I have an Asterisk server, running 1.6.0, on my home network. It has an IP
address of 192.168.0.201. The server is behind my Linksys router that does
NAT for my home devices. I have a softphone on my Mac that is on the same
NATed network as the Asterisk server. It has an IP address of 192.168.0.1. I
have tried this with a Grandstream phone with the same results.
The Asterisk server registers with a SIP provider and seems to maintain a
registration.
When I try to dial out from the soft phone or the Grandstream via the SIP
provider I get a "Bad Request" response from the provider. I am
guessing
that many of the packets have my internal address and need my public
address.
Is there some way to resolve this?
Here is the relevant entry in extensions.conf:
exten => _88X.,1,Dial(SIP/${EXTEN:2}@proxy01.sipphone.com,20,r)
exten => _88X.,n,Hangup()
Here are my entries in sip.conf:
[proxy01.sipphone.com]
type=peer
context=mine
disallow=all
allow=ulaw
dtmfmode=rfc2833
host=proxy01.sipphone.com
fromdomain=proxy01.sipphone.com
insecure=port,invite
qualify=yes
fromuser=myid
authuser=myid
defaultuser=myid
secret=mypwd
canreinvite=no
[GXP280]
type=friend
context=mine
nat=no
canreinvite=no
host=dynamic
secret=mypwd
callerid=GXP280 <109>
mailbox=109 at ourvm
busylevel=2
[dickenson]
type=friend
context=empl
nat=no
canreinvite=no
host=dynamic
secret=mypwd
callerid=Jim Dickenson <108>
mailbox=108 at ourvm
Looking at the SIP debug transaction I do not see my public IP address at
all. There must be some setting about the server having at NATed address.
--
Jim Dickenson
mailto:dickenson at cfmc.com
CfMC
cfmc.com
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Investigate the externip= option for your SIP peers to the provider. Jim Dickenson wrote:> I have an Asterisk server, running 1.6.0, on my home network. It has an > IP address of 192.168.0.201. The server is behind my Linksys router that > does NAT for my home devices. I have a softphone on my Mac that is on > the same NATed network as the Asterisk server. It has an IP address of > 192.168.0.1. I have tried this with a Grandstream phone with the same > results. > > The Asterisk server registers with a SIP provider and seems to maintain > a registration. > > When I try to dial out from the soft phone or the Grandstream via the > SIP provider I get a "Bad Request" response from the provider. I am > guessing that many of the packets have my internal address and need my > public address. > > Is there some way to resolve this? > > Here is the relevant entry in extensions.conf: > > exten => _88X.,1,Dial(SIP/${EXTEN:2}@proxy01.sipphone.com,20,r) > exten => _88X.,n,Hangup() > > > Here are my entries in sip.conf: > > [proxy01.sipphone.com] > type=peer > context=mine > disallow=all > allow=ulaw > dtmfmode=rfc2833 > host=proxy01.sipphone.com > fromdomain=proxy01.sipphone.com > insecure=port,invite > qualify=yes > fromuser=myid > authuser=myid > defaultuser=myid > secret=mypwd > canreinvite=no > > [GXP280] > type=friend > context=mine > nat=no > canreinvite=no > host=dynamic > secret=mypwd > callerid=GXP280 <109> > mailbox=109 at ourvm > busylevel=2 > > [dickenson] > type=friend > context=empl > nat=no > canreinvite=no > host=dynamic > secret=mypwd > callerid=Jim Dickenson <108> > mailbox=108 at ourvm > > Looking at the SIP debug transaction I do not see my public IP address > at all. There must be some setting about the server having at NATed address. > > -- > Jim Dickenson > mailto:dickenson at cfmc.com > > CfMC > cfmc.com > > > ------------------------------------------------------------------------ > > _______________________________________________ > -- Bandwidth and Colocation Provided by api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > lists.digium.com/mailman/listinfo/asterisk-users-- Alex Balashov Evariste Systems Web : evaristesys.com Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599
Add nat=yes for your provider [proxy01.sipphone.com] should do the trick. Qualify=60 is good for keeping firewall ports open. You could supplement that with externip= in the [general] section but it is probably not needed. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) On Wed, Nov 12, 2008 at 4:13 PM, Alex Balashov <abalashov at evaristesys.com>wrote:> Investigate the externip= option for your SIP peers to the provider. > > Jim Dickenson wrote: > > > I have an Asterisk server, running 1.6.0, on my home network. It has an > > IP address of 192.168.0.201. The server is behind my Linksys router that > > does NAT for my home devices. I have a softphone on my Mac that is on > > the same NATed network as the Asterisk server. It has an IP address of > > 192.168.0.1. I have tried this with a Grandstream phone with the same > > results. > > > > The Asterisk server registers with a SIP provider and seems to maintain > > a registration. > > > > When I try to dial out from the soft phone or the Grandstream via the > > SIP provider I get a "Bad Request" response from the provider. I am > > guessing that many of the packets have my internal address and need my > > public address. > > > > Is there some way to resolve this? > > > > Here is the relevant entry in extensions.conf: > > > > exten => _88X.,1,Dial(SIP/${EXTEN:2}@proxy01.sipphone.com<EXTEN%3A2%7D at proxy01.sipphone.com> > ,20,r) > > exten => _88X.,n,Hangup() > > > > > > Here are my entries in sip.conf: > > > > [proxy01.sipphone.com] > > type=peer > > context=mine > > disallow=all > > allow=ulaw > > dtmfmode=rfc2833 > > host=proxy01.sipphone.com > > fromdomain=proxy01.sipphone.com > > insecure=port,invite > > qualify=yes > > fromuser=myid > > authuser=myid > > defaultuser=myid > > secret=mypwd > > canreinvite=no > > > > [GXP280] > > type=friend > > context=mine > > nat=no > > canreinvite=no > > host=dynamic > > secret=mypwd > > callerid=GXP280 <109> > > mailbox=109 at ourvm > > busylevel=2 > > > > [dickenson] > > type=friend > > context=empl > > nat=no > > canreinvite=no > > host=dynamic > > secret=mypwd > > callerid=Jim Dickenson <108> > > mailbox=108 at ourvm > > > > Looking at the SIP debug transaction I do not see my public IP address > > at all. There must be some setting about the server having at NATed > address. > > > > -- > > Jim Dickenson > > mailto:dickenson at cfmc.com > > > > CfMC > > cfmc.com > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > -- Bandwidth and Colocation Provided by api-digital.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > lists.digium.com/mailman/listinfo/asterisk-users > > > -- > Alex Balashov > Evariste Systems > Web : evaristesys.com > Tel : (+1) (678) 954-0670 > Direct : (+1) (678) 954-0671 > Mobile : (+1) (706) 338-8599 >-------------- next part -------------- An HTML attachment was scrubbed... URL: lists.digium.com/pipermail/asterisk-users/attachments/20081112/d152b0f1/attachment.htm
AST --> FW NAT --> CARRIER sip.conf externip=<PUBLIC IP GW OF ASTERISK> nat=route localnet=<IP AND LOCAL MASK>(ex .192.168.0.0/255.255.0.0) ;Carrier example [Carrier] type=friend host=CARRIER IP fromdomain= CARRIER IP context=incoming disallow=all allow=g729 canreinvite=no insecure=very Good Luck -----Mensaje original----- De: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] En nombre de Alex Balashov Enviado el: Wednesday, November 12, 2008 7:14 PM Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [asterisk-users] SIP provider and NAT Investigate the externip= option for your SIP peers to the provider. Jim Dickenson wrote:> I have an Asterisk server, running 1.6.0, on my home network. It has an > IP address of 192.168.0.201. The server is behind my Linksys router that > does NAT for my home devices. I have a softphone on my Mac that is on > the same NATed network as the Asterisk server. It has an IP address of > 192.168.0.1. I have tried this with a Grandstream phone with the same > results. > > The Asterisk server registers with a SIP provider and seems to maintain > a registration. > > When I try to dial out from the soft phone or the Grandstream via the > SIP provider I get a "Bad Request" response from the provider. I am > guessing that many of the packets have my internal address and need my > public address. > > Is there some way to resolve this? > > Here is the relevant entry in extensions.conf: > > exten => _88X.,1,Dial(SIP/${EXTEN:2}@proxy01.sipphone.com,20,r) > exten => _88X.,n,Hangup() > > > Here are my entries in sip.conf: > > [proxy01.sipphone.com] > type=peer > context=mine > disallow=all > allow=ulaw > dtmfmode=rfc2833 > host=proxy01.sipphone.com > fromdomain=proxy01.sipphone.com > insecure=port,invite > qualify=yes > fromuser=myid > authuser=myid > defaultuser=myid > secret=mypwd > canreinvite=no > > [GXP280] > type=friend > context=mine > nat=no > canreinvite=no > host=dynamic > secret=mypwd > callerid=GXP280 <109> > mailbox=109 at ourvm > busylevel=2 > > [dickenson] > type=friend > context=empl > nat=no > canreinvite=no > host=dynamic > secret=mypwd > callerid=Jim Dickenson <108> > mailbox=108 at ourvm > > Looking at the SIP debug transaction I do not see my public IP address > at all. There must be some setting about the server having at NATedaddress.> > -- > Jim Dickenson > mailto:dickenson at cfmc.com > > CfMC > cfmc.com > > > ------------------------------------------------------------------------ > > _______________________________________________ > -- Bandwidth and Colocation Provided by api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > lists.digium.com/mailman/listinfo/asterisk-users-- Alex Balashov Evariste Systems Web : evaristesys.com Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 _______________________________________________ -- Bandwidth and Colocation Provided by api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: lists.digium.com/mailman/listinfo/asterisk-users