Unable to dial using SIP


#1

Hi all,

I’m unable to dial the SIP channel from asterisk but i’m able to dial from the SIP phone. Please Help!

It appears that * is using the wrong credential to authenticate with the termination proxy server.

Here’s the output from asterisk:

-- Accepting call from '8002891489' to '18589226688' on channel 0/7, span 1
-- Executing Dial("Zap/7-1", "SIP/intlno/18589226688") in new stack
-- Called intlno/18589226688
-- SIP/intlno-c8d7 is making progress passing it to Zap/7-1

May 31 09:58:51 WARNING[25946]: chan_sip.c:9476 handle_response_invite: Forbidden - wrong password on authentication for INVITE to ‘“8002891489” sip:yyyy@206.169.193.100;tag=as688dc052’
– SIP/intlno-c8d7 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
– Executing Congestion(“Zap/7-1”, “”) in new stack
== Spawn extension (outgoing, 18589226688, 2) exited non-zero on ‘Zap/7-1’
– Hungup ‘Zap/7-1’

extensions.conf

[outgoing]
exten => _1NXXNXXXXXX,1,Dial(SIP/intlno/${EXTEN})
exten => _1NXXNXXXXXX,2,Congestion

sip.conf
[general]
context=default
register=> yyyy:xxxx@sip.intlno.com

[intlno]
type=peer
secret=xxxx
host=sip.intlno.com
username=yyyy
context=intlno
disallow=all
qualify=no
allow=ulaw

Regards,
peter


#2

try to add in sip.conf in provider configuration
[intlno]
insecure=invite

-FD


#3

FD,

Thanks for the response but i’m still getting the same error msg.

May 31 14:50:05 WARNING[25946]: chan_sip.c:9476 handle_response_invite: Forbidden - wrong password on authentication for INVITE


#4

register => login:password@sip.provider

[sip.provider]
type = friend
host=sip.provider.address
fromdomain = sip.provider.address
user = login
fromuser = login
username = login
secret = password
insecure = invite
qualify = yes
nat = yes
context = context_for_incoming_calls

I use something like this above.
If i remove ‘insecure=invite’ i’m getting warning as You do:
…handle_response_invite: Forbidden - wrong password on authentication for INVITE…

voip-info.org/wiki/index.php … g+sip.conf


#5

FD,

Thanks again for your help with this. I’ve made the changes as you suggested as shown below. i’ve replaced login and password with xxx and yyyy but i’m gettting the same error.

I’ve also pasted the content of sip debug on here as well

thanks!

[general]
context=default
register=> xxxx:yyyy@sip.intlno.com

[sip.intlno.com]
type = peer
host = sip.intlno.com
fromdomain = sip.intlno.com
user = xxxx
fromuser = xxxx
username = xxxx
secret = yyyy
insecure = invite
canreinvite = yes
context = intlno
disallow = all
qualify = no
allow = ulaw

IP Debugging enabled
Destroying call '5ee3ffbb5244b81725d389015d35910a@sip.intlno.com
– Accepting call from ‘8002891489’ to ‘18589226688’ on channel 0/11, span 1
– Executing Dial(“Zap/11-1”, “SIP/18589226688@sip.intlno.com|30”) in new stack
We’re at 206.169.193.100 port 15544
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 10 lines
Reliably Transmitting (no NAT) to 66.35.63.247:5060:
INVITE sip:18589226688@sip.intlno.com SIP/2.0
Via: SIP/2.0/UDP 206.169.193.100:5060;branch=z9hG4bK7d095c39;rport
From: “8002891489” sip:ProC1@sip.intlno.com;tag=as43c5fb48
To: sip:18589226688@sip.intlno.com
Contact: sip:ProC1@206.169.193.100
Call-ID: 17d450ce081a59dd6d4f077d43694d2f@sip.intlno.com
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 31 May 2006 21:43:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 218

v=0
o=root 747 747 IN IP4 206.169.193.100
s=session
c=IN IP4 206.169.193.100
t=0 0
m=audio 15544 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -


-- Called [18589226688@sip.intlno.com](mailto:18589226688@sip.intlno.com)

Retransmitting #1 (no NAT) to 66.35.63.247:5060:
INVITE sip:18589226688@sip.intlno.com SIP/2.0
Via: SIP/2.0/UDP 206.169.193.100:5060;branch=z9hG4bK7d095c39;rport
From: “8002891489” sip:ProC1@sip.intlno.com;tag=as43c5fb48
To: sip:18589226688@sip.intlno.com
Contact: sip:ProC1@206.169.193.100
Call-ID: 17d450ce081a59dd6d4f077d43694d2f@sip.intlno.com
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 31 May 2006 21:43:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 218

v=0
o=root 747 747 IN IP4 206.169.193.100
s=session
c=IN IP4 206.169.193.100
t=0 0
m=audio 15544 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -


ASTERISK*CLI>
<-- SIP read from 66.35.63.247:5060:
SIP/2.0 100 trying – your call is important to us
Via: SIP/2.0/UDP 206.169.193.100:5060;branch=z9hG4bK7d095c39;rport=5060
From: “8002891489” sip:ProC1@sip.intlno.com;tag=as43c5fb48
To: sip:18589226688@sip.intlno.com
Call-ID: 17d450ce081a59dd6d4f077d43694d2f@sip.intlno.com
CSeq: 102 INVITE
Server: Sip EXpress router (0.9.4 (i386/freebsd))
Content-Length: 0

— (8 headers 0 lines)—
ASTERISK*CLI>
<-- SIP read from 66.35.63.247:5060:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 206.169.193.100:5060;branch=z9hG4bK7d095c39;rport=5060
Record-Route: sip:66.35.63.247;ftag=as43c5fb48;lr
From: 8002891489 sip:ProC1@sip.intlno.com;tag=as43c5fb48
To: sip:18589226688@sip.intlno.com;tag=42e7a2e9106ccf683c2b6e8b06dcdf25
Call-ID: 17d450ce081a59dd6d4f077d43694d2f@sip.intlno.com
CSeq: 102 INVITE
Server: Sippy
Content-Length: 440
Content-Type: application/sdp

v=0
o=root 6579 6579 IN IP4 66.35.63.247
s=session
t=0 0
m=audio 37798 RTP/AVP 18 4 2 3 7 110 97 5 0 8 101
c=IN IP4 66.35.63.247
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:7 LPC/8000
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

— (10 headers 19 lines)—
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 2
Found RTP audio format 3
Found RTP audio format 7
Found RTP audio format 110
Found RTP audio format 97
Found RTP audio format 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 66.35.63.247:37798
Found description format G729
Found description format G723
Found description format G726-32
Found description format GSM
Found description format LPC
Found description format speex
Found description format iLBC
Found description format DVI4
Found description format PCMU
Found description format PCMA
Found description format telephone-event
Capabilities: us - 0x4 (ulaw), peer - audio=0x7bf (g723|gsm|ulaw|alaw|g726|adpcm|lpc10|g729|speex|ilbc)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
– SIP/sip.intlno.com-b12f is making progress passing it to Zap/11-1
<-- SIP read from 66.35.63.247:5060:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 206.169.193.100:5060;branch=z9hG4bK7d095c39;rport=5060
Record-Route: sip:66.35.63.247;ftag=as43c5fb48;lr
From: 8002891489 sip:ProC1@sip.intlno.com;tag=as43c5fb48
To: sip:18589226688@sip.intlno.com;tag=42e7a2e9106ccf683c2b6e8b06dcdf25
Call-ID: 17d450ce081a59dd6d4f077d43694d2f@sip.intlno.com
CSeq: 102 INVITE
Server: Sippy

— (8 headers 0 lines)—
Transmitting (no NAT) to 66.35.63.247:5060:
ACK sip:18589226688@sip.intlno.com SIP/2.0
Via: SIP/2.0/UDP 206.169.193.100:5060;branch=z9hG4bK7d095c39;rport
From: “8002891489” sip:ProC1@sip.intlno.com;tag=as43c5fb48
To: sip:18589226688@sip.intlno.com;tag=42e7a2e9106ccf683c2b6e8b06dcdf25
Contact: sip:ProC1@206.169.193.100
Call-ID: 17d450ce081a59dd6d4f077d43694d2f@sip.intlno.com
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


May 31 14:43:48 WARNING[25946]: chan_sip.c:9476 handle_response_invite: Forbidden - wrong password on authentication for INVITE to ‘“8002891489” sip:ProC1@sip.intlno.com;tag=as43c5fb48’
– SIP/sip.intlno.com-b12f is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
– Executing Congestion(“Zap/11-1”, “”) in new stack
== Spawn extension (outgoing, 18589226688, 2) exited non-zero on ‘Zap/11-1’
– Hungup ‘Zap/11-1’ug
Destroying call '17d450ce081a59dd6d4f077d43694d2f@sip.intlno.com