ZOIPER message when dialing out

I got this message:

Sep 9 07:01:22] NOTICE[2323] chan_sip.c: Registration from ‘"oliver"sip:oliver@192.168.1.66;transport=UDP’ failed for ‘192.168.1.64’ - No matching peer found
[Sep 9 07:01:31] WARNING[2436] app_dial.c: Unable to create channel of type ‘SIP’ (cause 3 - No route to destination)
[Sep 9 07:01:31] WARNING[2436] app_dial.c: Unable to create channel of type ‘IAX2’ (cause 3 - No route to destination)
[Sep 9 07:01:52] NOTICE[2323] chan_sip.c: Registration from ‘"oliver"sip:oliver@192.168.1.66;transport=UDP’ failed for ‘192.168.1.64’ - No matching peer found

" [b]

I got this message No route to destination everytime I wanna use the PSTN or when I dial 9 plus the 7 digit number.
I setup my Dialplan already but error still exist.
Please help me.
Thanks in advance for your time.

[/b]

post the relevant portions of your dial plan and sip.conf

G2010,
Can you please elaborate your answer?
Thanks.

there is not nearly enough information in your original post to offer any substantial help.

what I am seeing in the error is:

1 - sip.conf is not properly configured to register “oliver” from 192.168.1.64

2 - you are trying to do a Dial(SIP/) to something which isn’t properly registered

3 - you are trying to do a Dial(IAX/) to something which isn’t properly registered

I can’t tell you what you are doing unless you share some of your dialplan configuration as well as your sip/iax config.

g2010,
I’m so glad I found you as my mentor on this *NOW. First of all THANK YOU SO MUCH. Secondly, I think I have to start from the beginning. I’m new to this *NOW but I know analog PABX. I have my *NOW up and running using generic card with one FXO and FXS. I tried reading books and forums but it seems they published the old versions. What I’m trying to do is I’m using ZOIPER and X-LITE for my extesions. I like ZOIPER because it works when accessing the VM but I can’t dial out using PSTN and I can’t call one of my extensions. If I tried dialing 9 plus 7 digit the ZOIPER says NO ROUTE DESTINATION. I already configured the Dialing rules, but maybe I’m wrong. When I look at the ASTERISK LOG this is what I found :

Sep 9 07:01:22] NOTICE[2323] chan_sip.c: Registration from ‘"oliver"sip:oliver@192.168.1.66;transport=UDP’ failed for ‘192.168.1.64’ - No matching peer found
[Sep 9 07:01:31] WARNING[2436] app_dial.c: Unable to create channel of type ‘SIP’ (cause 3 - No route to destination)
[Sep 9 07:01:31] WARNING[2436] app_dial.c: Unable to create channel of type ‘IAX2’ (cause 3 - No route to destination)
[Sep 9 07:01:52] NOTICE[2323] chan_sip.c: Registration from ‘"oliver"sip:oliver@192.168.1.66;transport=UDP’ failed for ‘192.168.1.64’ - No matching peer found

I don’t know what to do first. Could you please guide me I really needs your help. Thank you very much again. More power to you.

please post up the contents of extensions.conf and sip.conf (removing any passwords if necessary). Not sure where they are installed in *NOW, but they are probably in /etc/asterisk/

Dear g2010,
These are the following info:

sip.conf files:

[general]
context=default
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
[sip]
type=friend
context=internal
username=oliver
secret=123456
host=dynamic
canreinvite=no
dtfmode=rfc2833
disallow=all
allow=ulaw
subscribecontext=internal
mailbox=6050@internal
[authentication]
type=friend
context=internal
username=oliver
secret=123456
host=dynamic
canreinvite=no
dtfmode=rfc2833
disallow=all
allow=ulaw
subscribecontext=internal
mailbox=6050@internal

Extensions.conf files:

[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no
[globals]
trunk_1=Zap/g1
trunk_2=SIP/trunk_2
[dundi-e164-canonical]

[dundi-e164-customers]

[dundi-e164-via-pstn]

[dundi-e164-local]
include=dundi-e164-canonical
include=dundi-e164-customers
include=dundi-e164-via-pstn
[dundi-e164-switch]
switch=DUNDi/e164
[dundi-e164-lookup]
include=dundi-e164-local
include=dundi-e164-switch
[macro-dundi-e164]
exten=s,1,Goto(${ARG1},1)
include=dundi-e164-lookup
[iaxtel700]
exten=_91700XXXXXXX,1,Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel)
[iaxprovider]

[trunkint]
exten=_9011.,1,Macro(dundi-e164,${EXTEN:4})
exten=_9011.,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunkld]
exten=_91NXXNXXXXXX,1,Macro(dundi-e164,${EXTEN:1})
exten=_91NXXNXXXXXX,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunklocal]
exten=_9NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunktollfree]
exten=_91800NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten=_91888NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten=_91877NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten=_91866NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[international]
ignorepat=9
include=longdistance
include=trunkint
[longdistance]
ignorepat=9
include=local
include=trunkld
[local]
ignorepat=9
include=default
include=parkedcalls
include=trunklocal
include=iaxtel700
include=trunktollfree
include=iaxprovider
[macro-stdexten]
exten=s,1,Dial(${ARG2},20)
exten=s,2,Goto(s-${DIALSTATUS},1)
exten=s-NOANSWER,1,Voicemail(${ARG1},u)
exten=s-NOANSWER,2,Goto(default,s,1)
exten=s-BUSY,1,Voicemail(${ARG1},b)
exten=s-BUSY,2,Goto(default,s,1)
exten=_s-.,1,Goto(s-NOANSWER,1)
exten=a,1,VoicemailMain(${ARG1})
[macro-stdPrivacyexten]
exten=s,1,Dial(${ARG2},20|p)
exten=s,2,Goto(s-${DIALSTATUS},1)
exten=s-NOANSWER,1,Voicemail(u${ARG1})
exten=s-NOANSWER,2,Goto(default,s,1)
exten=s-BUSY,1,Voicemail(b${ARG1})
exten=s-BUSY,2,Goto(default,s,1)
exten=s-DONTCALL,1,Goto(${ARG3},s,1)
exten=s-TORTURE,1,Goto(${ARG4},s,1)
exten=_s-.,1,Goto(s-NOANSWER,1)
exten=a,1,VoicemailMain(${ARG1})
[macro-page]
exten=s,1,ChanIsAvail(${ARG1}|js)
exten=s,n,GoToIf([${AVAILSTATUS} = “1”]?autoanswer:fail)
exten=s,n(autoanswer),Set(_ALERT_INFO=“RA”)
exten=s,n,SIPAddHeader(Call-Info: Answer-After=0)
exten=s,n,NoOp()
exten=s,n,Dial(${ARG1}||)
exten=s,n(fail),Hangup
[demo]
exten=s,1,Wait(1)
exten=s,n,Answer
exten=s,n,Set(TIMEOUT(digit)=5)
exten=s,n,Set(TIMEOUT(response)=10)
exten=s,n(restart),BackGround(demo-congrats)
exten=s,n(instruct),BackGround(demo-instruct)
exten=s,n,WaitExten
exten=2,1,BackGround(demo-moreinfo)
exten=2,n,Goto(s,instruct)
exten=3,1,Set(LANGUAGE()=fr)
exten=3,n,Goto(s,restart)
exten=1000,1,Goto(default,s,1)
exten=1234,1,Playback(transfer,skip)
exten=1234,n,Macro(stdexten,1234,${CONSOLE})
exten=1235,1,Voicemail(u1234)
exten=1236,1,Dial(Console/dsp)
exten=1236,n,Voicemail(u1234)
exten=#,1,Playback(demo-thanks)
exten=#,n,Hangup
exten=t,1,Goto(#,1)
exten=i,1,Playback(invalid)
exten=500,1,Playback(demo-abouttotry)
exten=500,n,Dial(IAX2/guest@misery.digium.com/s@default)
exten=500,n,Playback(demo-nogo)
exten=500,n,Goto(s,6)
exten=600,1,Playback(demo-echotest)
exten=600,n,Echo
exten=600,n,Playback(demo-echodone)
exten=600,n,Goto(s,6)
exten=76245,1,Macro(page,SIP/Grandstream1)
exten=_7XXX,1,Macro(page,SIP/${EXTEN})
exten=7999,1,Set(TIMEOUT(absolute)=60)
exten=7999,2,Page(Local/Grandstream1@page&Local/Xlite1@page&Local/1234@page/n|d)
exten=8500,1,VoicemailMain
exten=8500,n,Goto(s,6)
[page]
exten=_X.,1,Macro(page,SIP/${EXTEN})
[default]
exten=6050,1,VoiceMailMain
exten=7000,1,Goto(voicemenu-custom-1|s|1)
exten=o,1,Goto(default,8000,1)
exten=9000,1,MeetMe(${EXTEN}|MsI)
[voicemenu-custom-1]
include=default
comment=Welcome
alias_exten=7000
exten=s,1,Answer
exten=s,2,Wait(1)
exten=s,3,Background(thank-you-for-calling)
exten=s,4,Background(if-u-know-ext-dial)
exten=s,5,Background(otherwise)
exten=s,6,Background(to-reach-operator)
exten=s,7,Background(pls-hold-while-try)
exten=s,8,WaitExten(6)
exten=1,1,Goto(default|8000|1)
[numberplan-custom-1]
plancomment=DialPlan1
include=default
include=parkedcalls
exten=_91XXXXXXXXXX!,1,Macro(trunkdial,${}/${EXTEN:1})
comment=_91XXXXXXXXXX!,1,Longdistance,standard
exten=_9011XXXXXXX!,1,Macro(trunkdial,${}/${EXTEN:1})
comment=_9011XXXXXXX!,1,International,standard
exten=_911!,1,Macro(trunkdial,${}/${EXTEN:0})
comment=_911!,1,911,standard
exten=_91700XXXXXXX!,1,Macro(trunkdial,${trunk_1}/${EXTEN:1},${trunk_1_cid})
comment=_91700XXXXXXX!,1,IAXTEL,standard
exten=_9XXXXXXX!,1,Macro(trunkdial,${trunk_1}/${EXTEN:1},${trunk_1_cid})
comment=_9XXXXXXX!,1,Local,standard
exten=_9XXXXXXX!,1,Macro(trunkdial,${trunk_1}/${EXTEN:4},${trunk_1_cid})
comment=_9XXXXXXX!,1,Local,standard
[macro-trunkdial]
exten=s,1,set(CALLERID(all)=${IF($["${LEN(${CALLERID(num)})}" > “6”]?${CALLERID(all)}:${ARG2})})
exten=s,n,Dial(${ARG1})
exten=s,n,Goto(s-${DIALSTATUS},1)
exten=s-NOANSWER,1,Hangup
exten=s-BUSY,1,Hangup
exten=_s-.,1,NoOp
[asterisk_guitools]
exten=executecommand,1,System(${command})
exten=executecommand,n,Hangup()
exten=record_vmenu,1,Answer
exten=record_vmenu,n,Playback(vm-intro)
exten=record_vmenu,n,Record(${var1})
exten=record_vmenu,n,Playback(vm-saved)
exten=record_vmenu,n,Playback(vm-goodbye)
exten=record_vmenu,n,Hangup
exten=play_file,1,Answer
exten=play_file,n,Playback(${var1})
exten=play_file,n,Hangup
[DID_trunk_1]
include=default
exten=_X.,1,Goto(default|7000|1)
exten=s,1,ExecIf($[ “${CALLERID(num)}”="" ],SetCallerPres,unavailable)
exten=s,2,ExecIf($[ “${CALLERID(num)}”="" ],Set,CALLERID(all)=unknown <0000000>)
exten=s,3,Goto(default|7000|1)
[numberplan-custom-2]
include=default
plancomment=DialPlan2
exten=_9XXXXXXX,1,Macro(trunkdial,${trunk_1}/${EXTEN:0},${trunk_1_cid})
comment=_9XXXXXXX,1,Dial out,standard
[numberplan-custom-3]
include=default
plancomment=DialPlan3SIP
exten=_9XXXXXXX!,1,Macro(trunkdial,${trunk_1}/${EXTEN:0},${trunk_1_cid})
comment=_9XXXXXXX!,1,SIP,standard
exten=_98XXXXXXX,1,Macro(trunkdial,${trunk_2}/${EXTEN:0},${trunk_2_cid})
comment=_98XXXXXXX,1,SIP,standard
[ringroups-custom-1]
gui_ring_groupname=testingRings
exten=s,1,NoOp(RINGGROUP)
exten=s,n,Dial(SIP/8000&SIP/6000&SIP/8001,20)
exten=s,n,Goto(voicemenu-custom-1|s|1)
[DID_trunk_2]
include=default
exten=_X.,1,Goto(default|7000|1)
exten=s,1,ExecIf($[ “${CALLERID(num)}”="" ],SetCallerPres,unavailable)
exten=s,2,ExecIf($[ “${CALLERID(num)}”="" ],Set,CALLERID(all)=unknown <0000000>)
exten=s,3,Goto(default|7000|1)

I hope this is enough. Thanks again for your kindness.

Well, I am not sure what you have configured in sip.conf is valid… I may be wrong, but I have just never seen nor done anything like that. The way I would configure the user “oliver” would be:

sip.conf

[general]
context=default
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes

[oliver]
type=friend
context=internal
username=oliver
secret=123456
host=dynamic
canreinvite=no
dtfmode=rfc2833
disallow=all
allow=ulaw
subscribecontext=internal
mailbox=6050@internal

Make that change and then see if things are working better. Once you are registered you may see that things start working better.

thanks,
g