Re: [Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server
I have made all the changes to sip.conf for my broadvoice peer
friend(and I have tried it as peer) and I am still seeing this response
(on call out). Any suggestions? I don't think it is a problem with the
phones themselves authenticating, as Asterisk takes care of all the
authentication from my understanding.
Free world does work for calling out however. So I know at least that
works.
-- Got SIP response 400 Bad request back from 147.135.0.128
Mar 9 01:11:09 NOTICE[12284]: chan_sip.c:6798 handle_response: Failed
to authenticate on INVITE to 'PP
sip:[EMAIL PROTECTED];tag=as5b80cade'
On Wed, 2005-03-09 at 00:48, Chris Nibeck wrote:
First off... please cancel previous amplification request. I have
implemented your ideas with the same errored result.
I am not sure that we are not making it thru authentication. From my
digging and comparing packet dumps comparing the soft phone to asterisk
they have identical transactions through the ACK reply (the last one
on the debug below). The softphone seems to be authenticated after the
ACK. I am a newbie to debugging this stuff. I just want to get it
working.
Thanks everyone in advance for your help. I am certainly very very
happy to try anything.
Based on Luki's suggestions I...
Changed sip.conf...
[broadvoice1]
type=peer
;user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=8475100139
secret=zjh018g8f8
username=8475100139
insecure=very
context=default
authname=8475100139
dtmfmode=inband
dtmf=inband
;Disable canreinvite if you are behind a NAT
canreinvite=no
nat=no
Changed extensions.conf...
exten = _8X.,1, dial(SIP/${EXTEN:[EMAIL PROTECTED],30) ; Dial Broadvoice
for 30 seconds
exten = _8X.,2, congestion() ; No answer, nothing
exten = _8X., 102, busy() ;
End result...
Mar 9 01:30:42 NOTICE[15376]: chan_sip.c:5047 handle_response: Failed
to authenticate on INVITE to '6050
sip:[EMAIL PROTECTED];tag=as545ccba3'
SIP debug...
-- Executing Dial(SIP/6050-132b,
SIP/[EMAIL PROTECTED]|30) in new stack
We're at xxx.xxx.xxx.xxx port 18212
Answering with capability 2
Answering with capability 4
Answering with capability 8
12 headers, 10 lines
Reliably Transmitting:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0
From: 6050 sip:[EMAIL PROTECTED];tag=as545ccba3
To: sip:[EMAIL PROTECTED]
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Wed, 09 Mar 2005 07:30:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 205
v=0
o=root 3998 3998 IN IP4 xxx.xxx.xxx.xxx
s=session
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 18212 RTP/AVP 3 0 8
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
(no NAT) to 147.135.8.128:5060
-- Called [EMAIL PROTECTED]
com*CLI
Sip read:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 64.4.192.110:5060;branch=z9hG4bK-1b5b3221
From: 6050 sip:[EMAIL PROTECTED];tag=7e2776985d5a0891o0
To: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest
username=6050,realm=asterisk,nonce=42d82e9b,uri=sip:
[EMAIL PROTECTED],algorithm=MD5,response=420e39b35648a10c
129dd4fb5f97ec47
Contact: 6050 sip:[EMAIL PROTECTED]:5060
Expires: 240
User-Agent: Sipura/SPA3000-2.0.10(GWf)
Content-Length: 241
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp
v=0
o=- 1138990026 1138990026 IN IP4 64.4.192.110
s=-
c=IN IP4 64.4.192.110
t=0 0
m=audio 16388 RTP/AVP 0 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
15 headers, 12 lines
Ignoring this request
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 64.4.192.110:5060;branch=z9hG4bK-1b5b3221
From: 6050 sip:[EMAIL PROTECTED];tag=7e2776985d5a0891o0
To: sip:[EMAIL PROTECTED];tag=as2f065f18
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:[EMAIL PROTECTED]
Content-Length: 0
to 64.4.192.110:5060
com*CLI
Sip read:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0
From: 6050 sip:[EMAIL PROTECTED];tag=as545ccba3
To: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
6 headers, 0 lines
com*CLI
Sip read:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0
From: 6050 sip:[EMAIL PROTECTED];tag=as545ccba3
To: sip:[EMAIL PROTECTED];tag=SD38rq699-
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
WWW-Authenticate: DIGEST
realm=BroadWorks,algorithm=MD5,nonce=1110353299563
Content-Length: 0
8 headers, 0 lines
Transmitting:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Re: [Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server
Try changing the extension from Broadvoice1 to the actual phone number
(and don't send your secret in a public email or maybe that's Chris'):
[*8475100139*]
type=peer
;user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=8475100139
secret=XXX
username=8475100139
Zanzamar Majere wrote:
I have made all the changes to sip.conf for my broadvoice peer
friend(and I have tried it as peer) and I am still seeing this response
(on call out). Any suggestions? I don't think it is a problem with the
phones themselves authenticating, as Asterisk takes care of all the
authentication from my understanding.
Free world does work for calling out however. So I know at least that
works.
-- Got SIP response 400 Bad request back from 147.135.0.128
Mar 9 01:11:09 NOTICE[12284]: chan_sip.c:6798 handle_response: Failed
to authenticate on INVITE to 'PP
sip:[EMAIL PROTECTED];tag=as5b80cade'
On Wed, 2005-03-09 at 00:48, Chris Nibeck wrote:
First off... please cancel previous amplification request. I have
implemented your ideas with the same errored result.
I am not sure that we are not making it thru authentication. From my
digging and comparing packet dumps comparing the soft phone to asterisk
they have identical transactions through the ACK reply (the last one
on the debug below). The softphone seems to be authenticated after the
ACK. I am a newbie to debugging this stuff. I just want to get it
working.
Thanks everyone in advance for your help. I am certainly very very
happy to try anything.
Based on Luki's suggestions I...
Changed sip.conf...
[broadvoice1]
type=peer
;user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=8475100139
secret=DELETED
username=8475100139
insecure=very
context=default
authname=8475100139
dtmfmode=inband
dtmf=inband
;Disable canreinvite if you are behind a NAT
canreinvite=no
nat=no
Changed extensions.conf...
exten = _8X.,1, dial(SIP/${EXTEN:[EMAIL PROTECTED],30) ; Dial Broadvoice
for 30 seconds
exten = _8X.,2, congestion() ; No answer, nothing
exten = _8X., 102, busy() ;
End result...
Mar 9 01:30:42 NOTICE[15376]: chan_sip.c:5047 handle_response: Failed
to authenticate on INVITE to '6050
sip:[EMAIL PROTECTED];tag=as545ccba3'
SIP debug...
-- Executing Dial(SIP/6050-132b,
SIP/[EMAIL PROTECTED]|30) in new stack
We're at xxx.xxx.xxx.xxx port 18212
Answering with capability 2
Answering with capability 4
Answering with capability 8
12 headers, 10 lines
Reliably Transmitting:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0
From: 6050 sip:[EMAIL PROTECTED];tag=as545ccba3
To: sip:[EMAIL PROTECTED]
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Wed, 09 Mar 2005 07:30:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 205
v=0
o=root 3998 3998 IN IP4 xxx.xxx.xxx.xxx
s=session
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 18212 RTP/AVP 3 0 8
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
(no NAT) to 147.135.8.128:5060
-- Called [EMAIL PROTECTED]
com*CLI
Sip read:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 64.4.192.110:5060;branch=z9hG4bK-1b5b3221
From: 6050 sip:[EMAIL PROTECTED];tag=7e2776985d5a0891o0
To: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest
username=6050,realm=asterisk,nonce=42d82e9b,uri=sip:
[EMAIL PROTECTED],algorithm=MD5,response=420e39b35648a10c
129dd4fb5f97ec47
Contact: 6050 sip:[EMAIL PROTECTED]:5060
Expires: 240
User-Agent: Sipura/SPA3000-2.0.10(GWf)
Content-Length: 241
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp
v=0
o=- 1138990026 1138990026 IN IP4 64.4.192.110
s=-
c=IN IP4 64.4.192.110
t=0 0
m=audio 16388 RTP/AVP 0 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
15 headers, 12 lines
Ignoring this request
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 64.4.192.110:5060;branch=z9hG4bK-1b5b3221
From: 6050 sip:[EMAIL PROTECTED];tag=7e2776985d5a0891o0
To: sip:[EMAIL PROTECTED];tag=as2f065f18
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:[EMAIL PROTECTED]
Content-Length: 0
to 64.4.192.110:5060
com*CLI
Sip read:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0
From: 6050 sip:[EMAIL PROTECTED];tag=as545ccba3
To: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
6 headers, 0 lines
com*CLI
Sip read:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0
From: 6050 sip:[EMAIL PROTECTED];tag=as545ccba3
To: sip:[EMAIL PROTECTED];tag=SD38rq699-
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
WWW-Authenticate: DIGEST
Re: [Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server
Thank you for the response. I still have the errors mentioned below, sip
response and Failed to authenticate on INVITE
[PP]
type=peer
username=PP
fromuser=PP
authuser=PP
fromdomain=sip.broadvoice.com
secret=XX
host=sip.broadvoice.com
dtmfmode=inband
insecure=very
context=sip
qualify=yes
disallow=all
allow=ulaw
allow=gsm
;Disable canreinvite if you are behind a NAT
;canreinvite=no
nat=no
Does anyone else have any other suggestions?
On Wednesday 09 March 2005 06:56 am, MF Hulber wrote:
Try changing the extension from Broadvoice1 to the actual phone number
(and don't send your secret in a public email or maybe that's Chris'):
[*8475100139*]
type=peer
;user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=8475100139
secret=XXX
username=8475100139
Zanzamar Majere wrote:
I have made all the changes to sip.conf for my broadvoice peer
friend(and I have tried it as peer) and I am still seeing this response
(on call out). Any suggestions? I don't think it is a problem with the
phones themselves authenticating, as Asterisk takes care of all the
authentication from my understanding.
Free world does work for calling out however. So I know at least that
works.
-- Got SIP response 400 Bad request back from 147.135.0.128
Mar 9 01:11:09 NOTICE[12284]: chan_sip.c:6798 handle_response: Failed
to authenticate on INVITE to 'PP
sip:[EMAIL PROTECTED];tag=as5b80cade'
On Wed, 2005-03-09 at 00:48, Chris Nibeck wrote:
First off... please cancel previous amplification request. I have
implemented your ideas with the same errored result.
I am not sure that we are not making it thru authentication. From my
digging and comparing packet dumps comparing the soft phone to asterisk
they have identical transactions through the ACK reply (the last one
on the debug below). The softphone seems to be authenticated after the
ACK. I am a newbie to debugging this stuff. I just want to get it
working.
Thanks everyone in advance for your help. I am certainly very very
happy to try anything.
Based on Luki's suggestions I...
Changed sip.conf...
[broadvoice1]
type=peer
;user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=8475100139
secret=DELETED
username=8475100139
insecure=very
context=default
authname=8475100139
dtmfmode=inband
dtmf=inband
;Disable canreinvite if you are behind a NAT
canreinvite=no
nat=no
Changed extensions.conf...
exten = _8X.,1, dial(SIP/${EXTEN:[EMAIL PROTECTED],30) ; Dial Broadvoice
for 30 seconds
exten = _8X.,2, congestion() ; No answer, nothing
exten = _8X., 102, busy() ;
End result...
Mar 9 01:30:42 NOTICE[15376]: chan_sip.c:5047 handle_response: Failed
to authenticate on INVITE to '6050
sip:[EMAIL PROTECTED];tag=as545ccba3'
SIP debug...
-- Executing Dial(SIP/6050-132b,
SIP/[EMAIL PROTECTED]|30) in new stack
We're at xxx.xxx.xxx.xxx port 18212
Answering with capability 2
Answering with capability 4
Answering with capability 8
12 headers, 10 lines
Reliably Transmitting:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0
From: 6050 sip:[EMAIL PROTECTED];tag=as545ccba3
To: sip:[EMAIL PROTECTED]
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Wed, 09 Mar 2005 07:30:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 205
v=0
o=root 3998 3998 IN IP4 xxx.xxx.xxx.xxx
s=session
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 18212 RTP/AVP 3 0 8
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
(no NAT) to 147.135.8.128:5060
-- Called [EMAIL PROTECTED]
com*CLI
Sip read:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 64.4.192.110:5060;branch=z9hG4bK-1b5b3221
From: 6050 sip:[EMAIL PROTECTED];tag=7e2776985d5a0891o0
To: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest
username=6050,realm=asterisk,nonce=42d82e9b,uri=sip:
[EMAIL PROTECTED],algorithm=MD5,response=420e39b35648a10c
129dd4fb5f97ec47
Contact: 6050 sip:[EMAIL PROTECTED]:5060
Expires: 240
User-Agent: Sipura/SPA3000-2.0.10(GWf)
Content-Length: 241
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp
v=0
o=- 1138990026 1138990026 IN IP4 64.4.192.110
s=-
c=IN IP4 64.4.192.110
t=0 0
m=audio 16388 RTP/AVP 0 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
15 headers, 12 lines
Ignoring this request
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 64.4.192.110:5060;branch=z9hG4bK-1b5b3221
From: 6050 sip:[EMAIL PROTECTED];tag=7e2776985d5a0891o0
To: sip:[EMAIL PROTECTED];tag=as2f065f18
Call-ID: [EMAIL
Re: [Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server
Have you tried this: http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup Zanzamar Majere wrote: Thank you for the response. I still have the errors mentioned below, sip response and Failed to authenticate on INVITE [PP] type=peer username=PP fromuser=PP authuser=PP fromdomain=sip.broadvoice.com secret=XX host=sip.broadvoice.com dtmfmode=inband insecure=very context=sip qualify=yes disallow=all allow=ulaw allow=gsm ;Disable canreinvite if you are behind a NAT ;canreinvite=no nat=no Does anyone else have any other suggestions? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broadvoice latest changes and still not working-An Additional Server
Jerry- Thank you I accidently sent my password on the LISTSERV last night so I just changed (pasted) the new one in. Still the same problem... Mar 9 09:51:13 NOTICE[15376]: chan_sip.c:5047 handle_response: Failed to authenticate on INVITE to 'Chris Nibeck sip:[EMAIL PROTECTED];tag=as4b70f2e7' Incoming works fine still. Anyone can call me at that number. Please do. It is a free call from another BV account. Chris On Mar 9, 2005, at 7:42 AM, Jerry Geis wrote: CHris, I had the exact same problem with the exact same error. My password was entered incorrectly in context section. The register line had the correct password. That is why you get incoming calls. and not outgoing. Jerry ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server
Thanks MF,
Yes that was me that sent my PW :-) It is changed now.
Same error...
Mar 9 10:12:46 NOTICE[15376]: chan_sip.c:5047 handle_response: Failed
to authenticate on INVITE to 'Chris Nibeck
sip:[EMAIL PROTECTED];tag=as0cefa74c'
Sip.conf...
[*8475100139*]
type=peer
;user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=8475100139
secret=x
username=8475100139
insecure=very
context=default
authname=8475100139
dtmfmode=inband
dtmf=inband
;Disable canreinvite if you are behind a NAT
canreinvite=no
nat=no
extensions.conf...
exten = _8X.,1, dial(SIP/${EXTEN:[EMAIL PROTECTED],30) ; Dial Broadvoice
for 30 seconds
exten = _8X.,2, congestion() ; No answer, nothing
exten = _8X., 102, busy() ;
On Mar 9, 2005, at 7:56 AM, MF Hulber wrote:
Try changing the extension from Broadvoice1 to the actual phone number
(and don't send your secret in a public email or maybe that's Chris'):
[*8475100139*]
type=peer
;user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=8475100139
secret=XXX
username=8475100139
Zanzamar Majere wrote:
I have made all the changes to sip.conf for my broadvoice peer
friend(and I have tried it as peer) and I am still seeing this
response
(on call out). Any suggestions? I don't think it is a problem with
the
phones themselves authenticating, as Asterisk takes care of all the
authentication from my understanding.
Free world does work for calling out however. So I know at least that
works.
-- Got SIP response 400 Bad request back from 147.135.0.128
Mar 9 01:11:09 NOTICE[12284]: chan_sip.c:6798 handle_response: Failed
to authenticate on INVITE to 'PP
sip:[EMAIL PROTECTED];tag=as5b80cade'
On Wed, 2005-03-09 at 00:48, Chris Nibeck wrote:
First off... please cancel previous amplification request. I have
implemented your ideas with the same errored result.
I am not sure that we are not making it thru authentication. From
my digging and comparing packet dumps comparing the soft phone to
asterisk they have identical transactions through the ACK reply
(the last one on the debug below). The softphone seems to be
authenticated after the ACK. I am a newbie to debugging this
stuff. I just want to get it working.
Thanks everyone in advance for your help. I am certainly very very
happy to try anything.
Based on Luki's suggestions I...
Changed sip.conf...
[broadvoice1]
type=peer
;user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=8475100139
secret=DELETED
username=8475100139
insecure=very
context=default
authname=8475100139
dtmfmode=inband
dtmf=inband
;Disable canreinvite if you are behind a NAT
canreinvite=no
nat=no
Changed extensions.conf...
exten = _8X.,1, dial(SIP/${EXTEN:[EMAIL PROTECTED],30) ; Dial
Broadvoice for 30 seconds
exten = _8X.,2, congestion() ; No answer, nothing
exten = _8X., 102, busy() ;
End result...
Mar 9 01:30:42 NOTICE[15376]: chan_sip.c:5047 handle_response:
Failed to authenticate on INVITE to '6050
sip:[EMAIL PROTECTED];tag=as545ccba3'
SIP debug...
-- Executing Dial(SIP/6050-132b,
SIP/[EMAIL PROTECTED]|30) in new stack
We're at xxx.xxx.xxx.xxx port 18212
Answering with capability 2
Answering with capability 4
Answering with capability 8
12 headers, 10 lines
Reliably Transmitting:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0
From: 6050 sip:[EMAIL PROTECTED];tag=as545ccba3
To: sip:[EMAIL PROTECTED]
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Wed, 09 Mar 2005 07:30:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 205
v=0
o=root 3998 3998 IN IP4 xxx.xxx.xxx.xxx
s=session
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 18212 RTP/AVP 3 0 8
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
(no NAT) to 147.135.8.128:5060
-- Called [EMAIL PROTECTED]
com*CLI
Sip read:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 64.4.192.110:5060;branch=z9hG4bK-1b5b3221
From: 6050 sip:[EMAIL PROTECTED];tag=7e2776985d5a0891o0
To: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest
username=6050,realm=asterisk,nonce=42d82e9b,uri=sip:
[EMAIL PROTECTED],algorithm=MD5,response=420e39b35648a
10c 129dd4fb5f97ec47
Contact: 6050 sip:[EMAIL PROTECTED]:5060
Expires: 240
User-Agent: Sipura/SPA3000-2.0.10(GWf)
Content-Length: 241
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp
v=0
o=- 1138990026 1138990026 IN IP4 64.4.192.110
s=-
c=IN IP4 64.4.192.110
t=0 0
m=audio 16388 RTP/AVP 0 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
15 headers, 12 lines
Ignoring this request
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 64.4.192.110:5060;branch=z9hG4bK-1b5b3221
Re: [Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server
there are two of us with the same problem so I will answer for me. Yes I tried the below instructions. The current thinking by multiple people is * never tries authenticating so removing the FQDN will force * to go to the related section named by either a phone number or a non Fully Qualified Domain Name. But I still don't have it working so who knows. Anyone that wishes to call me via BV my number is 8475100139 and it is up. Chris On Mar 9, 2005, at 9:23 AM, Mike Matthews wrote: Have you tried this: http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup Zanzamar Majere wrote: Thank you for the response. I still have the errors mentioned below, sip response and Failed to authenticate on INVITE [PP] type=peer username=PP fromuser=PP authuser=PP fromdomain=sip.broadvoice.com secret=XX host=sip.broadvoice.com dtmfmode=inband insecure=very context=sip qualify=yes disallow=all allow=ulaw allow=gsm ;Disable canreinvite if you are behind a NAT ;canreinvite=no nat=no Does anyone else have any other suggestions? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Fwd: Re: [Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server ****SOLVED****
This configuration solved my problem. I could have sworn I tried this before. I guess not. I did not need to apply the patch. Also, I am using a regular Registration setup in my sip.conf not broadvoice's funky one... The only thing I can surmise is that order of the variables matters. This is what worked for me: [PP] type=peer user=phone host=sip.broadvoice.com fromdomain=sip.broadvoice.com fromuser=PP secret=XX username=PP insecure=very context=sip authname=PP dtmfmode=inband dtmf=inband ;Disable canreinvite if you are behind a NAT canreinvite=no Thank you On Wednesday 09 March 2005 08:23 am, Mike Matthews wrote: Have you tried this: http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup Zanzamar Majere wrote: Thank you for the response. I still have the errors mentioned below, sip response and Failed to authenticate on INVITE [PP] type=peer username=PP fromuser=PP authuser=PP fromdomain=sip.broadvoice.com secret=XX host=sip.broadvoice.com dtmfmode=inband insecure=very context=sip qualify=yes disallow=all allow=ulaw allow=gsm ;Disable canreinvite if you are behind a NAT ;canreinvite=no nat=no Does anyone else have any other suggestions? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broadvoice latest changes and still not working-An
Just wondering. How are you getting this debug. I
am having problems to and I cant seem to track it down.
- Original Message -
From:
Joe
To: asterisk-users@lists.digium.com
Sent: Wednesday, March 09, 2005 10:41
AM
Subject: [Asterisk-Users] Broadvoice
latest changes and still not working-An
Ive tried everything with the * box after this
weekend. I have read every
document on the problems people are having with them after this weekend as
well, but none of them address my problem.
I checked my settings in my sips which I have below as
well,
I have changed the host file a few times, but this was new to me and I never had
modified it before. I have and
the same results happened.
I have always used the CHI proxy until this past
weekend.
I get a 404 not found when the invite goes out.
Below is my debug for broadvoice, which I think tells the whole
story, but for the life of me, I
can not figure out where the 404 is coming from.
I have listed my sip file below as
well.
Inbound calls work and I am
registered.
Before we go into the debug, I get this message when I reload my
configs files.
Mar 9 13:23:48 NOTICE[3161]: chan_sip.c:6779 handle_response:
Outbound Registration: Expiry for sip.broadvoice.com is 1948 sec (Scheduling
reregistration in 1933000 ms)
Below is the debug:
-- Executing Dial("OSS/dsp", "SIP/[EMAIL PROTECTED]|30") in
new stack
We're at outsideIPaddress port
14842
Answering with preferred capability 0x4
(ulaw)
12 headers, 8 lines
Reliably Transmitting:
INVITE sip:[EMAIL PROTECTED]
SIP/2.0
Via: SIP/2.0/UDP
outsideIPaddress:5060;branch=z9hG4bK1572d8bc
From: "asterisk"
sip:[EMAIL PROTECTED];tag=as6ed673e9
To:
sip:[EMAIL PROTECTED]
Contact:
sip:[EMAIL PROTECTED]
Call-ID:
[EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Wed, 09 Mar 2005 18:15:18
GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE,
REFER
Content-Type:
application/sdp
Content-Length: 164
v=0
o=root 17647 17647 IN IP4
outsideIPaddress
s=session
c=IN IP4 outsideIPaddress
t=0 0
m=audio 14842 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
(no NAT)
to 147.135.8.128:5060
-- Called [EMAIL PROTECTED]
asterisk1*CLI
Sip read:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
outsideIPaddress:5060;branch=z9hG4bK1572d8bc
From: "asterisk"
sip:[EMAIL PROTECTED];tag=as6ed673e9
To:
sip:[EMAIL PROTECTED]
Call-ID:
[EMAIL PROTECTED]
CSeq: 102 INVITE
6 headers, 0 lines
asterisk1*CLI
Sip read:
SIP/2.0 404
Not Found
Via: SIP/2.0/UDP
outsideIPaddress:5060;branch=z9hG4bK1572d8bc
From: "asterisk"
sip:[EMAIL PROTECTED];tag=as6ed673e9
To:
sip:[EMAIL PROTECTED];tag=SD4ou5a99-
Call-ID:
[EMAIL PROTECTED]
CSeq: 102
INVITE
Content-Length: 0
7 headers, 0 lines
-- Got SIP response 404 "Not Found" back from
147.135.8.128
Transmitting:
ACK sip:[EMAIL PROTECTED]
SIP/2.0
Via: SIP/2.0/UDP
outsideIPaddress:5060;branch=z9hG4bK1572d8bc
From: "asterisk"
sip:[EMAIL PROTECTED];tag=as6ed673e9
To:
sip:[EMAIL PROTECTED];tag=SD4ou5a99-
Contact:
sip:[EMAIL PROTECTED]
Call-ID:
[EMAIL PROTECTED]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0
(no NAT)
to 147.135.8.128:5060
-- SIP/sip.broadvoice.com-2a2c is
circuit-busy
==
Everyone is busy/congested at this time
-- Executing Busy("OSS/dsp", "") in new
stack
Destroying call
'[EMAIL PROTECTED]'
asterisk1*CLI hangup
== Spawn
extension (default, 509, 102) exited non-zero on
'OSS/dsp'
Hangup on console
[sip.broadvoice.com]
type=peer
host=proxy.lax.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser= BB
username= BB
;authuser= BB
secret= secret
context=sip
nat=no
insecure=very
dtmfmode=inband
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RE: [Asterisk-Users] Broadvoice latest changes and still not working-An
The problem that you have it was the one that I stabled
across the very first time tried to setup BV.
The 404 not found that you are getting is because there is
no such phone number [EMAIL PROTECTED]
But there is a [EMAIL PROTECTED].
This is like saying [EMAIL PROTECTED](you
are going to get a 404)
The chi worked because it was a test server (beta/debug)
that I read somewhere in this list.
So the fix for you will be to change the
host=proxy.lax.broadvoice.com
to
host=sip.broadvoice.com
Now if you are getting better responses from lax then
change your host file to
147.135.8.128 sip.broadvoice.com
This is because sip.broadvoice.com resolves to
proxy.dca.broadvoice.com.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
JoeSent: Wednesday, March 09, 2005 1:41 PMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] Broadvoice
latest changes and still not working-An
Ive tried everything with the * box after this
weekend. I have read every
document on the problems people are having with them after this weekend as
well, but none of them address my problem.
I checked my settings in my sips which I have below as
well,
I have changed the host file a few times, but this was new to me and I never had
modified it before. I have and
the same results happened.
I have always used the CHI proxy until this past
weekend.
I get a 404 not found when the invite goes out.
Below is my debug for broadvoice, which I think tells the whole
story, but for the life of me, I
can not figure out where the 404 is coming from.
I have listed my sip file below as
well.
Inbound calls work and I am
registered.
Before we go into the debug, I get this message when I reload my
configs files.
Mar 9 13:23:48 NOTICE[3161]: chan_sip.c:6779 handle_response:
Outbound Registration: Expiry for sip.broadvoice.com is 1948 sec (Scheduling
reregistration in 1933000 ms)
Below is the debug:
-- Executing Dial("OSS/dsp", "SIP/[EMAIL PROTECTED]|30") in
new stack
We're at outsideIPaddress port
14842
Answering with preferred capability 0x4
(ulaw)
12 headers, 8 lines
Reliably Transmitting:
INVITE sip:[EMAIL PROTECTED]
SIP/2.0
Via: SIP/2.0/UDP
outsideIPaddress:5060;branch=z9hG4bK1572d8bc
From: "asterisk"
sip:[EMAIL PROTECTED];tag=as6ed673e9
To:
sip:[EMAIL PROTECTED]
Contact:
sip:[EMAIL PROTECTED]
Call-ID:
[EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Wed, 09 Mar 2005 18:15:18
GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE,
REFER
Content-Type:
application/sdp
Content-Length: 164
v=0
o=root 17647 17647 IN IP4
outsideIPaddress
s=session
c=IN IP4 outsideIPaddress
t=0 0
m=audio 14842 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
(no NAT)
to 147.135.8.128:5060
-- Called [EMAIL PROTECTED]
asterisk1*CLI
Sip read:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
outsideIPaddress:5060;branch=z9hG4bK1572d8bc
From: "asterisk"
sip:[EMAIL PROTECTED];tag=as6ed673e9
To:
sip:[EMAIL PROTECTED]
Call-ID:
[EMAIL PROTECTED]
CSeq: 102 INVITE
6 headers, 0 lines
asterisk1*CLI
Sip read:
SIP/2.0 404
Not Found
Via: SIP/2.0/UDP
outsideIPaddress:5060;branch=z9hG4bK1572d8bc
From: "asterisk"
sip:[EMAIL PROTECTED];tag=as6ed673e9
To:
sip:[EMAIL PROTECTED];tag=SD4ou5a99-
Call-ID:
[EMAIL PROTECTED]
CSeq: 102
INVITE
Content-Length: 0
7 headers, 0 lines
-- Got SIP response 404 "Not Found" back from
147.135.8.128
Transmitting:
ACK sip:[EMAIL PROTECTED]
SIP/2.0
Via: SIP/2.0/UDP
outsideIPaddress:5060;branch=z9hG4bK1572d8bc
From: "asterisk"
sip:[EMAIL PROTECTED];tag=as6ed673e9
To:
sip:[EMAIL PROTECTED];tag=SD4ou5a99-
Contact:
sip:[EMAIL PROTECTED]
Call-ID:
[EMAIL PROTECTED]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0
(no NAT)
to 147.135.8.128:5060
-- SIP/sip.broadvoice.com-2a2c is
circuit-busy
==
Everyone is busy/congested at this time
-- Executing Busy("OSS/dsp", "") in new
stack
Destroying call
'[EMAIL PROTECTED]'
asterisk1*CLI hangup
== Spawn
extension (default, 509, 102) exited non-zero on
'OSS/dsp'
Hangup on console
[sip.broadvoice.com]
type=peer
host=proxy.lax.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser= BB
username= BB
;authuser= BB
secret= secret
context=sip
nat=no
insecure=very
dtmfmode=inband
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RE: [Asterisk-Users] Broadvoice latest changes and still not working
Don't you need 1 in front of the number?
Attempting call on SIP/Broadvoice/5068012
It should be Attempting call on SIP/Broadvoice/1(area code)5068012
Try it and see if you can place outgoing.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Geis
Sent: Tuesday, March 08, 2005 9:11 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Broadvoice latest changes and still not working
I have added the three lines to the sip.conf file based on the latest
changes
from broadvoice. I can receive incoming calls but cannot place any
outgoing calls.
The error I get is:
*CLI -- Registered to '69.73.19.178', who sees us as IPADDRESS:4569
-- Attempting call on SIP/Broadvoice/5068012 for application
Playback(demo-congrats) (Retry 1)
Mar 8 08:35:21 NOTICE[29290]: chan_sip.c:6814 handle_response: Failed
to authenticate on INVITE to 'asterisk
sip:[EMAIL PROTECTED];tag=as1304fa68'
Any ideas on why I cannot place calls?
THanks very much.
Jerry
--
; Broadvoice
register = [EMAIL PROTECTED]:SECRET:[EMAIL PROTECTED]/PHONE
[Broadvoice]
type=friend
username=PHONE
authuser=PHONE
fromuser=PHONE
secret=secret
host=sip.broadvoice.com
port=5060
context=default
fromdomain=sip.broadvoice.com
canreinvite=no
dtmfmode=inband
insecure=very
permit=sip.broadvoice.com
qualify=yes
disallow=all
allow=ulaw
maxexpirey=180
defaultexpirey=160
videosupport=no
exten =
_9XXX,1,Dial(SIP/Broadvoice/${EXTEN:1},${SMVOICE_DIAL_LONG_TIMEOUT},mtT)
exten =
_91XX,1,Dial(SIP/Broadvoice/${EXTEN:1},${SMVOICE_DIAL_LONG_TIMEOUT},mtT)
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RE: [Asterisk-Users] Broadvoice latest changes and still not working
Can you can post the relevant information from your sip.conf and extensions.conf ? Don't forget to hide password/phone/... From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Geis Sent: Tuesday, March 08, 2005 12:49 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Broadvoice latest changes and still not working The call is a local call so that should be fine. This worked in the past and I tried it the other way also with the same error message about Failed to authenticate on INVITE. Thanks Jerry Don't you need 1 in front of the number? Attempting call on SIP/Broadvoice/5068012 It should be Attempting call on SIP/Broadvoice/1(area code)5068012 Try it and see if you can place outgoing. -Original Message- From: asterisk-users-bounces at lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users [mailto:asterisk-users-bounces at lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users ] On Behalf Of Jerry Geis Sent: Tuesday, March 08, 2005 9:11 AM To: asterisk-users at lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users Subject: [Asterisk-Users] Broadvoice latest changes and still not working I have added the three lines to the sip.conf file based on the latest changes from broadvoice. I can receive incoming calls but cannot place any outgoing calls. The error I get is: *CLI -- Registered to '69.73.19.178', who sees us as IPADDRESS:4569 -- Attempting call on SIP/Broadvoice/5068012 for application Playback(demo-congrats) (Retry 1) Mar 8 08:35:21 NOTICE[29290]: chan_sip.c:6814 handle_response: Failed to authenticate on INVITE to 'asterisk sip:PHONE at sip.broadvoice.com http://lists.digium.com/mailman/listinfo/asterisk-users ;tag=as1304fa68' Any ideas on why I cannot place calls? THanks very much. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broadvoice latest changes and still not working
Does anybody have Broadvoice outbound working?
On Tue, 08 Mar 2005 13:18:12 -0500, Jerry Geis [EMAIL PROTECTED]
wrote:
Here is my configs. from a previous post...
Jerry
--
; Broadvoice
register = PHONE at sip.broadvoice.com
http://lists.digium.com/mailman/listinfo/asterisk-users:SECRET:PHONE
at sip.broadvoice.com
http://lists.digium.com/mailman/listinfo/asterisk-users/PHONE
[Broadvoice]
type=friend
username=PHONE
authuser=PHONE
fromuser=PHONE
secret=secret
host=sip.broadvoice.com
port=5060
context=default
fromdomain=sip.broadvoice.com
canreinvite=no
dtmfmode=inband
insecure=very
permit=sip.broadvoice.com
qualify=yes
disallow=all
allow=ulaw
maxexpirey=180
defaultexpirey=160
videosupport=no
exten =
_9XXX,1,Dial(SIP/Broadvoice/${EXTEN:1},${SMVOICE_DIAL_LONG_TIMEOUT},mtT)
exten =
_91XX,1,Dial(SIP/Broadvoice/${EXTEN:1},${SMVOICE_DIAL_LONG_TIMEOUT},mtT)
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--
James Taylor
MetroTel
3505 Summerihll Road
Suite 11
Texarkana, Texas 75503
903-793-1956
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RE: [Asterisk-Users] Broadvoice latest changes and still not working
Yes it is working just fine for me with the same sip.conf that you have. ??
Except the permit=sip.broadvoice.com
You can see my config at
http://lists.digium.com/pipermail/asterisk-users/2005-March/093047.html
Also what is your extensions.conf ?
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Taylor
Sent: Tuesday, March 08, 2005 2:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Broadvoice latest changes and still not working
Does anybody have Broadvoice outbound working?
On Tue, 08 Mar 2005 13:18:12 -0500, Jerry Geis [EMAIL PROTECTED]
wrote:
Here is my configs. from a previous post...
Jerry
--
; Broadvoice
register = PHONE at sip.broadvoice.com
http://lists.digium.com/mailman/listinfo/asterisk-users:SECRET:PHONE
at sip.broadvoice.com
http://lists.digium.com/mailman/listinfo/asterisk-users/PHONE
[Broadvoice]
type=friend
username=PHONE
authuser=PHONE
fromuser=PHONE
secret=secret
host=sip.broadvoice.com
port=5060
context=default
fromdomain=sip.broadvoice.com
canreinvite=no
dtmfmode=inband
insecure=very
permit=sip.broadvoice.com
qualify=yes
disallow=all
allow=ulaw
maxexpirey=180
defaultexpirey=160
videosupport=no
exten =
_9XXX,1,Dial(SIP/Broadvoice/${EXTEN:1},${SMVOICE_DIAL_LONG_TIMEOUT},mtT)
exten =
_91XX,1,Dial(SIP/Broadvoice/${EXTEN:1},${SMVOICE_DIAL_LONG_TIMEOUT},mtT)
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--
James Taylor
MetroTel
3505 Summerihll Road
Suite 11
Texarkana, Texas 75503
903-793-1956
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Re: [Asterisk-Users] Broadvoice latest changes and still not working
Ok, used your sip.conf inbound works. Outbound gets:
SIP/2.0 604 Does not exist anywhere
Any ideas?
James
On Tue, 8 Mar 2005 14:18:11 -0500, Marios Andreou [EMAIL PROTECTED]
wrote:
Yes it is working just fine for me with the same sip.conf that you have.
??
Except the permit=sip.broadvoice.com
You can see my config at
http://lists.digium.com/pipermail/asterisk-users/2005-March/093047.html
Also what is your extensions.conf ?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James
Taylor
Sent: Tuesday, March 08, 2005 2:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Broadvoice latest changes and still not
working
Does anybody have Broadvoice outbound working?
On Tue, 08 Mar 2005 13:18:12 -0500, Jerry Geis [EMAIL PROTECTED]
wrote:
Here is my configs. from a previous post...
Jerry
--
; Broadvoice
register = PHONE at sip.broadvoice.com
http://lists.digium.com/mailman/listinfo/asterisk-users:SECRET:PHONE
at sip.broadvoice.com
http://lists.digium.com/mailman/listinfo/asterisk-users/PHONE
[Broadvoice]
type=friend
username=PHONE
authuser=PHONE
fromuser=PHONE
secret=secret
host=sip.broadvoice.com
port=5060
context=default
fromdomain=sip.broadvoice.com
canreinvite=no
dtmfmode=inband
insecure=very
permit=sip.broadvoice.com
qualify=yes
disallow=all
allow=ulaw
maxexpirey=180
defaultexpirey=160
videosupport=no
exten =
_9XXX,1,Dial(SIP/Broadvoice/${EXTEN:1},${SMVOICE_DIAL_LONG_TIMEOUT},mtT)
exten =
_91XX,1,Dial(SIP/Broadvoice/${EXTEN:1},${SMVOICE_DIAL_LONG_TIMEOUT},mtT)
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James Taylor
MetroTel
3505 Summerihll Road
Suite 11
Texarkana, Texas 75503
903-793-1956
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Re: [Asterisk-Users] Broadvoice latest changes and still not working
My broadvoice works perfectly. I am using a standard registration string, however. Not the funky one broadvoice says to use. I can make outbound and receive inbound calls over broadvoice. I'm using AMP also. register=phonenumber:[EMAIL PROTECTED] sip.conf: [952XX] username=952XX type=friend secret=password regexten=952XXX insecure=very host=sip.broadvoice.com fromuser=952XX fromdomain=sip.broadvoice.com dtmfmode=inband context=from-pstn canreinvite=yes authuser=952 [sdfdsf] - Original Message - From: Marios Andreou [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Tuesday, March 08, 2005 1:18 PM Subject: RE: [Asterisk-Users] Broadvoice latest changes and still not working Yes it is working just fine for me with the same sip.conf that you have. ?? Except the permit=sip.broadvoice.com You can see my config at http://lists.digium.com/pipermail/asterisk-users/2005-March/093047.html Also what is your extensions.conf ? -Original Message- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Broadvoice latest changes and still not working
Hmm!!
OK I'm missing some variables (like what is ${OPERATOR}) but that is OK because
incoming you said is working.
But what I don't know is the context for the SIP/ext 's
It should be [smvoice-sip] so they can dial the _91X extension.
OR their context should include = smvoice-sip
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Geis
Sent: Tuesday, March 08, 2005 2:45 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Broadvoice latest changes and still not working
My extension.conf is below.
Jerry
-
[default]
exten = s,1,Wait,1 ; Wait before speaking
exten = s,2,Answer ; Answer the line
exten = s,3,DigitTimeout,2 ; Set Digit Timeout to
5 seconds
exten = s,4,ResponseTimeout,20 ; Set Response Timeout
to 10 seconds
exten = s,5,ChanIsAvail(SIP/201SIP/202SIP/203SIP/204SIP/205SIP/206)
exten = s,6,Cut(thechannel=AVAILCHAN,,1)
exten = s,7,Dial(${thechannel},${DIAL_TIMEOUT},tT)
exten = s,8,background(SM_ATTENDANT)
exten = s,9,noop(background done)
exten = s,10,SetVar(SMVOICE_EXTEN=${OPERATOR})
exten = s,11,Goto(default,operator,1)
exten = PHONE,1,Goto(default,s,1)
[smvoice-sip]
exten = 11,1,playback(demo-congrats)
exten = 11,2,hangup
exten =
_9XXX,1,Dial(SIP/Broadvoice/${EXTEN:1},${SMVOICE_DIAL_LONG_TIMEOUT},mtT)
exten =
_91XX,1,Dial(SIP/Broadvoice/${EXTEN:1},${SMVOICE_DIAL_LONG_TIMEOUT},mtT)
Yes it is working just fine for me with the same sip.conf that you have. ??
Except the permit=sip.broadvoice.com
You can see my config at
http://lists.digium.com/pipermail/asterisk-users/2005-March/093047.html
Also what is your extensions.conf ?
-Original Message-
From: asterisk-users-bounces at lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
[mailto:asterisk-users-bounces at lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users] On Behalf Of James
Taylor
Sent: Tuesday, March 08, 2005 2:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Broadvoice latest changes and still not working
Does anybody have Broadvoice outbound working?
On Tue, 08 Mar 2005 13:18:12 -0500, Jerry Geis geisj at pagestation.com
http://lists.digium.com/mailman/listinfo/asterisk-users
wrote:
/ Here is my configs. from a previous post...
//
// Jerry
//
// --
//
// ; Broadvoice
// register = PHONE at sip.broadvoice.com
// http://lists.digium.com/mailman/listinfo/asterisk-users:SECRET:PHONE
// at sip.broadvoice.com
// http://lists.digium.com/mailman/listinfo/asterisk-users/PHONE
//
// [Broadvoice]
// type=friend
// username=PHONE
// authuser=PHONE
// fromuser=PHONE
// secret=secret
// host=sip.broadvoice.com
// port=5060
// context=default
// fromdomain=sip.broadvoice.com
// canreinvite=no
// dtmfmode=inband
// insecure=very
// permit=sip.broadvoice.com
// qualify=yes
// disallow=all
// allow=ulaw
// maxexpirey=180
// defaultexpirey=160
// videosupport=no
//
//
// exten =
// _9XXX,1,Dial(SIP/Broadvoice/${EXTEN:1},${SMVOICE_DIAL_LONG_TIMEOUT},mtT)
//
// exten =
//
_91XX,1,Dial(SIP/Broadvoice/${EXTEN:1},${SMVOICE_DIAL_LONG_TIMEOUT},mtT)
//
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/
--
James Taylor
MetroTel
3505 Summerihll Road
Suite 11
Texarkana, Texas 75503
903-793-1956
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RE: [Asterisk-Users] Broadvoice latest changes and still not working
Can you call anywhere or is this problem just with broadvoice? Is there any type of firewall like a netscreen or iptables configured in your setup, which may be blocking outbound UDP? Do you have a packet capture of the traffic that is leaving your network for broadvoice? You should try to inspect the signaling exchange with their proxy. -Original Message- From: Jerry Geis [mailto:[EMAIL PROTECTED] Sent: Tuesday, March 08, 2005 2:53 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Broadvoice latest changes and still not working I tried removing the permit and that made no difference. I can still call in to the box but no calls out. Jerry -- Yes it is working just fine for me with the same sip.conf that you have. ?? Except the permit=sip.broadvoice.com You can see my config at http://lists.digium.com/pipermail/asterisk-users/2005-March/093047.html Also what is your extensions.conf ? -Original Message- From: asterisk-users-bounces at lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users [mailto:asterisk-users-bounces at lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users] On Behalf Of James Taylor Sent: Tuesday, March 08, 2005 2:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Broadvoice latest changes and still not working Does anybody have Broadvoice outbound working? On Tue, 08 Mar 2005 13:18:12 -0500, Jerry Geis geisj at pagestation.com http://lists.digium.com/mailman/listinfo/asterisk-users wrote: / Here is my configs. from a previous post... // // Jerry // // -- / ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Broadvoice latest changes and still not working
What is the output of the show version ?
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Geis
Sent: Tuesday, March 08, 2005 3:19 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Broadvoice latest changes and still not working
Marios,
You are correct. Every phone on the system (8 or so) has a context=smvoice-sip
in the config for every phone.
This config was all working uptil last saturday when broadvoice made the
changes.
It has not worked for outgoing calls since then. Incoming is still working.
This is one of my extensions.
[405]
type=friend
dtmfmode=rfc2833
username=405
secret=SECRET
disallow=all
allow=ulaw
allow=alaw
host=dynamic
context=smvoice-sip
insecure=very
callerid=Fred Smith 405
Jerry
--
Hmm!!
OK I'm missing some variables (like what is ${OPERATOR}) but that is OK because
incoming you said is working.
But what I don't know is the context for the SIP/ext 's
It should be [smvoice-sip] so they can dial the _91X extension.
OR their context should include = smvoice-sip
-Original Message-
From: asterisk-users-bounces at lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
[mailto:asterisk-users-bounces at lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users] On Behalf Of Jerry
Geis
Sent: Tuesday, March 08, 2005 2:45 PM
To: asterisk-users at lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
Subject: [Asterisk-Users] Broadvoice latest changes and still not working
My extension.conf is below.
Jerry
-
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Re: [Asterisk-Users] Broadvoice latest changes and still not working
On Tuesday March 08 2005 2:58 pm, James Taylor wrote:
Ok, used your sip.conf inbound works. Outbound gets:
SIP/2.0 604 Does not exist anywhere
Any ideas?
James
On Tue, 8 Mar 2005 14:18:11 -0500, Marios Andreou [EMAIL PROTECTED]
wrote:
Yes it is working just fine for me with the same sip.conf that you have.
??
Except the permit=sip.broadvoice.com
You can see my config at
http://lists.digium.com/pipermail/asterisk-users/2005-March/093047.html
Also what is your extensions.conf ?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James
Taylor
Sent: Tuesday, March 08, 2005 2:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Broadvoice latest changes and still not
working
Does anybody have Broadvoice outbound working?
On Tue, 08 Mar 2005 13:18:12 -0500, Jerry Geis [EMAIL PROTECTED]
snip
I have a same sip.conf and out and in are working well. I have
sip.broadvoice.com mapped to proxy.lax.broadvoice.com in my hosts file. this
is nice for me as i can use sip.broadvoice.com in all .conf and if i need to
change the proxy i do so in the hosts file. I do not use ${EXTEN:1} in my
outbound dial and i always dial 10 digits.
John Millican
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RE: [Asterisk-Users] Broadvoice latest changes and still not working
Since he can call into the box, he is registering with BV. Otherwise he would not be able to call in. So, his outgoing calls are messed up one way or another. How about doing a sip debug while placing a call via BV and post the results? Can you call anywhere or is this problem just with broadvoice? Is there any type of firewall like a netscreen or iptables configured in your setup, which may be blocking outbound UDP? Do you have a packet capture of the traffic that is leaving your network for broadvoice? You should try to inspect the signaling exchange with their proxy. -Original Message- From: Jerry Geis [mailto:[EMAIL PROTECTED] Sent: Tuesday, March 08, 2005 2:53 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Broadvoice latest changes and still not working I tried removing the permit and that made no difference. I can still call in to the box but no calls out. Jerry -- Yes it is working just fine for me with the same sip.conf that you have. ?? Except the permit=sip.broadvoice.com You can see my config at http://lists.digium.com/pipermail/asterisk-users/2005-March/093047.html Also what is your extensions.conf ? -Original Message- From: asterisk-users-bounces at lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users [mailto:asterisk-users-bounces at lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users] On Behalf Of James Taylor Sent: Tuesday, March 08, 2005 2:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Broadvoice latest changes and still not working Does anybody have Broadvoice outbound working? On Tue, 08 Mar 2005 13:18:12 -0500, Jerry Geis geisj at pagestation.com http://lists.digium.com/mailman/listinfo/asterisk-users wrote: / Here is my configs. from a previous post... // // Jerry // // -- / ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server
Chris,
first of all, if your server has been up for 200 days, I suggest you
update the kernel -- you don't say if it's Linux, but chances are that
yes... and there have been some security bugs patched recently.
That aside. I'm not sure, but it's possible that since you are using a
valid host name ('sip.broadvoice.com') in your dial statement, perhaps
* tried to talk to it directly and does not consider the section in
sip.conf. Just a guess. You will notice from the the sip debug output
that * does not even try to authenticate, as if it didn't know about
the user/secret.
I use the BV number as the section name, so the dial statement
essentially looks like: Dial([EMAIL PROTECTED])
Try changing yours to say broadvoice and then the corresponding
section in sip.conf. I'm using the DCA server, and didn't have an
issue at all when they introduced INVITE authentication on the
weekend. This is how my section looks like:
[360350]
type=peer
dtmfmode=inband
username=360350
fromuser=360350
secret=XX
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
canreinvite=no
nat=no
insecure=very
context=incoming
outgoinglimit=2
In /etc/hosts I have:
147.135.0.128 sip.broadvoice.com
It's the proxy.dca.broadvoice.com server. Hope this helps...
--Luki
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Re: [Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server
First, thanks for your help.
I have been changing these to different values but not getting it.
Could you further amplify your statement...
Try changing yours to say broadvoice and then the corresponding
section in sip.conf.
Thanks!
Chris
On Mar 9, 2005, at 12:08 AM, Luki wrote:
Chris,
first of all, if your server has been up for 200 days, I suggest you
update the kernel -- you don't say if it's Linux, but chances are that
yes... and there have been some security bugs patched recently.
That aside. I'm not sure, but it's possible that since you are using a
valid host name ('sip.broadvoice.com') in your dial statement, perhaps
* tried to talk to it directly and does not consider the section in
sip.conf. Just a guess. You will notice from the the sip debug output
that * does not even try to authenticate, as if it didn't know about
the user/secret.
I use the BV number as the section name, so the dial statement
essentially looks like: Dial([EMAIL PROTECTED])
Try changing yours to say broadvoice and then the corresponding
section in sip.conf. I'm using the DCA server, and didn't have an
issue at all when they introduced INVITE authentication on the
weekend. This is how my section looks like:
[360350]
type=peer
dtmfmode=inband
username=360350
fromuser=360350
secret=XX
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
canreinvite=no
nat=no
insecure=very
context=incoming
outgoinglimit=2
In /etc/hosts I have:
147.135.0.128 sip.broadvoice.com
It's the proxy.dca.broadvoice.com server. Hope this helps...
--Luki
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Re: [Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server
First off... please cancel previous amplification request. I have
implemented your ideas with the same errored result.
I am not sure that we are not making it thru authentication. From my
digging and comparing packet dumps comparing the soft phone to asterisk
they have identical transactions through the ACK reply (the last one
on the debug below). The softphone seems to be authenticated after the
ACK. I am a newbie to debugging this stuff. I just want to get it
working.
Thanks everyone in advance for your help. I am certainly very very
happy to try anything.
Based on Luki's suggestions I...
Changed sip.conf...
[broadvoice1]
type=peer
;user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=8475100139
secret=zjh018g8f8
username=8475100139
insecure=very
context=default
authname=8475100139
dtmfmode=inband
dtmf=inband
;Disable canreinvite if you are behind a NAT
canreinvite=no
nat=no
Changed extensions.conf...
exten = _8X.,1, dial(SIP/${EXTEN:[EMAIL PROTECTED],30) ; Dial Broadvoice
for 30 seconds
exten = _8X.,2, congestion() ; No answer, nothing
exten = _8X., 102, busy() ;
End result...
Mar 9 01:30:42 NOTICE[15376]: chan_sip.c:5047 handle_response: Failed
to authenticate on INVITE to '6050
sip:[EMAIL PROTECTED];tag=as545ccba3'
SIP debug...
-- Executing Dial(SIP/6050-132b,
SIP/[EMAIL PROTECTED]|30) in new stack
We're at xxx.xxx.xxx.xxx port 18212
Answering with capability 2
Answering with capability 4
Answering with capability 8
12 headers, 10 lines
Reliably Transmitting:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0
From: 6050 sip:[EMAIL PROTECTED];tag=as545ccba3
To: sip:[EMAIL PROTECTED]
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Wed, 09 Mar 2005 07:30:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 205
v=0
o=root 3998 3998 IN IP4 xxx.xxx.xxx.xxx
s=session
c=IN IP4 xxx.xxx.xxx.xxx
t=0 0
m=audio 18212 RTP/AVP 3 0 8
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
(no NAT) to 147.135.8.128:5060
-- Called [EMAIL PROTECTED]
com*CLI
Sip read:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 64.4.192.110:5060;branch=z9hG4bK-1b5b3221
From: 6050 sip:[EMAIL PROTECTED];tag=7e2776985d5a0891o0
To: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest
username=6050,realm=asterisk,nonce=42d82e9b,uri=sip:
[EMAIL PROTECTED],algorithm=MD5,response=420e39b35648a10c
129dd4fb5f97ec47
Contact: 6050 sip:[EMAIL PROTECTED]:5060
Expires: 240
User-Agent: Sipura/SPA3000-2.0.10(GWf)
Content-Length: 241
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp
v=0
o=- 1138990026 1138990026 IN IP4 64.4.192.110
s=-
c=IN IP4 64.4.192.110
t=0 0
m=audio 16388 RTP/AVP 0 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
15 headers, 12 lines
Ignoring this request
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 64.4.192.110:5060;branch=z9hG4bK-1b5b3221
From: 6050 sip:[EMAIL PROTECTED];tag=7e2776985d5a0891o0
To: sip:[EMAIL PROTECTED];tag=as2f065f18
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:[EMAIL PROTECTED]
Content-Length: 0
to 64.4.192.110:5060
com*CLI
Sip read:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0
From: 6050 sip:[EMAIL PROTECTED];tag=as545ccba3
To: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
6 headers, 0 lines
com*CLI
Sip read:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0
From: 6050 sip:[EMAIL PROTECTED];tag=as545ccba3
To: sip:[EMAIL PROTECTED];tag=SD38rq699-
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
WWW-Authenticate: DIGEST
realm=BroadWorks,algorithm=MD5,nonce=1110353299563
Content-Length: 0
8 headers, 0 lines
Transmitting:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0
From: 6050 sip:[EMAIL PROTECTED];tag=as545ccba3
To: sip:[EMAIL PROTECTED];tag=SD38rq699-
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0
(no NAT) to 147.135.8.128:5060
Mar 9 01:30:42 NOTICE[15376]: chan_sip.c:5047 handle_response: Failed
to authenticate on INVITE to '6050
sip:[EMAIL PROTECTED];tag=as545ccba3'
On Mar 9, 2005, at 12:08 AM, Luki wrote:
Chris,
first of all, if your server has been up for 200 days, I suggest you
update the kernel -- you don't say if it's Linux, but chances are that
yes... and there have been some security bugs patched recently.
That aside. I'm not sure, but it's possible that since you are using a
valid host name ('sip.broadvoice.com') in your dial statement, perhaps
