Rejected because extension not found

Asterisk Asterisk Support
1

Hi,

I have been trying to get my 1.4 asterisk PBX working with Skype Connect. I have a SIP phone registered to the Asterisk but I cannot make outbound calls. I always fail with this message “chan_sip.c:14035 handle_request_invite: Call from ‘100’ to extension ‘442071792222’ rejected because extension not found.”

Here is the config can anyone see where the extension is going wrong?:

Extension.conf

[skype_in]
exten => 9905010011xxxx,Noop(${CALLERID(name)},${CALLERID(num)})

exten => 9905010011xxxx,n,Dial(SIP/9905010011xxxx,30,t,r)

exten => 9905010011xxxx,n,voicemail(100|u)

[skype_out]

exten => _90Z.,1,Set(CALLERID(num)=9905010011xxxx)

exten => _90Z.,n,Dial(SIP/0044${EXTEN:2}@9905010011xxxx)

exten => _900.,1,Set(CALLERID(num)=9905010011xxxx)

exten => _900.,n,Dial(SIP/${EXTEN:1}@9905010011xxxx)

[numberplan-custom-1]
plancomment = Default DialPlan
include = default
include = parkedcalls
exten = _0XXXXX!,1,Macro(trunkdial,${trunk_1}/${EXTEN:0},${trunk_1_cid})
comment = _0XXXXX!,1,test,standard
include => numberplan-customer-1

exten => _91NXXNXXXXXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
exten => _91NXXNXXXXXX,n,Playtones(congestion)
exten => _91NXXNXXXXXX,n,Hangup()

sip.conf

register => 9905100000xxx:PASSWORD@sip.skype.com/99051000003852

;[99051000003852]
;type=friend
;username=9905100000xxxx
;fromdomain=sip.skype.com

;fromuser=9905100000xxxx

;realm=sip.skype.com

;host=dynamic

;dtmfmode=rfc2833

;secret=PAsSSWORD

;nat=no ;This should be set to reflect your network NAT configuration

;canreinvite=yes

;insecure=invite

;qualify=yes

;allow=g729 ; Uncomment this if you have G729 licences

;allow=ulaw

;allow=alaw

;amaflags=default

;trustrpid=no

;sendrpid=yes

;context=skypetest

[9905010011xxxx]
user=9905010011xxxx
mailbox=100
callerid=“myskypetrunk” <9905100000xxxx>
type=friend
host=dynamic
context=international
nat=no
canreinvite=no
dtmfmode=rfc2833
pickupgroup=1
callgroup=1
subscribecontext=default
notifyringing=yes
disallow=all
allow=alaw
allow=ulaw
allow=gsm

;[99051000116684]
;username=9905010011xxxx
;fromdomain=sip.skype.com
;ealm=sip.skype.com
;context=default
;type=friend
;qualify=yes
;host=dynamic
;careinvite=no

[100]
username=100
type=friend
;secret=skypexxx
host=dynamic
contex=internal
insecure=port
[root@localhost asterisk]#

Any help here guys would be greatly appreciated.

  • M4rt0k
2

Your have no “internal” context.

3

Thanks David55,

I am pretty inexperienced with Asterisk (obviously…) how should the internal context look?

  • M4rt0k
4

[internal]
with the rules which you deem to be necessary.

On your sip phone 100 you said it is part of the context=internal. When it goes to dial, its rules are in [internal].

If your really green with asterisk, you should look over the book http://astbook.asteriskdocs.org/.

5

Hi David,

Thanks for the reply. Relating to the query. As you said I changed the context and below is the basic rule that I am using in extensions.conf.

exten => _00X.,1,Answer
exten => _00X.,n,Set(CALLERID(num)=99051000003852)
exten => _00X.,n,Dial(SIP/${100}@99051000003852)
exten => _00X.,n,Hangup

My sip.conf looks like this

[99051000xxxxxx]
user=990xxxxxxxxx
mailbox=100
callerid=“myskypetrunk” <99051xxxxxx>
type=friend
host=dynamic
context=international
nat=no
canreinvite=no
dtmfmode=rfc2833
pickupgroup=1
callgroup=1
subscribecontext=default
notifyringing=yes
disallow=all
allow=alaw
allow=ulaw
allow=gsm

[100]
user=100
mailbox=100
callerid=“myskypetrunk” <100>
type=friend
host=dynamic
context=international
nat=no
canreinvite=no
dtmfmode=rfc2833
pickupgroup=1
callgroup=1
subscribecontext=internal
notifyringing=yes
disallow=all
allow=alaw
allow=ulaw
allow=gsm

It gives me the following error

localhost*CLI>
<— SIP read from 172.16.62.4:65030 —>
INVITE sip:00447768271167@sip.skype.com SIP/2.0
Via: SIP/2.0/UDP 172.16.62.4:65030;branch=z9hG4bK-d8754z-2a8d1ef8fce4ff59-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:100@172.16.62.4:65030
To: sip:00447768271167@sip.skype.com
From: "sami"sip:100@sip.skype.com;tag=a7830447
Call-ID: MjViMWM5ZmQ4MGZmZGNkNDM4ZmI1MzFlM2NjYmNjMTg.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: Bria 3.0 release 3.0 stamp 56430
Content-Length: 352

v=0
o=- 12935157989375000 12935157989375000 IN IP4 172.16.62.4
s=
c=IN IP4 172.16.62.4
t=0 0
m=audio 57424 RTP/AVP 0 8 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=candidate:1 1 UDP 659136 172.16.62.4 57424 typ host
a=candidate:1 2 UDP 659134 172.16.62.4 57425 typ host

<------------->
— (12 headers 13 lines) —
Sending to 172.16.62.4 : 65030 (NAT)
Using INVITE request as basis request - MjViMWM5ZmQ4MGZmZGNkNDM4ZmI1MzFlM2NjYmNjMTg.
Found user '100’
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 172.16.62.4:57424
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 172.16.62.4:57424
Looking for 00447768271167 in international (domain sip.skype.com)
list_route: hop: sip:100@172.16.62.4:65030
localhost*CLI>
<— Transmitting (no NAT) to 172.16.62.4:65030 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.62.4:65030;branch=z9hG4bK-d8754z-2a8d1ef8fce4ff59-1—d8754z-;received=172.16.62.4;rport=65030
From: "sami"sip:100@sip.skype.com;tag=a7830447
To: sip:00447768271167@sip.skype.com
Call-ID: MjViMWM5ZmQ4MGZmZGNkNDM4ZmI1MzFlM2NjYmNjMTg.
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:00447768271167@172.16.62.2
Content-Length: 0

<------------>
– Executing [00447768271167@international:1] Answer(“SIP/100-08ec9be0”, “”) in new stack
Audio is at 172.16.62.2 port 16294
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
localhost*CLI>
<— Reliably Transmitting (no NAT) to 172.16.62.4:65030 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.62.4:65030;branch=z9hG4bK-d8754z-2a8d1ef8fce4ff59-1—d8754z-;received=172.16.62.4;rport=65030
From: "sami"sip:100@sip.skype.com;tag=a7830447
To: sip:00447768271167@sip.skype.com;tag=as0e847940
Call-ID: MjViMWM5ZmQ4MGZmZGNkNDM4ZmI1MzFlM2NjYmNjMTg.
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:00447768271167@172.16.62.2
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 13953 13953 IN IP4 172.16.62.2
s=session
c=IN IP4 172.16.62.2
t=0 0
m=audio 16294 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
– Executing [00447768271167@international:2] Set(“SIP/100-08ec9be0”, “CALLERID(num)=99051000003852”) in new stack
– Executing [00447768271167@international:3] Dial(“SIP/100-08ec9be0”, “SIP/@99051000003852”) in new stack
[Nov 25 11:22:24] WARNING[15696]: chan_sip.c:2921 create_addr: No such host: 99051000003852
Really destroying SIP dialog ‘54dbe8b847158ed5172c8233521d551a@127.0.0.1’ Method: INVITE
[Nov 25 11:22:24] WARNING[15696]: app_dial.c:1183 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
– Executing [00447768271167@international:4] Hangup(“SIP/100-08ec9be0”, “”) in new stack
== Spawn extension (international, 00447768271167, 4) exited non-zero on 'SIP/100-08ec9be0’
Scheduling destruction of SIP dialog ‘MjViMWM5ZmQ4MGZmZGNkNDM4ZmI1MzFlM2NjYmNjMTg.’ in 32000 ms (Method: INVITE)
localhost*CLI>
<— SIP read from 172.16.62.4:65030 —>
ACK sip:00447768271167@172.16.62.2 SIP/2.0
Via: SIP/2.0/UDP 172.16.62.4:65030;branch=z9hG4bK-d8754z-2d3fd790e9453fac-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:100@172.16.62.4:65030
To: sip:00447768271167@sip.skype.com;tag=as0e847940
From: "sami"sip:100@sip.skype.com;tag=a7830447
Call-ID: MjViMWM5ZmQ4MGZmZGNkNDM4ZmI1MzFlM2NjYmNjMTg.
CSeq: 1 ACK
User-Agent: Bria 3.0 release 3.0 stamp 56430
Content-Length: 0

<------------->
— (10 headers 0 lines) —
set_destination: Parsing sip:100@172.16.62.4:65030 for address/port to send to
set_destination: set destination to 172.16.62.4, port 65030
Reliably Transmitting (no NAT) to 172.16.62.4:65030:
BYE sip:100@172.16.62.4:65030 SIP/2.0
Via: SIP/2.0/UDP 172.16.62.2:5060;branch=z9hG4bK32e6db12;rport
From: sip:00447768271167@sip.skype.com;tag=as0e847940
To: "sami"sip:100@sip.skype.com;tag=a7830447
Call-ID: MjViMWM5ZmQ4MGZmZGNkNDM4ZmI1MzFlM2NjYmNjMTg.
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

Scheduling destruction of SIP dialog ‘MjViMWM5ZmQ4MGZmZGNkNDM4ZmI1MzFlM2NjYmNjMTg.’ in 32000 ms (Method: ACK)
localhost*CLI>
<— SIP read from 172.16.62.4:65030 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.62.2:5060;branch=z9hG4bK32e6db12;rport=5060
Contact: sip:100@172.16.62.4:65030
To: "sami"sip:100@sip.skype.com;tag=a7830447
From: sip:00447768271167@sip.skype.com;tag=as0e847940
Call-ID: MjViMWM5ZmQ4MGZmZGNkNDM4ZmI1MzFlM2NjYmNjMTg.
CSeq: 102 BYE
User-Agent: Bria 3.0 release 3.0 stamp 56430
Content-Length: 0

<------------->
— (9 headers 0 lines) —
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog ‘MjViMWM5ZmQ4MGZmZGNkNDM4ZmI1MzFlM2NjYmNjMTg.’ Method: ACK
Reliably Transmitting (no NAT) to 172.16.62.3:41156:
OPTIONS sip:99051000116684@172.16.62.3:41156 SIP/2.0
Via: SIP/2.0/UDP 172.16.62.2:5060;branch=z9hG4bK566f3d2e;rport
From: “asterisk” sip:asterisk@172.16.62.2;tag=as485441e6
To: sip:99051000116684@172.16.62.3:41156
Contact: sip:asterisk@172.16.62.2
Call-ID: 03267f623f146ab4749dcd843cf48e04@172.16.62.2
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 25 Nov 2010 11:22:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

localhost*CLI>
<— SIP read from 172.16.62.3:41156 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.62.2:5060;branch=z9hG4bK566f3d2e;rport=5060
From: “asterisk” sip:asterisk@172.16.62.2;tag=as485441e6
To: sip:99051000116684@172.16.62.3:41156;tag=1630151995
Call-ID: 03267f623f146ab4749dcd843cf48e04@172.16.62.2
CSeq: 102 OPTIONS
Supported: replaces, path, timer
User-Agent: Grandstream GXV3140 1.0.7.3
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

Any ideas.Thanks a lot for your help.

6

Where does the variable, ${100} get set?

Is the xxxxxxx in sip.conf, the right value to match the number in the Dial application parameters?

7

Hi David,

I am completely new to this. I am not sure about the $100 variable as I copied it from a dialplan on net.

About the xxxxxx in sip.conf, it is an extension number. Plz ignore the extension(9905xxxxxxxxxx) as we are now concentrating on extension [100].

Here is the extension.conf file and sip.conf file for extension ‘100’

extension.conf

exten => _00X.,1,Answer
exten => _00X.,n,Set(CALLERID(num)=99051000003852)
exten => _00X.,n,Dial(SIP/${100}@99051000003852)
exten => _00X.,n,Hangup

sip.conf

[100]
user=100
mailbox=100
callerid=“myskypetrunk” <100>
type=friend
host=dynamic
context=international
nat=no
canreinvite=no
dtmfmode=rfc2833
pickupgroup=1
callgroup=1
subscribecontext=internal
notifyringing=yes
disallow=all
allow=alaw
allow=ulaw
allow=gsm

Plz let me know if Ineed to do any changes. Sorry for the confusion

8

Cut and paste coding is rarely a good idea. As someone suggested, you need to take some time to read the Asterisk book. You should also look at the sample configuration provided with Asterisk source distributions.

9

Hi,

I just created my own dial plan from asterisk tutorial and gets this error message

[Nov 25 13:04:14] WARNING[16027]: channel.c:3025 ast_request: No channel type registered for ‘’
[Nov 25 13:04:14] WARNING[16027]: app_dial.c:1183 dial_exec_full: Unable to create channel of type ‘’ (cause 66 - Channel not implemented)
== Everyone is busy/congested at this time (1:0/0/1)
– Executing [9447768271167@international:2] PlayTones(“SIP/100-08bba490”, “congestion”) in new stack

Any ideas?

Thanks

10

There is an error in the first parameter of the Dial application, before the /. Possibly nothing before the /. Possibly misuse of " on a recent version of Asterisk.

You need to provide, at least, the — Executing line for the Dial application, so we can see what was actually passed.

11

<------------>
– Executing [9447768271167@international:1] Dial(“SIP/100-08bba8f0”, “/”) in new stack
[Nov 25 13:18:19] WARNING[16043]: channel.c:3025 ast_request: No channel type registered for ‘’
[Nov 25 13:18:19] WARNING[16043]: app_dial.c:1183 dial_exec_full: Unable to create channel of type ‘’ (cause 66 - Channel not implemented)
== Everyone is busy/congested at this time (1:0/0/1)
– Executing [9447768271167@international:2] PlayTones(“SIP/100-08bba8f0”, “congestion”) in new stack
Audio is at 172.16.62.2 port 17158
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Transmitting (no NAT) to 172.16.62.4:65030 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.16.62.4:65030;branch=z9hG4bK-d8754z-e9a0624b576d0a1e-1—d8754z-;received=172.16.62.4;rport=65030
From: "sami"sip:100@sip.skype.com;tag=6b31c97e
To: sip:9447768271167@sip.skype.com;tag=as323b3a9e
Call-ID: MzlhMWM2MWE2ZWQ3NDJjNGQ0NzI2Mzg1ZTUzNDM0OTc.
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:9447768271167@172.16.62.2
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 13953 13953 IN IP4 172.16.62.2
s=session
c=IN IP4 172.16.62.2
t=0 0
m=audio 17158 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
== Auto fallthrough, channel ‘SIP/100-08bba8f0’ status is 'CHANUNAVAIL’
localhost*CLI>
<— Transmitting (no NAT) to 172.16.62.4:65030 —>
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 172.16.62.4:65030;branch=z9hG4bK-d8754z-e9a0624b576d0a1e-1—d8754z-;received=172.16.62.4;rport=65030
From: "sami"sip:100@sip.skype.com;tag=6b31c97e
To: sip:9447768271167@sip.skype.com;tag=as323b3a9e
Call-ID: MzlhMWM2MWE2ZWQ3NDJjNGQ0NzI2Mzg1ZTUzNDM0OTc.
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:9447768271167@172.16.62.2
Content-Length: 0
X-Asterisk-HangupCause: Channel not implemented
X-Asterisk-HangupCauseCode: 66

My Dial plan looks like this(extensions.conf)
[incoming]
exten => s,1,Answer()
exten => s,1,Hangup()

[outgoing]
exten => 100,1,Answer()
exten => 100,n,Set()
exten => 100,n,Dial(SIP/${100}@99051000003852)
exten => 100,n,Hangup()

---------------(sip.conf)--------------------
[general]
type=friend
username=99051000003852
fromdomain=sip.skype.com
fromuser=99051000003852
realm=sip.skype.com
host=dynamic
dtmfmode=rfc2833
secret=Ujjv4dNW47s3ha
nat=no ;This should be set to reflect your network NAT configuration
canreinvite=yes
insecure=invite
qualify=yes
allow=g729 ; Uncomment this if you have G729 licences
allow=ulaw
allow=alaw
amaflags=default
trustrpid=no
sendrpid=yes
context=skypetest

[99051000116684]
user=99051000116684
mailbox=100
callerid=“myskypetrunk” <99051000003852>
type=friend
host=dynamic
context=international
nat=no
canreinvite=no
dtmfmode=rfc2833
pickupgroup=1
callgroup=1
subscribecontext=default
notifyringing=yes
disallow=all
allow=alaw
allow=ulaw
allow=gsm

[100]
user=100
mailbox=100
callerid=“myskypetrunk” <100>
type=friend
host=dynamic
context=international
nat=no
canreinvite=no
dtmfmode=rfc2833
pickupgroup=1
callgroup=1
subscribecontext=internal
notifyringing=yes
disallow=all
allow=alaw

12

The error is in the international context part of the dialplan, which you haven’t listed.

You are dialling with nothing before the /.

13

I entered this in the extensions.conf file

exten => _91NXXNXXXXXX,1,Dial(${99051000003852}/${100:1})
exten => _91NXXNXXXXXX,n,Playtones(congestion)
exten => _91NXXNXXXXXX,n,Hangup()

I tried dialling the number but the error is

— SIP read from 172.16.62.4:65030 —>
INVITE sip:914086277950@sip.skype.com SIP/2.0
Via: SIP/2.0/UDP 172.16.62.4:65030;branch=z9hG4bK-d8754z-e0d52f316055a9ce-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:100@172.16.62.4:65030
To: sip:914086277950@sip.skype.com
From: "sami"sip:100@sip.skype.com;tag=8a32d79c
Call-ID: Nzc5OTZlMmI4ZTI5ZDNmYzEyNDhmYjgzZGQzYWVjYjE.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: Bria 3.0 release 3.0 stamp 56430
Content-Length: 352

v=0
o=- 12935169730750000 12935169730750000 IN IP4 172.16.62.4
s=
c=IN IP4 172.16.62.4
t=0 0
m=audio 54990 RTP/AVP 0 8 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=candidate:1 1 UDP 659136 172.16.62.4 54990 typ host
a=candidate:1 2 UDP 659134 172.16.62.4 54991 typ host

<------------->
— (12 headers 13 lines) —
Sending to 172.16.62.4 : 65030 (NAT)
Using INVITE request as basis request - Nzc5OTZlMmI4ZTI5ZDNmYzEyNDhmYjgzZGQzYWVjYjE.
Found user '100’
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 172.16.62.4:54990
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 172.16.62.4:54990
Looking for 914086277950 in international (domain sip.skype.com)
list_route: hop: sip:100@172.16.62.4:65030
localhost*CLI>
<— Transmitting (no NAT) to 172.16.62.4:65030 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.62.4:65030;branch=z9hG4bK-d8754z-e0d52f316055a9ce-1—d8754z-;received=172.16.62.4;rport=65030
From: "sami"sip:100@sip.skype.com;tag=8a32d79c
To: sip:914086277950@sip.skype.com
Call-ID: Nzc5OTZlMmI4ZTI5ZDNmYzEyNDhmYjgzZGQzYWVjYjE.
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:914086277950@172.16.62.2
Content-Length: 0

<------------>
– Executing [914086277950@international:1] Dial(“SIP/100-08bb4ee8”, “/”) in new stack
[Nov 25 14:38:05] WARNING[16097]: channel.c:3025 ast_request: No channel type registered for ‘’
[Nov 25 14:38:05] WARNING[16097]: app_dial.c:1183 dial_exec_full: Unable to create channel of type ‘’ (cause 66 - Channel not implemented)
== Everyone is busy/congested at this time (1:0/0/1)
– Executing [914086277950@international:2] PlayTones(“SIP/100-08bb4ee8”, “congestion”) in new stack
Audio is at 172.16.62.2 port 17238
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
localhost*CLI>
<— Transmitting (no NAT) to 172.16.62.4:65030 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.16.62.4:65030;branch=z9hG4bK-d8754z-e0d52f316055a9ce-1—d8754z-;received=172.16.62.4;rport=65030
From: "sami"sip:100@sip.skype.com;tag=8a32d79c
To: sip:914086277950@sip.skype.com;tag=as0c4beecb
Call-ID: Nzc5OTZlMmI4ZTI5ZDNmYzEyNDhmYjgzZGQzYWVjYjE.
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:914086277950@172.16.62.2
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 13953 13953 IN IP4 172.16.62.2
s=session
c=IN IP4 172.16.62.2
t=0 0
m=audio 17238 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
– Executing [914086277950@international:3] Hangup(“SIP/100-08bb4ee8”, “”) in new stack
== Spawn extension (international, 914086277950, 3) exited non-zero on 'SIP/100-08bb4ee8’
Scheduling destruction of SIP dialog ‘Nzc5OTZlMmI4ZTI5ZDNmYzEyNDhmYjgzZGQzYWVjYjE.’ in 32000 ms (Method: INVITE)
localhost*CLI>
<— Reliably Transmitting (no NAT) to 172.16.62.4:65030 —>
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 172.16.62.4:65030;branch=z9hG4bK-d8754z-e0d52f316055a9ce-1—d8754z-;received=172.16.62.4;rport=65030
From: "sami"sip:100@sip.skype.com;tag=8a32d79c
To: sip:914086277950@sip.skype.com;tag=as0c4beecb
Call-ID: Nzc5OTZlMmI4ZTI5ZDNmYzEyNDhmYjgzZGQzYWVjYjE.
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:914086277950@172.16.62.2
Content-Length: 0

14

Where do the variables 99051000003852 and 100 get set? You really do need to spend two or three days reading the book.

(I rather suspect these should not be variables.)

15

Hi,

I get this error . Though I have tne DNS set as ‘yes’ and it is resolving from name to IP address. I get this error.

Executing [00447768271167@international:2] Set(“SIP/100-092735b8”, “CALLERID(100)=99051000003852”) in new stack
[Nov 26 14:24:00] ERROR[19192]: func_callerid.c:136 callerid_write: Unknown callerid data type ‘100’.
– Executing [00447768271167@international:3] Dial(“SIP/100-092735b8”, “SIP/@99051000003852”) in new stack
Really destroying SIP dialog ‘7aa24652775aa2ff5ec68d5f7a259d76@127.0.0.1’ Method: INVITE
[Nov 26 14:24:00] e[ 1;31;40mWARNING[19192]: app_dial.c:1183 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
– Executing [00447768271167@international:4] Hangup(“SIP/100-092735b8”, “”) in new stack
== Spawn extension (international, 00447768271167, 4) exited non-zero on 'SIP/100-0

Any suggestions?

Thanks

16

I would fix the two obvious errors first:

  1. passing an out of domain (maths sense) parameter to CALLERID;
  2. using an empty called party number.
17

yeah I did make a note of it. I also tried to change the ‘9905xxxx’ in extn.conf to domain name and this time I get this error

SIP/domain name-08390e20 is circuit-busy circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
– Executing [00447768271167@international:4] Hangup(“SIP/100-08394978”, “”) in new stack
== Spawn extension (international, 00447768271167, 4) exited non-zero on ‘SIP/100-08394978’

Thanks for your help

Sami