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Terry Evans

2004-Oct-23 14:32 UTC

[Asterisk-Users] Asterisk and Broadvoice, no incoming voice

I just signed up for the BroadVoice service a few hours ago, but for
the life of me I can't get any incoming voice.  The incoming
connection is fine as it rings my extension from outside, but I can't
hear anyone talking.   Outgoing voice is working fine though.

I've been looking through the archives, but I haven't found a solution
to the problem yet.  I even tried another router since someone had a
problem with that, but still no dice.

I've had my Asterisk server running fine for a few months, but this is
the first time I've tried a VOIP service with it.  I just downloaded
and installed the lastest CVS and the problem is still there also.

Here's some of my configuration information:

sip.conf (I've tried with nat=no and it didn't help)

[general]
context=from-sip       ; Default context for incoming calls
port=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)
maxexpirey=3600
defaultexpirey=120
callerid=No CallID
tos=lowdelay; 0x18 ; reliabile before
dtmfmode=inband
srvlookup=yes
;progressinband=no
nat=yes
notifymimetype=text/plain

[broadvoice]
type=friend
username=801527xxxx (hid real number)
fromuser=801527xxxx  (hid real number)
secret=xxxxxxxxxxxx (hid real password)
fromdomain=sip.broadvoice.com
host=sip.broadvoice.com
canreinvite=no
dtmfmode=inband
context=broadvoice-inbound
nat=yes (tried nat=never also)
disallow=all
allow=ulaw
insecure=very

I have the following ports forwarded to my linux server (it's behind a
NAT router):

5060, 20000-21000 (from my rdp.conf file), 4445, and 4569.  All of
those have both TCP and UDP forwarded for now.

I've tried several different combinations from different posts,
including splitting the broadvoice section up into parts for incoming
and outgoing, but it still didn't work.

Anyone have any ideas?  Let me know if traces, etc. will help and I'll
capture and post some.

Thanks,
Terry

Greg Hill

2004-Oct-23 16:39 UTC

head link

[Asterisk-Users] Asterisk and Broadvoice, no incoming voice

On Sat, 23 Oct 2004, Terry Evans wrote:
> I just signed up for the BroadVoice service a few hours ago, but for
> the life of me I can't get any incoming voice.  The incoming
> connection is fine as it rings my extension from outside, but I can't
> hear anyone talking.   Outgoing voice is working fine though.
(snip)> I have the following ports forwarded to my linux server (it's behind a
> NAT router):
>
> 5060, 20000-21000 (from my rdp.conf file), 4445, and 4569.  All of
> those have both TCP and UDP forwarded for now.
It really sounds like a NAT problem to me.. If your NAT supports the
notion of a "DMZ host" then give that a try. Or if the NAT has some
sort
of logging feature to let you know when the nat receives unexpected
packets and discards them, then look through the log. It may be that BV
isn't sending RTP in the 20000-21000 port range, and that these packets
are being dropped by the NAT. Outgoing RTP (voice) would work fine, of
course, because the NAT is designed to work that direction.

FYI, I just placed a call to my BV number and ran 'netstat -nupa'. UDP
connections showed up on ports 14704, 14705, 19838, 19839. These
disappeared when I hung up the call.

While it might be a config issue, I'm inclined to believe that NAT is
making life unpleasant for you.

Greg

Tim Jackson

2004-Oct-23 22:29 UTC

head link

[Asterisk-Users] Asterisk and Broadvoice, no incoming voice

I'm having the same issue, and I'm not behind NAT.

Maybe this is a BV issue?

-Tim

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Terry
Evans
Sent: Saturday, October 23, 2004 4:33 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Asterisk and Broadvoice, no incoming voice

I just signed up for the BroadVoice service a few hours ago, but for
the life of me I can't get any incoming voice.  The incoming
connection is fine as it rings my extension from outside, but I can't
hear anyone talking.   Outgoing voice is working fine though.

I've been looking through the archives, but I haven't found a solution
to the problem yet.  I even tried another router since someone had a
problem with that, but still no dice.

I've had my Asterisk server running fine for a few months, but this is
the first time I've tried a VOIP service with it.  I just downloaded
and installed the lastest CVS and the problem is still there also.

Here's some of my configuration information:

sip.conf (I've tried with nat=no and it didn't help)

[general]
context=from-sip       ; Default context for incoming calls
port=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)
maxexpirey=3600
defaultexpirey=120
callerid=No CallID
tos=lowdelay; 0x18 ; reliabile before
dtmfmode=inband
srvlookup=yes
;progressinband=no
nat=yes
notifymimetype=text/plain

[broadvoice]
type=friend
username=801527xxxx (hid real number)
fromuser=801527xxxx  (hid real number)
secret=xxxxxxxxxxxx (hid real password)
fromdomain=sip.broadvoice.com
host=sip.broadvoice.com
canreinvite=no
dtmfmode=inband
context=broadvoice-inbound
nat=yes (tried nat=never also)
disallow=all
allow=ulaw
insecure=very

I have the following ports forwarded to my linux server (it's behind a
NAT router):

5060, 20000-21000 (from my rdp.conf file), 4445, and 4569.  All of
those have both TCP and UDP forwarded for now.

I've tried several different combinations from different posts,
including splitting the broadvoice section up into parts for incoming
and outgoing, but it still didn't work.

Anyone have any ideas?  Let me know if traces, etc. will help and I'll
capture and post some.

Thanks,
Terry
_______________________________________________
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JeffPOwen@comcast.net

2004-Oct-26 06:30 UTC

head link

[Asterisk-Users] Asterisk and Broadvoice, no incoming voice

I have it working fine....here are my configs:
On my NAT Firewall I have port 5060 UDP and 10000-20000 UDP open to my * box.
In SIP.conf I have the following:
[general]
context=incoming
port=5060
bindaddr=192.168.234.111
maxexpirey=180
defaultexpirey=160
tos=0x08
nat=no
srvlookup=yes
videosupport=no
dtmfmode=inband
disallow=all ; Disallow all codecs
allow=ulaw
language=en
externip=<no-ip.com_hostname>
localnet=192.168.234.0/255.255.255.0
register=<phone_number>:<password>@sip.broadvoice.com/broadvoice
[broadvoice]
type=friend
username=<phonenumber>
fromuser=<phonenumber>
secret=<password>
host=sip.broadvoice.com
maxexpirey=15
fromdomain=sip.broadvoice.com
nat=yes
canreinvite=no
insecure=very
qualify=yes
dtmfmode=inband
disallow=all ; Disallow all codecs
allow=ulaw
In the EXTENSIONS.conf I have an extension defined to forward the
"broadvoice" extension to my Sipura SPA-2000 Line 1 extension.
It works fine for me thru NAT.
Hope this helps.
-Jeff



-------------------------------------------------------------------------------------------------------------------------
I'm having the same issue too. Just signed up, never had a problem before
with a sipura box at a friends house configured as an extension off my Asterisk
box. I'm behind NAT, and my firewall has the asterisk box configured as a
DMZ, also I forwarded the sip and RTP ports just to be sure.
Anyone solve this one yet? I sent an email off to broadvoice but they were less
than helpful.

Tim Jackson <tim@angelinacounty.net> [2004-10-24 00:29:02
-0500]:> I'm having the same issue, and I'm not behind NAT.
> 
> Maybe this is a BV issue?
> 
> -Tim
> 
> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com
> [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Terry 
> Evans
> Sent: Saturday, October 23, 2004 4:33 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Asterisk and Broadvoice, no incoming voice
> 
> I just signed up for the BroadVoice service a few hours ago, but for 
> the life of me I can't get any incoming voice. The incoming 
> connection is fine as it rings my extension from outside, but I can't
> hear anyone talking. Outgoing voice is working fine though.
> 
> I've been looking through the archives, but I haven't found a
solution
> to the problem yet. I even tried another router since someone had a 
> problem with that, but still no dice.
> 
> I've had my Asterisk server running fine for a few months, but this is 
> the first time I've tried a VOIP service with it. I just downloaded 
> and installed the lastest CVS and the problem is still there also.
> 
> Here's some of my configuration information:
> 
> sip.conf (I've tried with nat=no and it didn't help)
> 
> [general]
> context=from-sip ; Default context for incoming calls
> port=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)
> maxexpirey=3600
> defaultexpirey=120
> callerid=No CallID
> tos=lowdelay; 0x18 ; reliabile before
> dtmfmode=inband
> srvlookup=yes
> ;progressinband=no
> nat=yes
> notifymimetype=text/plain
> 
> [broadvoice]
> type=friend
> username=801527xxxx (hid real number)
> fromuser=801527xxxx (hid real number) secret=xxxxxxxxxxxx (hid real 
> password) fromdomain=sip.broadvoice.com host=sip.broadvoice.com 
> canreinvite=no dtmfmode=inband context=broadvoice-inbound nat=yes 
> (tried nat=never also) disallow=all allow=ulaw insecure=very
> 
> I have the following ports forwarded to my linux server (it's behind a 
> NAT router):
> 
> 5060, 20000-21000 (from my rdp.conf file), 4445, and 4569. All of 
> those have both TCP and UDP forwarded for now.
> 
> I've tried several different combinations from different posts, 
> including splitting the broadvoice section up into parts for incoming 
> and outgoing, but it still didn't work.
> 
> Anyone have any ideas? Let me know if traces, etc. will help and I'll 
> capture and post some.
> 
> Thanks,
> Terry
> _______________________________________________
> 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_______________________________________________
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Jason Schafer

2005-Sep-26 08:12 UTC

head link

[Asterisk-Users] Asterisk and Broadvoice, no incoming voice

Hi:

I am running AAH and setup Broadvoice, but when I call in to the BV 
number I cannot send dial commands to my auto attendant or speak if I 
use a did to send the inbound calls to a specific extension.  I'll 
gladly capture an SID debug and place a call, or post any necessary conf 
files.

TIA

Jason

Darren Wright

2005-Sep-26 10:34 UTC

head link

[Asterisk-Users] Asterisk and Broadvoice, no incoming voice

I am also a long time client, and have no incoming BV today.
 
-Darren
 

________________________________

From: asterisk-users-bounces@lists.digium.com on behalf of Jason Schafer
Sent: Mon 9/26/2005 12:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk and Broadvoice, no incoming voice


> Does asterisk says something in the verbose console?
I'm not sure what the verbose console is, but I can run sip debug and
post the output when I make an inbound call.
> please post your sip.conf relevant entries for BroadVoice.
[general]
port = 5060           ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0    ; Address to bind to (all addresses on machine)
disallow=all
allow=ulaw
allow=alaw
context = from-sip-external ; Send unknown SIP callers to this context
callerid = Unknown
pedantic=no ; added for Broadvoice support 8/3/05 EK
externip=216.xxx.xxx.xxx
localnet=172.xxx.xxx.0/255.255.255.0


I have just> cancelled with BroadVoice (too much latency for the places i wanted to
> call), so i never used the incoming number. But im glad to help if i can.
I have outbound setup on VOIPJet, my intent with the Broadvoice is to
setup a forward on busy with my landline to roll over to the BV number.

Here's the output from sip debug

m=audio 14008 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

13 headers, 12 lines
Using latest request as basis request
Sending to 147.135.0.128 : 5060 (non-NAT)
Found no matching peer or user for '147.135.0.128:5060'
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 147.135.0.128:14008
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 - 0xc (ulaw|alaw), peer - audio=0x51c
(ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined -
0x1 (g723)
Looking for s in from-sip-external
list_route: hop:
<sip:610253xxxx@147.135.0.128:5060;ep=147.135.0.129;transport=udp>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 147.135.0.128:5060;branch=z9hG4bK3qd0e4102041k9sak0k1.1sr
From: "Schafer Trish
"<sip:610253xxxx@147.135.0.129;user=phone>;tag=SD28clb01-1612693231-1127750324179
To: "Jason
Schafer"<sip:484549xxxx@sip.broadvoice.com;user=phone>
Call-ID: SD28clb01-d697b8d4cd8742e341c1f1942d1bf7e1-js11002
CSeq: 160704490 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:s@216.xxx.xxx.xxx>
Content-Length: 0


  to 147.135.0.128:5060
     -- Executing Wait("SIP/147.135.0.129-095da350", "1") in
new stack
     -- Executing Goto("SIP/147.135.0.129-095da350",
"from-pstn|s|1") in
new stack
     -- Goto (from-pstn,s,1)
     -- Executing GotoIf("SIP/147.135.0.129-095da350",
"1?from-pstn-reghours|s|1:") in new stack
     -- Goto (from-pstn-reghours,s,1)
     -- Executing GotoIf("SIP/147.135.0.129-095da350",
"0?from-pstn-reghours-nofax|s|1:2") in new stack
     -- Goto (from-pstn-reghours,s,2)
     -- Executing Answer("SIP/147.135.0.129-095da350", "")
in new stack
We're at 216.xxx.xxx.xxx port xxxxx
Answering with preferred capability 0x4 (ulaw)
Answering with preferred capability 0x8 (alaw)
Answering with non-codec capability 0x1 (telephone-event)
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 147.135.0.128:5060;branch=z9hG4bK3qd0e4102041k9sak0k1.1sr
From: "Schafer Trish
"<sip:610253xxxx@147.135.0.129;user=phone>;tag=SD28clb01-1612693231-1127750324179
To: "Jason
Schafer"<sip:484549xxxx@sip.broadvoice.com;user=phone>;tag=as2a994d31
Call-ID: SD28clb01-d697b8d4cd8742e341c1f1942d1bf7e1-js11002
CSeq: 160704490 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:s@216.xxx.xxx.xxx>
Content-Type: application/sdp
Content-Length: 238

v=0
o=root 1782 1782 IN IP4 216.xxx.xxx.xxx
s=session
c=IN IP4 216.xxx.xxx.xxx
t=0 0
m=audio 14138 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

  to 147.135.0.128:5060
     -- Executing Wait("SIP/147.135.0.129-095da350", "1") in
new stack
asterisk1*CLI>

Sip read:
ACK sip:s@172.xxx.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP 147.135.0.128:5060;branch=z9hG4bK3qd0u4103gtgb94c0080.1sr
From: "Schafer Trish
"<sip:610253xxxx@147.135.0.129;user=phone>;tag=SD28clb01-1612693231-1127750324179
To: "Jason
Schafer"<sip:484549xxxx@sip.broadvoice.com;user=phone>;tag=as2a994d31
Call-ID: SD28clb01-d697b8d4cd8742e341c1f1942d1bf7e1-js11002
CSeq: 160704490 ACK
Contact: <sip:610253xxxx@147.135.0.128:5060;transport=udp>
Max-Forwards: 69
Content-Length: 0


9 headers, 0 lines
     -- Executing SetVar("SIP/147.135.0.129-095da350",
"intype=aa_2") in
new stack
     -- Executing Cut("SIP/147.135.0.129-095da350",
"intype=intype|-|1")
in new stack
     -- Executing GotoIf("SIP/147.135.0.129-095da350",
"0?7:9") in new stack
     -- Goto (from-pstn-reghours,s,9)
     -- Executing GotoIf("SIP/147.135.0.129-095da350",
"0?10:12") in new
stack
     -- Goto (from-pstn-reghours,s,12)
     -- Executing GotoIf("SIP/147.135.0.129-095da350",
"0?13:15") in new
stack
     -- Goto (from-pstn-reghours,s,15)
     -- Executing Goto("SIP/147.135.0.129-095da350",
"aa_2|s|1") in new
stack
     -- Goto (aa_2,s,1)
     -- Executing GotoIf("SIP/147.135.0.129-095da350",
"0?4") in new stack
     -- Executing Answer("SIP/147.135.0.129-095da350", "")
in new stack
     -- Executing Wait("SIP/147.135.0.129-095da350", "1") in
new stack
asterisk1*CLI>

Sip read:


0 headers, 0 lines
     -- Executing SetVar("SIP/147.135.0.129-095da350",
"DIR-CONTEXT=general") in new stack
     -- Executing DigitTimeout("SIP/147.135.0.129-095da350",
"3") in new
stack
     -- Set Digit Timeout to 3
     -- Executing ResponseTimeout("SIP/147.135.0.129-095da350",
"7") in
new stack
     -- Set Response Timeout to 7
     -- Executing BackGround("SIP/147.135.0.129-095da350",
"custom/aa_2") in new stack
     -- Playing 'custom/aa_2' (language 'en')
asterisk1*CLI>

Sip read:


0 headers, 0 lines
12 headers, 0 lines
Reliably Transmitting:
REGISTER sip:sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP 216.xxx.xxx.xxx:5060;branch=z9hG4bK0cefd7e0
From: <sip:484549xxxx@sip.broadvoice.com>;tag=as630d5ca8
To: <sip:484549xxxx@sip.broadvoice.com>
Call-ID: 2d3c52c27ad7b993384cb89f5bca4c9b@127.0.0.1
CSeq: 135 REGISTER
User-Agent: Asterisk PBX
Authorization: Digest username="484549xxxx",
realm="BroadWorks",
algorithm=MD5, uri="sip:sip.broadvoice.com",
nonce="1127748659968",
response="a41b375d644bc72e8ef3a7049435c1e7", opaque=""
Expires: 120
Contact: <sip:s@216.xxx.xxx.xxx>
Event: registration
Content-Length: 0

  (no NAT) to 147.135.0.128:5060
asterisk1*CLI>

Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.xxx.xxx.xxx:5060;branch=z9hG4bK0cefd7e0
From: <sip:484549xxxx@sip.broadvoice.com>;tag=as630d5ca8
To: <sip:484549xxxx@sip.broadvoice.com>
Call-ID: 2d3c52c27ad7b993384cb89f5bca4c9b@127.0.0.1
CSeq: 135 REGISTER
Contact: <sip:s@172.xxx.xxx.xxx>;expires=1918


7 headers, 0 lines
Destroying call '2d3c52c27ad7b993384cb89f5bca4c9b@127.0.0.1'
asterisk1*CLI>

Sip read:
BYE sip:s@172.xxx.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP 147.135.0.128:5060;branch=z9hG4bK3qd0ee10do30sas9b3s1.1sr
From: "Schafer Trish
"<sip:610253xxxx@147.135.0.129;user=phone>;tag=SD28clb01-1612693231-1127750324179
To: "Jason
Schafer"<sip:484549xxxx@sip.broadvoice.com;user=phone>;tag=as2a994d31
Call-ID: SD28clb01-d697b8d4cd8742e341c1f1942d1bf7e1-js11002
CSeq: 160704491 BYE
Max-Forwards: 69
Content-Length: 0


8 headers, 0 lines
Sending to 147.135.0.128 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 147.135.0.128:5060;branch=z9hG4bK3qd0ee10do30sas9b3s1.1sr
From: "Schafer Trish
"<sip:610253xxxx@147.135.0.129;user=phone>;tag=SD28clb01-1612693231-1127750324179
To: "Jason
Schafer"<sip:484549xxxx@sip.broadvoice.com;user=phone>;tag=as2a994d31
Call-ID: SD28clb01-d697b8d4cd8742e341c1f1942d1bf7e1-js11002
CSeq: 160704491 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:s@216.xxx.xxx.xxx>
Content-Length: 0


  to 147.135.0.128:5060
   == Spawn extension (aa_2, s, 7) exited non-zero on
'SIP/147.135.0.129-095da350'
     -- Executing Hangup("SIP/147.135.0.129-095da350", "")
in new stack
   == Spawn extension (aa_2, h, 1) exited non-zero on
'SIP/147.135.0.129-095da350'
Destroying call 'SD28clb01-d697b8d4cd8742e341c1f1942d1bf7e1-js11002'
asterisk1*CLI>

Sip read:


0 headers, 0 lines
asterisk1*CLI>

Sip read:


0 headers, 0 lines
asterisk1*CLI>

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Jason Schafer

2005-Sep-26 10:45 UTC

head link

[Asterisk-Users] Asterisk and Broadvoice, no incoming voice

I have been trying on and off for a couple of weeks to no avail...

Darren Wright wrote:
> I am also a long time client, and have no incoming BV today.
>  
> -Darren
>    http://lists.digium.com/mailman/listinfo/asterisk-users

Janina Sajka

2005-Dec-01 09:20 UTC

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[Asterisk-Users] Asterisk and Broadvoice, no incoming voice

My incoming BV has been intermittant for the last two days as well. It
has gone down somewhere around 4:30 PM Eastern two days in a row, then
been back up in the morning. In the 10:00 AM hour today, it was down for
about ten minutes.

Jason Schafer writes:> I have been trying on and off for a couple of weeks to no avail...
> 
> Darren Wright wrote:
> 
> >I am also a long time client, and have no incoming BV today.
> > 
> >-Darren
> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> _______________________________________________
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Janina Sajka				Phone: +1.240.715.1272
Partner, Capital Accessibility LLC	http://www.CapitalAccessibility.Com

Marketing the Owasys 22C talking screenless cell phone in the U.S. and
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