SOLVED response 503 "PSTN Termination Currently Unavail

[panadyne]

Please help... I can't dial out... have a vicidailnow1.3 running

Watching asterisk log shows:

Got SIP response 503 "PSTN Termination Currently Unavailable"

Between the dialer and the outside world is pfsense firewall with siproxd.

The other asterisk server has no problems talking with this carrier, only the vidicdial does...

Dialer registers fine,

Any ideas??

VERSION: 2.0.5-174
BUILD: 90522-0506

[ticoit]

That seems to be a message returned by your sip proxy instead of something caused by vicidial (vicidialnow 1.3).

Could you post any details on how your carrier is setup. I think we can start from there.

-luis

[panadyne]

Attached are the config from carriers screen and the sip.conf

- Again, these settings work for another asterisk box with the same host...

- I have a packet capture, of the same actions, just don't wanna post it without sanitation.

[link2voip1]
type=friend
host=removed
username=removed
secret=removed
canreinvite=no
insecure=port,invite
qualify=5000
dtmfmode=rfc2833
nat=no
disallow=all
allow=g729
allow=ulaw

from sip.conf - the default settings except for an external ip and internal subnet

[general]
context=default ; Default context for incoming calls
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 ; Enable DNS SRV lookups on outbound calls
tos=lowdelay ; lowdelay,throughput,reliability,mincost,none
maxexpiry=3600 ; Max length of incoming registration we allow
defaultexpiry=120 ; Default length of incoming/outgoing registration
disallow=all ; First disallow all codecs
allow=ulaw ; Allow codecs in order of preference
allow=gsm ;
musicclass=default ; Sets the default music on hold class for all SIP calls
language=en ; Default language setting for all users/peers
relaxdtmf=yes ; Relax dtmf handling
rtptimeout=60 ; Terminate call if 60 seconds of no RTP activity
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
promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address
dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833
callevents=no ; generate manager events when sip ua performs events (e.g. hold)
externip =removed ; Address that we're going to put in outbound SIP messages
localnet=192.168.2.0/255.255.255.0; All RFC 1918 addresses are local networks
nat=no ; Global NAT settings (Affects all peers and users)
canreinvite=no

[ticoit]

The 503 message is being sent back from your sip proxy server. Is your extensions.conf configured the same way as in the working server? Check the relevant section of your dialplan.

-luis

[panadyne]

the dialplan part of the carrier configuration...

exten => _91NxxNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NxxNXXXXXX,2,Dial(${TESTSIPTRUNK}/${EXTEN:2},,tTor)
exten => _91NxxNXXXXXX,3,Hangup

and the extensions.conf is untouched from the install..

the working server is a pbxinaflash asterisk 1.4.21.2


Just to make sure that I am not leaving out any info... the asterisk log segment that shows the error

-- Executing MeetMe("Local/8600051@default-a21d,2", "8600051|F") in new stack
> Channel Local/8600051@default-a21d,1 was answered.
-- Executing AGI("Local/8600051@default-a21d,1", "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 Dial("Local/8600051@default-a21d,1", "SIP/link2voip1/XXXXXXXXXX||tTor") in new stack
-- Called link2voip1/3068806879
-- Got SIP response 503 "PSTN Termination Currently Unavailable" back from 66.51.110.210
-- SIP/link2voip1-08d60fb0 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Hangup("Local/8600051@default-a21d,1", "") in new stack
== Spawn extension (default, 91XXXXXXXXXX, 3) exited non-zero on 'Local/8600051@default-a21d,1'
-- Executing DeadAGI("Local/8600051@default-a21d,1", "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
== Manager 'sendcron' logged off from 127.0.0.1

[ticoit]

You could try changing your dialplan from:

exten => _91NxxNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NxxNXXXXXX,2,Dial(${TESTSIPTRUNK}/${EXTEN:2},,tTor)
exten => _91NxxNXXXXXX,3,Hangup

to:

exten => _91NxxNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NxxNXXXXXX,2,Dial(SIP/${EXTEN:2}@link2voip1,,tTor)
exten => _91NxxNXXXXXX,3,Hangup

Now for the dialplan pattern:

_91NxxNXXXXXX you send to your proxy server NxxNXXXXXX ->

N: any digit from 2-9
xx: any two lower case letters from a-z
N: any digits from 2-9
XXXXXX: any six digits from 0-9

Is this what you intended?

-luis

[williamconley]

ultimately, if they are both asterisk 1.2 and using sip to get to the carrier, there must be some OTHER difference for the carrier to know them apart and reject one.

unless you have some difference in the account being signed onto (i hope you're using two different accounts on the carrier, or two "devices" or whatever they want to call it, to avoid confusion). Or some other authenticatin method.

a way to check would be to use "sip debug" to find the Real reject message (not the "translated for console" version).

also: on the working box, what does this part look like:

non-working from the command line when "dialing":

Code: Select all

SIP/link2voip1/XXXXXXXXXX||tTor

if there is a difference in the structure of the dial request, that could set them apart and be part of the failure.

Other than SIP authentication and dial request and IP address/port, there should not be much to make the two servers different from one another.

You can also attempt to set up a SIP account independent of Vicidial (skip the GUI, make it in sip.conf/extensions.conf and get it working with the SAME settings as the other server, then make minor changes to make it compatible with Vicidial and *poof* you're on)

[panadyne]

Catching up on the solutions/ideas

Luis:

-changing the case of the dialplan to UPPER for the Nxx to NXX did not change anything.

-changing the dial plan as suggested does not help

William:

-yes, they are different accounts...

- having looked at the packet capture, the error is broadcast from the firewall/proxy... sip debug shows same thing

[williamconley]

do both machines have a firewall?

do both machines have the SAME firewall?

[panadyne]

yes...

same network, use the same pfsense firewall... I have other machines talking voip through that box to multiple carriers and I have no problems.

While I don't want to, the vicidial has a 2nd network port, I will hook it up to the internet without a firewall between ...

and if that doesn't work, I will resinstall vidcidialnow

[ticoit]

Any chance we could look at your sip proxy code? Asterisk is just reporting back to you that the call was refused by it so the sip proxy has weight on this as well.

Worth looking into it in my opinion.

-luis

[williamconley]

a way to check would be to use "sip debug" to find the Real reject message (not the "translated for console" version).

non-working from the command line when "dialing":

Code: Select all

SIP/link2voip1/XXXXXXXXXX||tTor

if there is a difference in the structure of the dial request, that could set them apart and be part of the failure.

still missing these two items. actual code from sip debug and the log or console output of the "working" vs "not working" boxes. obviously there is a difference between them or they would both work the same. after we see the differences, we can bring them into sync and this one should begin working.

[panadyne]

I have gone as far as to reinstall vicidial1.3 ( kept the previous install on a harddrive...)

The only change to sip.conf is to create an entry for externalIP and for localnet. I have copied and pasted the carrier configurations - A different carrier - lesnet, but same result-

- Variation that does not impact this problem- NAT= yes or no
____________________________________________________________

Below is a copy of sip debug when trying to manual dial from vicidialnow


--- (0 headers 1 lines) ---
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing MeetMe("Local/8600051@default-dc26,2", "8600051|F") in new stack
> Channel Local/8600051@default-dc26,1 was answered.
-- Executing AGI("Local/8600051@default-dc26,1", "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 Dial("Local/8600051@default-dc26,1", "SIP/lesnet/3069793286||tTor") in new stack
We're at 70.76.70.129 port 18970
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x10 (g726) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
14 headers, 14 lines
Reliably Transmitting (NAT) to 64.34.181.47:5060:
INVITE sip:3069793286@REMOVED SIP/2.0
Via: SIP/2.0/UDP 70.76.70.129:5060;branch=z9hG4bK4b7f5559;rport
From: "M1207233515000000003" <sip:0000000000@70.76.70.129>;tag=as78e3c2a4
To: <sip:3069793286@REMOVED>
Contact: <sip:0000000000@70.76.70.129>
Call-ID: 4875025c1539e7be13a4b30a229bed87@70.76.70.129
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "M1207233515000000003" <sip:0000000000@70.76.70.129>;privacy=off;screen=no
Date: Tue, 08 Dec 2009 04:35:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 315

v=0
o=root 2605 2605 IN IP4 70.76.70.129
s=session
c=IN IP4 70.76.70.129
t=0 0
m=audio 18970 RTP/AVP 0 18 3 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
-- Called lesnet/3069793286
vc*CLI>
<-- SIP read from 64.34.181.47:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 70.76.70.129:5060;branch=z9hG4bK4b7f5559;received=70.76.70.129;rport=53855
From: "M1207233515000000003" <sip:0000000000@70.76.70.129>;tag=as78e3c2a4
To: <sip:3069793286@REMOVED>
Call-ID: 4875025c1539e7be13a4b30a229bed87@70.76.70.129
CSeq: 102 INVITE
User-Agent: LESNETVoIP
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:3069793286@64.34.181.47>
Content-Length: 0


--- (10 headers 0 lines) ---
vc*CLI>
<-- SIP read from 64.34.181.47:5060:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 70.76.70.129:5060;branch=z9hG4bK4b7f5559;received=70.76.70.129;rport=53855
From: "M1207233515000000003" <sip:0000000000@70.76.70.129>;tag=as78e3c2a4
To: <sip:3069793286@REMOVED>;tag=as0bed412f
Call-ID: 4875025c1539e7be13a4b30a229bed87@70.76.70.129
CSeq: 102 INVITE
User-Agent: LESNETVoIP
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:3069793286@64.34.181.47>
Content-Length: 0


--- (10 headers 0 lines) ---
-- Got SIP response 503 "Service Unavailable" back from 64.34.181.47
Transmitting (NAT) to 64.34.181.47:5060:
ACK sip:3069793286@REMOVED SIP/2.0
Via: SIP/2.0/UDP 70.76.70.129:5060;branch=z9hG4bK4b7f5559;rport
From: "M1207233515000000003" <sip:0000000000@70.76.70.129>;tag=as78e3c2a4
To: <sip:3069793286@REMOVED>;tag=as0bed412f
Contact: <sip:0000000000@70.76.70.129>
Call-ID: 4875025c1539e7be13a4b30a229bed87@70.76.70.129
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "M1207233515000000003" <sip:0000000000@70.76.70.129>;privacy=off;screen=no
Content-Length: 0


---
-- SIP/lesnet-08e51b10 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Hangup("Local/8600051@default-dc26,1", "") in new stack
== Spawn extension (default, 913069793286, 3) exited non-zero on 'Local/8600051@default-dc26,1'
-- Executing DeadAGI("Local/8600051@default-dc26,1", "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
Destroying call '4875025c1539e7be13a4b30a229bed87@70.76.70.129'
== Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-dc26,2'
-- Executing DeadAGI("Local/8600051@default-dc26,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
vc*CLI>
<-- SIP read from 192.168.2.203:5060:

____________________________________________________________

Below is a sip debug from pbxinaflash that works




Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 66.51.127.173:5060:
INVITE sip:13068806879@REMOVED SIP/2.0

Via: SIP/2.0/UDP 70.76.70.129:5060;branch=z9hG4bK5514af59;rport

From: "REMOVED" <sip:3068806879@70.76.70.129>;tag=as38dd9f02

To: <sip:13068806879@REMOVED>

Contact: <sip:3068806879@70.76.70.129>

Call-ID: 6a40700c7124e70d310ad85c078acc63@70.76.70.129

CSeq: 102 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Wed, 09 Dec 2009 01:10:57 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Content-Type: application/sdp

Content-Length: 238



v=0

o=root 3745 3745 IN IP4 70.76.70.129

s=session

c=IN IP4 70.76.70.129

t=0 0

m=audio 16640 RTP/AVP 0 101

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -

a=ptime:20

a=sendrecv


<--- SIP read from 66.51.127.173:5060 --->
SIP/2.0 100 Trying

Via: SIP/2.0/UDP 70.76.70.129:5060;branch=z9hG4bK5514af59;rport=60125

From: "REMOVED" <sip:3068806879@70.76.70.129>;tag=as38dd9f02

To: <sip:13068806879@REMOVED>

Call-ID: 6a40700c7124e70d310ad85c078acc63@70.76.70.129

CSeq: 102 INVITE

Content-Length: 0



<--- SIP read from 66.51.127.173:5060 --->
SIP/2.0 407 Proxy Authentication Required

Via: SIP/2.0/UDP 70.76.70.129:5060;branch=z9hG4bK5514af59;rport=60125

From: "REMOVED" <sip:3068806879@70.76.70.129>;tag=as38dd9f02

To: <sip:13068806879@REMOVED>;tag=6da5cb3c58ecfc1b91772f44357856fa.c29b

Call-ID: 6a40700c7124e70d310ad85c078acc63@70.76.70.129

CSeq: 102 INVITE

Proxy-Authenticate: Digest realm="70.76.70.129", nonce="4b1efa4e4af575235daa3ed09166f322298ee456"

Content-Length: 0




Transmitting (no NAT) to 66.51.127.173:5060:
ACK sip:13068806879@REMOVED SIP/2.0

Via: SIP/2.0/UDP 70.76.70.129:5060;branch=z9hG4bK5514af59;rport

From: "REMOVED" <sip:3068806879@70.76.70.129>;tag=as38dd9f02

To: <sip:13068806879@REMOVED>;tag=6da5cb3c58ecfc1b91772f44357856fa.c29b

Contact: <sip:3068806879@70.76.70.129>

Call-ID: 6a40700c7124e70d310ad85c078acc63@70.76.70.129

CSeq: 102 ACK

User-Agent: Asterisk PBX

Max-Forwards: 70

Content-Length: 0




Reliably Transmitting (no NAT) to 66.51.127.173:5060:
INVITE sip:13068806879@REMOVED SIP/2.0

Via: SIP/2.0/UDP 70.76.70.129:5060;branch=z9hG4bK30567deb;rport

From: "REMOVED" <sip:3068806879@70.76.70.129>;tag=as38dd9f02

To: <sip:13068806879@REMOVED>

Contact: <sip:3068806879@70.76.70.129>

Call-ID: 6a40700c7124e70d310ad85c078acc63@70.76.70.129

CSeq: 103 INVITE

User-Agent: Asterisk PBX

Max-Forwards: 70

Proxy-Authorization: Digest username="REMOVED", realm="70.76.70.129", algorithm=MD5, uri="sip:13068806879@REMOVED", nonce="4b1efa4e4af575235daa3ed09166f322298ee456", response="b22318b2c6c869ead953db6793ec5952"

Date: Wed, 09 Dec 2009 01:10:57 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Content-Type: application/sdp

Content-Length: 238



v=0

o=root 3745 3746 IN IP4 70.76.70.129

s=session

c=IN IP4 70.76.70.129

t=0 0

m=audio 16640 RTP/AVP 0 101

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -

a=ptime:20

a=sendrecv


---



<--- SIP read from 66.51.127.173:5060 --->
SIP/2.0 100 Trying

Via: SIP/2.0/UDP 70.76.70.129:5060;branch=z9hG4bK30567deb;rport=60125

From: "REMOVED" <sip:3068806879@70.76.70.129>;tag=as38dd9f02

To: <sip:13068806879@REMOVED>

Call-ID: 6a40700c7124e70d310ad85c078acc63@70.76.70.129

CSeq: 103 INVITE

Content-Length: 0




<--- SIP read from 66.51.127.173:5060 --->
SIP/2.0 183 Session Progress

Via: SIP/2.0/UDP 70.76.70.129:5060;branch=z9hG4bK30567deb;rport=60125

Record-Route: <sip:66.51.127.173;lr;ftag=as38dd9f02;nat=yes>

From: "REMOVED" <sip:3068806879@70.76.70.129>;tag=as38dd9f02

To: <sip:13068806879@REMOVED>;tag=ratD1gFZe9pQr

Call-ID: 6a40700c7124e70d310ad85c078acc63@70.76.70.129

CSeq: 103 INVITE

Contact: <sip:ciscosip@66.51.127.162:5060;transport=udp>

User-Agent: Cisco-SIPGateway/IOS-12.x

Accept: application/sdp

Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, INFO

Supported: timer, precondition, path, replaces

Allow-Events: talk, refer

Content-Type: application/sdp

Content-Disposition: session

Content-Length: 301



v=0

o=CiscoSystemsSIP-GW-UserAgent 7920449753154907727 7459957886221735931 IN IP4 66.51.127.162

s=SIP Call

c=IN IP4 66.51.127.173

t=0 0

m=audio 13304 RTP/AVP 0 101

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -

a=ptime:20

a=nortpproxy:yes


Peer audio RTP is at port 66.51.127.173:13304


Found audio description format PCMU for ID 0


Found audio description format telephone-event for ID 101


Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)


Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)


Peer audio RTP is at port 66.51.127.173:13304


-- SIP/home_out_a-09fad1d8 is making progress passing it to USTM/255@REMOVED-0




<--- SIP read from 66.51.127.173:5060 --->
SIP/2.0 200 OK

Via: SIP/2.0/UDP 70.76.70.129:5060;branch=z9hG4bK30567deb;rport=60125

Record-Route: <sip:66.51.127.173;lr;ftag=as38dd9f02;nat=yes>

From: "REMOVED" <sip:3068806879@70.76.70.129>;tag=as38dd9f02

To: <sip:13068806879@REMOVED>;tag=ratD1gFZe9pQr

Call-ID: 6a40700c7124e70d310ad85c078acc63@70.76.70.129

CSeq: 103 INVITE

Contact: <sip:ciscosip@66.51.127.162:5060;transport=udp>

User-Agent: Cisco-SIPGateway/IOS-12.x

Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, INFO

Supported: timer, precondition, path, replaces

Allow-Events: talk, refer

Content-Type: application/sdp

Content-Disposition: session

Content-Length: 301



v=0

o=CiscoSystemsSIP-GW-UserAgent 7920449753154907727 7459957886221735931 IN IP4 66.51.127.162

s=SIP Call

c=IN IP4 66.51.127.173

t=0 0

m=audio 13304 RTP/AVP 0 101

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -

a=ptime:20

a=nortpproxy:yes


<------------->


--- (15 headers 12 lines) ---


Found RTP audio format 0


Found RTP audio format 101


Peer audio RTP is at port 66.51.127.173:13304


Found audio description format PCMU for ID 0


Found audio description format telephone-event for ID 101


Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)


Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)


Peer audio RTP is at port 66.51.127.173:13304


list_route: hop: <sip:66.51.127.173;lr;ftag=as38dd9f02;nat=yes>


set_destination: Parsing <sip:66.51.127.173;lr;ftag=as38dd9f02;nat=yes> for address/port to send to


set_destination: set destination to 66.51.127.173, port 5060


Transmitting (no NAT) to 66.51.127.173:5060:
ACK sip:ciscosip@66.51.127.162:5060;transport=udp SIP/2.0

Via: SIP/2.0/UDP 70.76.70.129:5060;branch=z9hG4bK3d54b97d;rport

Route: <sip:66.51.127.173;lr;ftag=as38dd9f02;nat=yes>

From: "REMOVED" <sip:3068806879@70.76.70.129>;tag=as38dd9f02

To: <sip:13068806879@REMOVED>;tag=ratD1gFZe9pQr

Contact: <sip:3068806879@70.76.70.129>

Call-ID: 6a40700c7124e70d310ad85c078acc63@70.76.70.129

CSeq: 103 ACK

User-Agent: Asterisk PBX

Max-Forwards: 70

Content-Length: 0



<--- SIP read from 66.51.127.173:5060 --->
BYE sip:3068806879@70.76.70.129:60125 SIP/2.0

Record-Route: <sip:66.51.127.173;lr;ftag=ratD1gFZe9pQr>

Via: SIP/2.0/UDP 66.51.127.173;branch=z9hG4bKa049.ce453293.0

Via: SIP/2.0/UDP 66.51.127.162;rport=5060;branch=z9hG4bKm2ty43gaFHDpS

Route: <sip:66.51.127.173;lr;ftag=as38dd9f02;nat=yes>

Max-Forwards: 69

From: <sip:13068806879@REMOVED>;tag=ratD1gFZe9pQr

To: "REMOVED" <sip:3068806879@70.76.70.129>;tag=as38dd9f02

Call-ID: 6a40700c7124e70d310ad85c078acc63@70.76.70.129

CSeq: 124042197 BYE

Contact: <sip:ciscosip@66.51.127.162:5060;transport=udp>

User-Agent: Cisco-SIPGateway/IOS-12.x

Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, INFO

Supported: timer, precondition, path, replaces

Reason: Q.850;cause=16;text="NORMAL_CLEARING"

Content-Length: 0


Sending to 66.51.127.173 : 5060 (no NAT)



<--- Transmitting (no NAT) to 66.51.127.173:5060 --->
SIP/2.0 200 OK

Via: SIP/2.0/UDP 66.51.127.173;branch=z9hG4bKa049.ce453293.0;received=66.51.127.173

Via: SIP/2.0/UDP 66.51.127.162;rport=5060;branch=z9hG4bKm2ty43gaFHDpS

Record-Route: <sip:66.51.127.173;lr;ftag=ratD1gFZe9pQr>

From: <sip:13068806879@REMOVED>;tag=ratD1gFZe9pQr

To: "REMOVED" <sip:3068806879@70.76.70.129>;tag=as38dd9f02

Call-ID: 6a40700c7124e70d310ad85c078acc63@70.76.70.129

CSeq: 124042197 BYE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Supported: replaces

Contact: <sip:3068806879@70.76.70.129>

Content-Length: 0

Reliably Transmitting (no NAT) to 66.51.110.210:5060:
REGISTER sip: SIP/2.0

Via: SIP/2.0/UDP 70.76.70.129:5060;branch=z9hG4bK2e435e8c;rport

From: <sip:REMOVED@>;tag=as055a6518

To: <sip:REMOVED@>

Call-ID: 0a82827d6a16833b52ccc0d9502d69bf@127.0.0.1

CSeq: 4286 REGISTER

User-Agent: Asterisk PBX

Max-Forwards: 70

Authorization: Digest username="REMOVED", realm="", algorithm=MD5, uri="sip:", nonce="4b1ef98a8cb0abc07cbff68967e3383c87072ee9", response="84e273b4eb91e21988b96bb625dee678"

Expires: 120

Contact: <sip:s@70.76.70.129>

Event: registration

Content-Length: 0



<--- SIP read from 66.51.110.210:5060 --->
SIP/2.0 100 Trying

Via: SIP/2.0/UDP 70.76.70.129:5060;branch=z9hG4bK2e435e8c;rport=64032

From: <sip:REMOVED@>;tag=as055a6518

To: <sip:REMOVED@>

Call-ID: 0a82827d6a16833b52ccc0d9502d69bf@127.0.0.1

CSeq: 4286 REGISTER

Content-Length: 0


<--- SIP read from 66.51.110.210:5060 --->
SIP/2.0 200 OK

Via: SIP/2.0/UDP 70.76.70.129:5060;branch=z9hG4bK2e435e8c;rport=64032

From: <sip:REMOVED@>;tag=as055a6518

To: <sip:REMOVED@>;tag=7a3986283a1d243977b41418b099c23d.f867

Call-ID: 0a82827d6a16833b52ccc0d9502d69bf@127.0.0.1

CSeq: 4286 REGISTER

Contact: <sip:s@70.76.70.129>;q=0;expires=120;received="sip:70.76.70.129:64032"

Content-Length: 0

____________________________________

The only glaring difference is useragent...

[williamconley]

a way to check would be to use "sip debug" to find the Real reject message (not the "translated for console" version).

non-working from the command line when "dialing":

Code: Select all

SIP/link2voip1/XXXXXXXXXX||tTor

if there is a difference in the structure of the dial request, that could set them apart and be part of the failure.

still missing these two items.

ok, we got most of it, but i only see one "dial" request, for the "not working", where is the "dial" for the working? i'm trying to compare those first before going any further, because if they are different there is no point in continuing if we have a dial requirement that is not being met.

i see:

Code: Select all

Executing Dial("Local/8600051@default-dc26,1", "SIP/lesnet/3069793286||tTor")

for "not-working", where is it for "working"? did i read too fast?

[panadyne]

all the examples that I have seen I have copied and pasted...

exten => _81NXXNXXXXXX,2,Dial(SIP/link2voip1/${EXTEN:2},,tTor)

to work required

exten => _81NXXNXXXXXX,2,Dial(SIP/link2voip1/${EXTEN:1},,tTor)

'one' not 'two' in the EXTEN...

[williamconley]

i am asking for a "dial" request from the working and the not working so that we can compare them.

[panadyne]

I have edited the two sip debug files in the above entry... and clearly indicated which is notworking /working...

I have also ensured that they are accessing the same carrier, but with different accounts... the working one will probably have some chaff, as there are more things going on in that server -- (the alternate carrier lesnet was presenting the same error with the same configuration, so I wasn't worried about which sip debug file I presented the last time...)

the nonworking server was sending an incorrectly formated number to the carrier based on the exten:2 variable -- when the enten:1 was correct.

Link2voip requires an 11 digit phone number be sent to it (with the leading 1NXXNXXXXXXX, otherwise it responds with the 503 error.

[williamconley]

! so it's working, there WAS a difference in the dial request and it was an easy fix. Cool.