Reg incoming call issue


#1

Hi friends,

i have asterisk and rregisterd with broadvoice and configured a extension. Here i am trying to make call from other pstn number to the sip broadvoice number… in the cli prompt it shows the following

-- Executing Dial("SIP/732xxxxxxx-c889", "SIP/$(EXTEN)@sip.broadvoice.com") in new stack
-- Called $(EXTEN)@sip.broadvoice.com
-- Got SIP response 480 "Temporarily Not Available" back from 147.135.8.128
-- SIP/sip.broadvoice.com-e066 is circuit-busy

== Everyone is busy/congested at this time (1:0/1/0)
– Executing Congestion(“SIP/732xxxxxxx-c889”, “”) in new stack
== Spawn extension (test, 732xxxxxxx, 2) exited non-zero on 'SIP/732xxxxxxx-c889

it seems connecting to asterisk and not able to reach the ext, i think and i following i provided the extension .com and sip.conf

extensions.conf

general]
static=yes
writepdotect=no
autofalthroguh=yes
clearglobalvars=no
[global]
Console=console/dsp

[test]
exten => _NXXNxxxxxx,1,dial(SIP/$ (EXTEN)@sip.broadvoice.com,30)
exten => _NXXNxxxxxx,2,congenstion()
exten => _NXXNxxxxxx,102,busy

exten => 1004,1,Dial(SIP/1,60,rT)
exten => 1005 ,1,Dial(SIP/2,60,rT)

and sip.conf

general]
context=default
bindport=5060
bindaddress=0.0.0.0
srvlookup=yes
pedantic=no

register=> number#@sip.broadvoice.com:password:number#@sip.broadvoice.com/1004

[sip.broadvocie.com]
type=peer
user=phone
host=sip.braodvoice.com
fromdomain=sip.broadvoice.com
fromuser=number#
secret=password
username=number#
insecure=very
context=test
authname=number#
dmf=inband
canreinvite=no

[1]
type=friend
secret =blak
host=dynamic
qualify=yes
conext=test
dtmfmode=rfc2833
canreinvite=no
allow=ulaw
[2]
type=friend
secret =blak
host=dynamic
qualify=yes
conext=test
dtmfmode=rfc2833
canreinvite=no
allow=ulaw

and asterisk is setup is public ip and the softphone too in public ip for testing the same.

can anyone find the bug and let me know the issue in my config please?


#2

try dialing it as 11 digit, ie 1-732-xxx-xxxx


#3

Hi iron,

it seems you did not get the issue i face…

when i try to dial 11 digit … the softphone says phone call failed :404 not found .

my first conern was to solve the icoming issue.

i am sure you will be helping me by your guidance.


#4

in your current configuration it would say 404 not found. exten => _NXXNxxxxxx won’t match an 11 digit number.

Copy the three extension lines and make them:

exten => _1NXXNxxxxxx
Also, not sure if its from retyping it, but the syntax and spelling is a bit off

You put:
exten => _NXXNxxxxxx,1,dial(SIP/$ (EXTEN)@sip.broadvoice.com,30)
exten => _NXXNxxxxxx,2,congenstion()
exten => _NXXNxxxxxx,102,busy

It should be:
exten => _NXXNXXXXXX,1,Dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten => _NXXNXXXXXX,2,Congestion()
exten => _NXXNXXXXXX,102,Congestion()

mainly note that its ${EXTEN} and not $ (EXTEN) (no space and needs {} not ())

If that doesn’t help, do a sip debug, make a call, and post the debug log here.


#5

Thanks friend,

Now i am able to make the calls from asterisk… but still i am not able to receive the call. more over whenever i try to dial the number which i got and registered with asterisk+broadvoice… from a pstn/cell phone … it plays voice mail tone from broad voice "The part you are trying to reach is busy and cannot take call so leave a msg "

i am sure you will be helping to resolve this issue.


#6

try sip show registry to see if you are correctly registered. If you are, what do you see on CLI when you dial? If nothing, do sip debug peer sip.broadvoice.com and look through that for clues, or post it here.


#7

Hi helix.

i am able to make outside calls from my softphone (extensions 1005). only problem is i am not able to receive the calls …

i am am very sure asterisk register with broadvocie . done sip show registry and checked again.

here i am posting the out sip debug peer sip.broadvoice.com

debug results, when dial 732xxxxxxx (sip broadvoice number ) from other pstn number (732xxxyyyy). it shows the status below for your guidance.

mail*CLI>
<-- SIP read from 147.135.8.128:5060:
INVITE sip:732xxxxxxx@aa.bb.cc.dd:5060 SIP/2.0
Call-ID: ff01f6-13@147.135.8.128
CSeq: 1 INVITE
From: "Cooperative Cmm"sip:732xxyyyy@147.135.8.128;user=phone;tag=dfhj
To: "Rizwan Mohammed"sip:1005@aa.bb.cc.dd;user=phone
Via: SIP/2.0/UDP 147.135.8.128:5060
Contact: sip:732xxyyyy@147.135.8.128:5060
Supported: 100rel
RPID-Privacy: party=calling;id-type=subscriber;privacy=off
Remote-Party-ID: sip:732xxyyyy@147.135.8.128;screen=yes;party=calling;privacy=off
Content-Length: 271
Content-Type: application/sdp

v=0
o=2475100407 10 10 IN IP4 147.135.8.247
s=-
c=IN IP4 147.135.8.250
t=0 0
m=audio 13318 RTP/AVP 0 8 2 18 96 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729/8000
a=rtpmap:96 iLBC/8000
a=rtpmap:101 telephone-event/8000

— (12 headers 12 lines)—
Using INVITE request as basis request - ff01f6-13@147.135.8.128
Sending to 147.135.8.128 : 5060 (NAT)
Found peer 'sip.broadvoice.com
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 2
Found RTP audio format 18
Found RTP audio format 96
Found RTP audio format 101
Peer audio RTP is at port 147.135.8.250:13318
Found description format PCMU
Found description format PCMA
Found description format G726-32
Found description format G729
Found description format iLBC
Found description format telephone-event
Capabilities: us - 0x4 (ulaw), peer - audio=0x51c (ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 732xxxxxxx in test (domain aa.bb.cc.dd)
Reliably Transmitting (NAT) to 147.135.8.128:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 147.135.8.128:5060;received=147.135.8.128
From: "Cooperative Cmm"sip:732xxyyyy@147.135.8.128;user=phone;tag=dfhj
To: "Rizwan Mohammed"sip:1005@aa.bb.cc.dd;user=phone;tag=as27743f41
Call-ID: ff01f6-13@147.135.8.128
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: sip:732xxxxxxx@aa.bb.cc.dd
Content-Length: 0


mail*CLI>
<-- SIP read from 147.135.8.128:5060:
INVITE sip:732xxxxxxx@aa.bb.cc.dd:5060 SIP/2.0
Call-ID: ff01f6-13@147.135.8.128
CSeq: 1 INVITE
From: "Cooperative Cmm"sip:732xxyyyy@147.135.8.128;user=phone;tag=dfhj
To: "Rizwan Mohammed"sip:1005@aa.bb.cc.dd;user=phone
Via: SIP/2.0/UDP 147.135.8.128:5060
Contact: sip:732xxyyyy@147.135.8.128:5060
Supported: 100rel
RPID-Privacy: party=calling;id-type=subscriber;privacy=off
Remote-Party-ID: sip:732xxyyyy@147.135.8.128;screen=yes;party=calling;privacy=off
Content-Length: 271
Content-Type: application/sdp

v=0
o=2475100407 10 10 IN IP4 147.135.8.247
s=-
c=IN IP4 147.135.8.250
t=0 0
m=audio 13318 RTP/AVP 0 8 2 18 96 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729/8000
a=rtpmap:96 iLBC/8000
a=rtpmap:101 telephone-event/8000

— (12 headers 12 lines)—
Ignoring this INVITE request
mail*CLI>
<-- SIP read from 147.135.8.128:5060:
ACK sip:1005@aa.bb.cc.dd:5060 SIP/2.0
Call-ID: ff01f6-13@147.135.8.128
CSeq: 1 ACK
From: "Cooperative Cmm"sip:732xxyyyy@147.135.8.128;user=phone;tag=dfhj
To: "Rizwan Mohammed"sip:1005@aa.bb.cc.dd;user=phone;tag=as27743f41
Via: SIP/2.0/UDP 147.135.8.128:5060;received=aa.bb.cc.dd
Content-Length: 0

— (7 headers 0 lines)—
Destroying call 'ff01f6-13@147.135.8.128’
Nov 30 03:03:34 NOTICE[12923]: chan_sip.c:5237 sip_reregister: – Re-registration for

732xxxxxxx@sip.broadvoice.com@sip.broadvoice.com
REGISTER 13 headers, 0 lines
Reliably Transmitting (NAT) to 147.135.8.128:5060:
REGISTER sip:sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP aa.bb.cc.dd:5060;branch=z9hG4bK15fe92f5;rport
From: sip:732xxxxxxx@sip.broadvoice.com;tag=as5769012a
To: sip:732xxxxxxx@sip.broadvoice.com
Call-ID: 7881e49e329e93148a62a1f49a@sip.broadvoice.com
CSeq: 345 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username=“732xxxxxxx”, realm=“BroadWorks”, algorithm=MD5, uri=“sip:sip.broadvoice.com”,

nonce=“1133294022851”, response=“2a06968f160f91fa21b2ba0ddba9”, opaque=""
Expires: 120
Contact: sip:1005@aa.bb.cc.dd
Event: registration
Content-Length: 0


mail*CLI>
<-- SIP read from 147.135.8.128:5060:
SIP/2.0 200 OK
Call-ID: 7881e49e329e931d06ea548a62a1f49a@sip.broadvoice.com
CSeq: 345 REGISTER
From: sip:732xxxxxxx@sip.broadvoice.com;tag=as5769012a
To: sip:732xxxxxxx@sip.broadvoice.com
Via: SIP/2.0/UDP sip.broadvoice.com:5060;branch=z9hG4bK15fe92f5
Contact: sip:1005@aa.bb.cc.dd
Expires: 30
Authorization: Digest username=“732xxxxxxx”, realm=“BroadWorks”, algorithm=MD5, uri=“sip:sip.broadvoice.com”,

nonce=“1133294022851”, response=“2a06968f160c55ff91fa21b2ba0ddba9”, opaque=""
Event: registration
User-Agent: Asterisk PBX
Content-Length: 0

— (12 headers 0 lines)—
Scheduling destruction of call '7881e49e329e931d06ea548a62a1f49a@sip.broadvoice.com’ in 32000 ms
Nov 30 03:03:35 NOTICE[12923]: chan_sip.c:9659 handle_response_register: Outbound Registration: Expiry for sip.broadvoice.com

is 30 sec (Scheduling reregistration in 23 s)
Destroying call '7881e49e329e931d06ea548a62a1f49a@sip.broadvoice.com
mail*CLI>

12 headers, 0 lines
Reliably Transmitting (NAT) to 147.135.8.128:5060:
OPTIONS sip:sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP aa.bb.cc.dd:5060;branch=z9hG4bK15b0d4bd;rport
From: “asterisk” sip:asterisk@aa.bb.cc.dd;tag=as0763c064
To: sip:sip.broadvoice.com
Contact: sip:asterisk@aa.bb.cc.dd
Call-ID: 610de7d257f6a8e647b62c977a76a276@aa.bb.cc.dd
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 29 Nov 2005 21:33:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


mail*CLI>
<-- SIP read from 147.135.8.128:5060:
SIP/2.0 200 OK
Call-ID: 610de7d257f6a8e647b62c977a76a276@aa.bb.cc.dd
CSeq: 102 OPTIONS
From: “asterisk” sip:asterisk@aa.bb.cc.dd;tag=as0763c064
To: sip:sip.broadvoice.com
Via: SIP/2.0/UDP aa.bb.cc.dd:5060;branch=z9hG4bK15b0d4bd
Supported: 100rel
Allow: INVITE, BYE, ACK, OPTIONS, CANCEL, PRACK
Accept: application/sdp
Accept-Encoding:
Accept-Language: en
Content-Length: 0

— (12 headers 0 lines)—
Destroying call '610de7d257f6a8e647b62c977a76a276@aa.bb.cc.dd
Nov 30 03:03:59 NOTICE[12923]: chan_sip.c:5237 sip_reregister: – Re-registration for

732xxxxxxx@sip.broadvoice.com@sip.broadvoice.com
REGISTER 13 headers, 0 lines
Reliably Transmitting (NAT) to 147.135.8.128:5060:
REGISTER sip:sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP aa.bb.cc.dd:5060;branch=z9hG4bK446b4110;rport
From: sip:732xxxxxxx@sip.broadvoice.com;tag=as6d9cedb9
To: sip:732xxxxxxx@sip.broadvoice.com
Call-ID: 7881e49e329e931d06ea548a62a1f49a@sip.broadvoice.com
CSeq: 346 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username=“732xxxxxxx”, realm=“BroadWorks”, algorithm=MD5, uri=“sip:sip.broadvoice.com”,

nonce=“1133294022851”, response=“2a06968f160c55ff91fa21b2ba0ddba9”, opaque=""
Expires: 120
Contact: sip:1005@aa.bb.cc.dd
Event: registration
Content-Length: 0


mail*CLI>
<-- SIP read from 147.135.8.128:5060:
SIP/2.0 200 OK
Call-ID: 7881e49e329e931d06ea548a62a1f49a@sip.broadvoice.com
CSeq: 346 REGISTER
From: sip:732xxxxxxx@sip.broadvoice.com;tag=as6d9cedb9
To: sip:732xxxxxxx@sip.broadvoice.com
Via: SIP/2.0/UDP sip.broadvoice.com:5060;branch=z9hG4bK446b4110
Contact: sip:1005@aa.bb.cc.dd
Expires: 30
Authorization: Digest username=“732xxxxxxx”, realm=“BroadWorks”, algorithm=MD5, uri=“sip:sip.broadvoice.com”,

nonce=“1133294022851”, response=“2a06968f160c55ff91fa21b2ba0ddba9”, opaque=""
Event: registration
User-Agent: Asterisk PBX
Content-Length: 0

mail*CLI>


#8

[quote]Looking for 732xxxxxxx in test (domain aa.bb.cc.dd)
Reliably Transmitting (NAT) to 147.135.8.128:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 147.135.8.128:5060;received=147.135.8.128
From: "Cooperative Cmm"sip:732xxyyyy@147.135.8.128;user=phone;tag=dfhj
To: "Rizwan Mohammed"sip:1005@aa.bb.cc.dd;user=phone;tag=as27743f41
Call-ID: ff01f6-13@147.135.8.128
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: sip:732xxxxxxx@aa.bb.cc.dd
Content-Length: 0
[/quote]

There’s the juicy bit. Add to your extensions.conf:

exten => 732xxxxxxx,1,Goto(1004,s,1)
and you should be all set (replace 732xxx with your bv number)


#9

Hi helix,

i included the same

exten => 732xxxxxxx,1,Goto(1004,s,1) in the extensions.conf of [test] section then reloaded the asterisk in cli prompt.

then i started calling from extern pstn number (732 xxx yyyy) to the
732 xxx xxxx … now i get single ring and went to broadvoice mail system .

732xxx xxxx -number i got from broadvoice.

As per suggestion , i did sip debug peer sip.broadvoice.com and found the result in cli prompts posted here for for your suggestion and help.

estroying call '031f19bd7588d724795c0bbb043a8eaa@sip.broadvoice.com
Nov 30 06:54:05 NOTICE[3879]: chan_sip.c:5237 sip_reregister: – Re-registration for 732xxxxxxx@sip.broadvoice.com@sip.broadvoice.com
REGISTER 13 headers, 0 lines
Reliably Transmitting (NAT) to 147.135.8.128:5060:
REGISTER sip:sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP aa.bb.cc.dd:5060;branch=z9hG4bK605edf9b;rport
From: sip:732xxxxxxx@sip.broadvoice.com;tag=as5a4be902
To: sip:732xxxxxxx@sip.broadvoice.com
Call-ID: 031f19bd7588d724795c0bbb043a8eaa@sip.broadvoice.com
CSeq: 149 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username=“732xxxxxxx”, realm=“BroadWorks”, algorithm=MD5, uri=“sip:sip.broadvoice.com”, nonce=“1133311925355”, response=“20bfde78c7b491ad42a7226004e18e1b”, opaque=""
Expires: 120
Contact: sip:1004@aa.bb.cc.dd
Event: registration
Content-Length: 0


<-- SIP read from 147.135.8.128:5060:
SIP/2.0 200 OK
Call-ID: 031f19bd7588d724795c0bbb043a8eaa@sip.broadvoice.com
CSeq: 149 REGISTER
From: sip:732xxxxxxx@sip.broadvoice.com;tag=as5a4be902
To: sip:732xxxxxxx@sip.broadvoice.com
Via: SIP/2.0/UDP sip.broadvoice.com:5060;branch=z9hG4bK605edf9b
Contact: sip:1004@aa.bb.cc.dd
Expires: 30
Authorization: Digest username=“732xxxxxxx”, realm=“BroadWorks”, algorithm=MD5, uri=“sip:sip.broadvoice.com”, nonce=“1133311925355”, response=“20bfde78c7b491ad42a7226004e18e1b”, opaque=""
Event: registration
User-Agent: Asterisk PBX
Content-Length: 0

— (12 headers 0 lines)—
Scheduling destruction of call '031f19bd7588d724795c0bbb043a8eaa@sip.broadvoice.com’ in 32000 ms
Nov 30 06:54:05 NOTICE[3879]: chan_sip.c:9659 handle_response_register: Outbound Registration: Expiry for sip.broadvoice.com is 30 sec (Scheduling reregistration in 23 s)
Destroying call '031f19bd7588d724795c0bbb043a8eaa@sip.broadvoice.com

12 headers, 0 lines
Reliably Transmitting (NAT) to 147.135.8.128:5060:
OPTIONS sip:sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP aa.bb.cc.dd:5060;branch=z9hG4bK5e4bb6e6;rport
From: “asterisk” sip:asterisk@aa.bb.cc.dd;tag=as77ac575a
To: sip:sip.broadvoice.com
Contact: sip:asterisk@aa.bb.cc.dd
Call-ID: 43fe63126d2e913d7f18ca6f308259a9@aa.bb.cc.dd
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 30 Nov 2005 01:24:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


Nov 30 06:54:29 NOTICE[3879]: chan_sip.c:5237 sip_reregister: – Re-registration for 732xxxxxxx@sip.broadvoice.com@sip.broadvoice.com
REGISTER 13 headers, 0 lines
Reliably Transmitting (NAT) to 147.135.8.128:5060:
REGISTER sip:sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP aa.bb.cc.dd:5060;branch=z9hG4bK0dc834f8;rport
From: sip:732xxxxxxx@sip.broadvoice.com;tag=as349972ee
To: sip:732xxxxxxx@sip.broadvoice.com
Call-ID: 031f19bd7588d724795c0bbb043a8eaa@sip.broadvoice.com
CSeq: 150 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username=“732xxxxxxx”, realm=“BroadWorks”, algorithm=MD5, uri=“sip:sip.broadvoice.com”, nonce=“1133311925355”, response=“20bfde78c7b491ad42a7226004e18e1b”, opaque=""
Expires: 120
Contact: sip:1004@aa.bb.cc.dd
Event: registration
Content-Length: 0


<-- SIP read from 147.135.8.128:5060:
SIP/2.0 200 OK
Call-ID: 43fe63126d2e913d7f18ca6f308259a9@aa.bb.cc.dd
CSeq: 102 OPTIONS
From: “asterisk” sip:asterisk@aa.bb.cc.dd;tag=as77ac575a
To: sip:sip.broadvoice.com
Via: SIP/2.0/UDP aa.bb.cc.dd:5060;branch=z9hG4bK5e4bb6e6
Supported: 100rel
Allow: INVITE, BYE, ACK, OPTIONS, CANCEL, PRACK
Accept: application/sdp
Accept-Encoding:
Accept-Language: en
Content-Length: 0

— (12 headers 0 lines)—
Destroying call '43fe63126d2e913d7f18ca6f308259a9@aa.bb.cc.dd

<-- SIP read from 147.135.8.128:5060:
SIP/2.0 200 OK
Call-ID: 031f19bd7588d724795c0bbb043a8eaa@sip.broadvoice.com
CSeq: 150 REGISTER
From: sip:732xxxxxxx@sip.broadvoice.com;tag=as349972ee
To: sip:732xxxxxxx@sip.broadvoice.com
Via: SIP/2.0/UDP sip.broadvoice.com:5060;branch=z9hG4bK0dc834f8
Contact: sip:1004@aa.bb.cc.dd
Expires: 30
Authorization: Digest username=“732xxxxxxx”, realm=“BroadWorks”, algorithm=MD5, uri=“sip:sip.broadvoice.com”, nonce=“1133311925355”, response=“20bfde78c7b491ad42a7226004e18e1b”, opaque=""
Event: registration
User-Agent: Asterisk PBX
Content-Length: 0

— (12 headers 0 lines)—
Scheduling destruction of call '031f19bd7588d724795c0bbb043a8eaa@sip.broadvoice.com’ in 32000 ms
Nov 30 06:54:30 NOTICE[3879]: chan_sip.c:9659 handle_response_register: Outbound Registration: Expiry for sip.broadvoice.com is 30 sec (Scheduling reregistration in 23 s)

<-- SIP read from 147.135.8.128:5060:
INVITE sip:732xxxxxxx@aa.bb.cc.dd:5060 SIP/2.0
Call-ID: ff032e-42@147.135.8.128
CSeq: 1 INVITE
From: "E Pro Inc"sip:732xxxyyyy@147.135.8.128;user=phone;tag=4689
To: "Rizwan Mohammed"sip:1004@aa.bb.cc.dd;user=phone
Via: SIP/2.0/UDP 147.135.8.128:5060
Contact: sip:732xxxyyyy@147.135.8.128:5060
Supported: 100rel
RPID-Privacy: party=calling;id-type=subscriber;privacy=off
Remote-Party-ID: sip:732xxxyyyy@147.135.8.128;screen=yes;party=calling;privacy=off
Content-Length: 271
Content-Type: application/sdp

v=0
o=2475100407 10 10 IN IP4 147.135.8.247
s=-
c=IN IP4 147.135.8.248
t=0 0
m=audio 18940 RTP/AVP 0 8 2 18 96 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729/8000
a=rtpmap:96 iLBC/8000
a=rtpmap:101 telephone-event/8000

— (12 headers 12 lines)—
Using INVITE request as basis request - ff032e-42@147.135.8.128
Sending to 147.135.8.128 : 5060 (NAT)
Found peer 'sip.broadvoice.com
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 2
Found RTP audio format 18
Found RTP audio format 96
Found RTP audio format 101
Peer audio RTP is at port 147.135.8.248:18940
Found description format PCMU
Found description format PCMA
Found description format G726-32
Found description format G729
Found description format iLBC
Found description format telephone-event
Capabilities: us - 0x4 (ulaw), peer - audio=0x51c (ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 732xxxxxxx in test (domain aa.bb.cc.dd)
Reliably Transmitting (NAT) to 147.135.8.128:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 147.135.8.128:5060;received=147.135.8.128
From: "E Pro Inc"sip:732xxxyyyy@147.135.8.128;user=phone;tag=4689
To: "Rizwan Mohammed"sip:1004@aa.bb.cc.dd;user=phone;tag=as3312c0ad
Call-ID: ff032e-42@147.135.8.128
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: sip:732xxxxxxx@aa.bb.cc.dd
Content-Length: 0


<-- SIP read from 147.135.8.128:5060:
ACK sip:1004@aa.bb.cc.dd:5060 SIP/2.0
Call-ID: ff032e-42@147.135.8.128
CSeq: 1 ACK
From: "E Pro Inc"sip:732xxxyyyy@147.135.8.128;user=phone;tag=4689
To: "Rizwan Mohammed"sip:1004@aa.bb.cc.dd;user=phone;tag=as3312c0ad
Via: SIP/2.0/UDP 147.135.8.128:5060;received=aa.bb.cc.dd
Content-Length: 0

— (7 headers 0 lines)—
Destroying call ‘ff032e-42@147.135.8.128’
Destroying call '031f19bd7588d724795c0bbb043a8eaa@sip.broadvoice.com

Expecting your kind help to resolv the incoming issue.


#10

please let me know any more info need to fix this issue

[/quote]


#11

Looking for 732xxxxxxx in test (domain aa.bb.cc.dd)
Reliably Transmitting (NAT) to 147.135.8.128:5060:
SIP/2.0 404 Not Found

for the domain thing, did you turn on domain support? try adding domain=no to sip.conf…
also dump the /1004 at the end of your register line and see what happens


#12

hi helix.

i did it and i did not get through and showed this while debugging

ist_route: hop: sip:7322830802@147.135.8.128:5060
Transmitting (NAT) to 147.135.8.128:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 147.135.8.128:5060;received=147.135.8.128
From: "E Pro Inc"sip:7322830802@147.135.8.128;user=phone;tag=adfg
To: "Rizwan Mohammed"sip:1004@202.54.191.71;user=phone
Call-ID: ff01e1-66@147.135.8.128
CSeq: 1 INVITE
User-Agent: Asterisk PBX

please look into that and favour me


#13

hi helix

This is my latest extenions.conf

[default]
exten => 732xxxxxxx,1,Goto(test,s,1)

[test]
exten => s,1,Answer()
exten => s,2,Dial(SIP/raju,60,r)
exten => s,3,Hangup()

exten => _NXXNXXXXXX,1,Dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten => _NXXNXXXXXX,2,Congestion()
exten => _NXXNXXXXXX,102,Busy()

exten => 1004,1,Dial(SIP/rizwan,60,r)
exten => 1004,2,VoiceMail(1004@mb_test)
exten => 1004,3,PlayBack(vm-goodbye)
exten => 1004,4,Hangup()

exten => 1005,1,Dial(SIP/raju,60,r)
exten => 1005,2,VoiceMail(1005@mb_test)
exten => 1005,3,PlayBack(vm-goodbye)
exten => 1005,4,Hangup()

exten => 850,1,Answer()
exten => 850,2,VoiceMailMain(@mb_test)
exten => 850,3 Hangup()

and error i got is in cli prompt is

mailCLI> set verbose 78
Verbosity is at least 78
– Executing Dial(“SIP/732xxxxxxx-f30a”, “SIP/732xxxxxxx@sip.broadvoice.com|30”) in new stack
– Called 732xxxxxxx@sip.broadvoice.com
– Got SIP response 480 “Temporarily Not Available” back from 147.135.8.128
– SIP/sip.broadvoice.com-c8dc is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
– Executing Busy(“SIP/732xxxxxxx-f30a”, “”) in new stack
== Spawn extension (test, 732xxxxxxx, 102) exited non-zero on 'SIP/732xxxxxxx-f30a’
mail
CLI> exit

looking for your help and sorry any confusion i did


#14

You have created a loop- a call comes in on bv to the bv number, which matches your outgoing call extension so it makes broadvoice try to dial itself. bv gets confused and gives up.

in sip.conf, for broadvoice set context=default and you should be good.


#15

Hi Helix,

Thanks for you kind help and giudance and got through it now after the changes you asked me do !!

Cheers