Help with 404 error message in Asterisk freePBX

Hello,

I hope some out there can assist me with this problem.

I am using freePXB 2.2 (no I haven’t upgraded yet, but may have to after the past weeks events) and as of last week my system stop receiving inbound calls.

I am using Bandwidth.Com for my provisioning and they say they haven’t changed anything, but that cannot be true. Of course we are getting the “We’re sorry, you have reached a number that is no longer in service or has been disconnected” - 404).

After reporting this to bandwidth, of course they are saying that this is due to something changing in the pbx. So after arguing for the past few days, after I checked all the settings and found nothing out of order. I restored the system to last months back, when everything was working properly, and we still get the message.

Bandwidth says there is still something wrong with my system even though I went back to where the system should have been working and you would think that would fix the problem, well it didn’t. So now I do not know what else to do.

Here is a copy of their trace they sent me…can you tell me where else to look or should I continue to fuss with bandwidth?

Thanks…desperate.

B. Ingram

TRACE:

U 2008/10/13 18:05:58.193675 216.82.224.202:5060 -> 70.63.224.58:5060
INVITE sip:+18036674336@70.63.224.58:5060;transport=udp SIP/2.0.
Record-Route: sip:216.82.224.202;lr;ftag=VPSF506071629460.
Record-Route: sip:4.79.212.229;lr;ftag=VPSF506071629460.
Via: SIP/2.0/UDP 216.82.224.202;branch=z9hG4bKca4c.1a7be205.0.
Via: SIP/2.0/UDP 4.79.212.229;branch=z9hG4bKca4c.648d45d.0.
Via: SIP/2.0/UDP 4.68.250.148:5060;branch=z9hG4bK506071629460-1207147391919.
From: sip:+19196778490@4.68.250.148;isup-oli=0;tag=VPSF506071629460.
To: sip:+18036674336@4.79.212.229:5060.
Call-ID: MCLMGC0520081013180557030540@209.244.62.67.
CSeq: 1 INVITE.
Contact: sip:+19196778490@4.68.250.148:5060;transport=udp.
Max-Forwards: 67.
Content-Type: application/sdp.
Content-Length: 171.
Remote-Party-ID: sip:+19196778490@4.68.250.148;party=calling;screen=yes;privacy=off.
.
v=0.
o=- 1223921157 1223921158 IN IP4 4.68.248.196.
s=-.
c=IN IP4 4.68.248.196.
t=0 0.
m=audio 62158 RTP/AVP 0 18 101.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.

U 2008/10/13 18:05:58.225168 70.63.224.58:5060 -> 216.82.224.202:5060
SIP/2.0 404 Not Found.
Via: SIP/2.0/UDP 216.82.224.202;branch=z9hG4bKca4c.1a7be205.0;received=216.82.224.202.
Via: SIP/2.0/UDP 4.79.212.229;branch=z9hG4bKca4c.648d45d.0.
Via: SIP/2.0/UDP 4.68.250.148:5060;branch=z9hG4bK506071629460-1207147391919.
From: sip:+19196778490@4.68.250.148;isup-oli=0;tag=VPSF506071629460.
To: sip:+18036674336@4.79.212.229:5060;tag=as5c247caa.
Call-ID: MCLMGC0520081013180557030540@209.244.62.67.
CSeq: 1 INVITE.
User-Agent: Asterisk PBX.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
Contact: sip:+18036674336@70.63.224.58.
Content-Length: 0.

MY CONTEXT ON THE SIP TRUNK:
;context=from-trunk
context=from-bandwidth
host=216.82.224.202&216.82.225.202
secret=
type=user

What does this return from the asterisk CLI

[ Context ‘from-bandwidth’ created by ‘pbx_config’ ]
’_+1.’ => 1. Set(DID=${EXTEN:2}) [pbx_config]
2. Goto(from-trunk|${DID}|1) [pbx_config]

-= 1 extension (2 priorities) in 1 context. =-
Verbosity is at least 3

Continue following this through to make sure it is being handled…

The next step would be:

dialplan show 8036674336@from-trunk

[ Included context ‘ext-did’ created by ‘pbx_config’ ]
’_X.’ => 1. Noop(Catch-All DID Match - Found ${EXTEN} - You probably want a DID for this.) [pbx_config]
2. Goto(ext-did|s|1) [pbx_config]

-= 1 extension (2 priorities) in 1 context. =-
Verbosity is at least 3

[ Included context ‘ext-did’ created by ‘pbx_config’ ]
‘8036674439’ => 1. Set(FROM_DID=8036674439) [pbx_config]
2. Gosub(app-blacklist-check|s|1) [pbx_config]
3. Goto(ext-meetme|8200|1) [pbx_config]
’_X.’ => 1. Noop(Catch-All DID Match - Found ${EXTEN} - You probably want a DID for this.) [pbx_config]
2. Goto(ext-did|s|1) [pbx_config]

-= 2 extensions (5 priorities) in 1 context. =-
Verbosity is at least 3

[ Included context ‘ext-did’ created by ‘pbx_config’ ]
‘8036674435’ => 1. Set(FROM_DID=8036674435) [pbx_config]
2. Gosub(app-blacklist-check|s|1) [pbx_config]
3. Goto(ext-local|6800|1) [pbx_config]
’_X.’ => 1. Noop(Catch-All DID Match - Found ${EXTEN} - You probably want a DID for this.) [pbx_config]
2. Goto(ext-did|s|1) [pbx_config]

-= 2 extensions (5 priorities) in 1 context. =-
Verbosity is at least 3

Ok, it looks like you are sending the call into MeetMe. When you call the number, do you see any problems on the CLI? For example no such conference, or unable to open psuedo device?

Is ztdummy loaded? (If you don’t have any zap cards)

lsmod|grep ztdummy

Edit maybe not… I read that wrong…

I sent 3 examples, sorry if I confused you, but

6674435 goes to extension 6800
6674439 is a conference number
6674336 is not setup yet

so let’s look at 35 or 39, thanks.