parsing '/etc/asterisk/sip_notify.conf': Found
-- Reloading module 'codec_gsm.so' (GSM Coder/Decoder)
-- Reloading module 'app_meetme.so' (MeetMe conference bridge)
== Parsing '/etc/asterisk/meetme.conf': Found
== Parsing '/etc/asterisk/meetme-vicidial.conf': Found
-- Reloading module 'chan_dahdi.so' (DAHDI Telephony w/PRI)
[Jul 19 08:28:00] ERROR[21646]: chan_dahdi.c:14066 setup_dahdi: Unable to load chan_dahdi.conf
-- Reloading module 'cdr_custom.so' (Customizable Comma Separated Values CDR Backend)
== Parsing '/etc/asterisk/cdr_custom.conf': Found
-- Reloading module 'app_amd.so' (Answering Machine Detection Application)
== Parsing '/etc/asterisk/amd.conf': Found
-- AMD defaults: initialSilence [2500] greeting [1500] afterGreetingSilence [800] totalAnalysisTime [5000] minimumWordLength [100] betweenWordsSilence [50] maximumNumberOfWords [3] silenceThreshold [256]
-- Reloading module 'codec_alaw.so' (A-law Coder/Decoder)
-- Reloading module 'pbx_dundi.so' (Distributed Universal Number Discovery (DUNDi))
== Parsing '/etc/asterisk/dundi.conf': Found
-- Reloading module 'res_crypto.so' (Cryptographic Digital Signatures)
-- Reloading module 'chan_agent.so' (Agent Proxy Channel)
== Parsing '/etc/asterisk/agents.conf': Found
-- Reloading module 'cdr_manager.so' (Asterisk Manager Interface CDR Backend)
== Parsing '/etc/asterisk/cdr_manager.conf': Found
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'cron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
-- Executing [90440614008290@default:1] AGI("SIP/8001-0000001a", "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 [90440614008290@default:2] Dial("SIP/8001-0000001a", "SIP/222/0614008290||tTor") in new stack
-- Called 222/0614008290
-- Got SIP response 503 "Service Unavailable" back from 192.168.2.160
-- SIP/222-0000001b is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [90440614008290@default:3] Hangup("SIP/8001-0000001a", "") in new stack
== Spawn extension (default, 90440614008290, 3) exited non-zero on 'SIP/8001-0000001a'
-- Executing [h@default:1] DeadAGI("SIP/8001-0000001a", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----34-----CONGESTION----------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Refreshing DNS lookups.
-- Executing [9270614008290@default:1] AGI("SIP/8001-0000001c", "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 [9270614008290@default:2] Dial("SIP/8001-0000001c", "SIP/grandstream/0614008290||tTor") in new stack
-- Called grandstream/0614008290
-- Got SIP response 503 "Service Unavailable" back from 192.168.2.160
-- SIP/grandstream-0000001d is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [9270614008290@default:3] Hangup("SIP/8001-0000001c", "") in new stack
== Spawn extension (default, 9270614008290, 3) exited non-zero on 'SIP/8001-0000001c'
-- Executing [h@default:1] DeadAGI("SIP/8001-0000001c", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----34-----CONGESTION----------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0

[DomeDan]

did you see my previous message?

Got SIP response 503 "Service Unavailable" back from 192.168.2.160
who is 192.168.2.160?

edit: wait a minute, you have set host=192.168.2.160
I doubt you got your provider on the local 192.168.2.1 network...
change that host line back to:
host=xxx.xxx.x.160

and dont "trial and error" change settings, read about the settings and understand what they are

[geb]

HiDomedan

Thanks for responding again

I changed all the settings in your earlier post that you recommended
The 192.168.2.160 is a grandstream 8 port fxo connecting to the pstn
The strange thing was that the settings above worked for a while and then stopped working
It must be something to do with the PSTN

Could it be this the reason

Jul 19 08:28:00] NOTICE[21646]: indications.c:502 ast_unregister_indication_country: Removed default indication country 'us'
-- Unregistered indication country 'us'

Jul 19 08:28:00] NOTICE[21646]: indications.c:502 ast_unregister_indication_country: Removed default indication country 'us'
-- Unregistered indication country 'us'
-- Unregistered indication country 'us-o'
-- Unregistered indication country 'tw'
-- Unregistered indication country 'za'

-- Registered indication country 'za'
-- Setting default indication country to 'us'
-- Reloading module 'codec_adpcm.so' (Adaptive Differential PCM Coder/Decoder)
-- Reloading module 'pbx_ael.so' (Asterisk Extension Language Compiler)

Ive changed the indications file to the following

[za]
description = South Africa
; xxxxxxxxxx/univercd/cc/td/doc/product/tel_pswt/vco_prod/safr_sup/saf02.htm
; (definitions for other countries can also be found there)
; Note, though, that South Africa uses two switch types in their network --
; Alcatel switches -- mainly in the Western Cape, and Siemens elsewhere.
; The former use 383+417 in dial, ringback etc. The latter use 400*33
; I've provided both, uncomment the ones you prefer
: GEB not sure what to do here
ringcadence = 400,200,400,2000
; dial/ring/callwaiting for the Siemens switches:
;dial = 400*33
;ring = 400*33/400,0/200,400*33/400,0/2000
; callwaiting = 400*33/250,0/250,400*33/250,0/250,400*33/250,0/250,400*33/250,0/250
; dial/ring/callwaiting for the Alcatel switches:
dial = 383+417
ring = 383+417/400,0/200,383+417/400,0/2000
callwaiting = 383+417/250,0/250,383+417/250,0/250,383+417/250,0/250,383+417/250,0/250

; Does the following three lines apply to both switches

congestion = 400/250,0/250
busy = 400/500,0/500
dialrecall = 350+440

; XXX Not sure about the RECORDTONE
record = 1400/500,0/10000
info = 950/330,1400/330,1800/330,0/330

; XXX Not sure about the stutter

stutter = !400*33/100,!0/100,!400*33/100,!0/100,!400*33/100,!0/100,!400*33/100,!0/100,!400*33/100,!0/100,!400*33/100,!0/100,400*33

[geb]

it also seems that the granmdstream in not hanging up properly and also calls are being diverted into a extension that is not working

[geb]

I’m pleased to report that Mohamed from Morocco , after logging into my Grandstream fixed my problem

He fixed my firmware update settings to latest ( not clear from manual) so I had them wrong
All I had to do was reboot the Grandstream and the lights flashed

I should have used , if you want to keep upgrading use "firmware.grandstream.com
the default path is fm.grandstream.com/gs don’t use this if you want to auto update
So Ill have to change this back to firmware.grandstream.com
He downloaded a tftp server which he decided to to use Tftpd32-4.00-setup
I suppose I can use this if I want local updates to take place from a local server
I had wait for dialtone set to Y – He changed it back to default N
He then prepared a line test as follows
He entered in the number below and then updated and then rebooted


He had started up wireshark which gave him in his words “ not very useful test results”
Not sure what he was looking for
Based on this He then dropped the following values from 480 to 400 ( from ITU manual )
He changed the on offs to match the ones below

South Africa (Republic of)
Busy tone - 400 0.5 on 0.5 off
Comfort tone - 950/950/1400 0.65 on 0.325 off 0.325 on 1.3 on 2.6 off
Congestion tone - 400 0.25 on 0.25 off
Dial tone - 400x33 continuous
Special dial tone - 400x33 0.25 on 0.25 off + continuous
Function acknowledge tone - 700/1100 0.25 on 0.25 off 0.25 on 0.25 off
Intrusion tone - 400 0.15 on 0.25 off 0.15 on 1.45 off
Notify tone - 900 0.2 on 0.2 off 0.2 on 0.2 off
Number unobtainable tone - 400 2.5 on 0.5 off
Ringing tone - 400x33 0.4 on 0.2 off 0.4 on 2.0 off
Special information tone - 950/1400/1800 3x0.33 on 1.0 off
Call waiting tone - 400x33 0.4 on 4.0 off


I was sure that he had a different on off ( no a 0/0) for dial tone. Perhaps it did not save
I now see that its 0/0 for continuous
It seemed that the save( update / reboot) had to take place line per line and you could not adjust all the settings and then save update and reboot once
He then tested hangup as I was not sure it worked I dialled in and after hangup it reset . I Could dial in again immediately
So after lots of frustration it was fixed in 15 min
I document this to save somebody else the hassle and moiney on consultants
With support like this Ill give the new Grandstream UCM6100 series IP PBX Appliance a go

[gardo]

Glad to see that it's working now. Thank you for posting your results.