Re: [asterisk-users] Unable to make outgoing calls with Internode
On Thu, 10 Feb 2011 13:08:29 +1000, Da Rock asterisk-us...@herveybayaustralia.com.au wrote: I have an asterisk 1.8 server running on FreeBSD 8.1, and another FreeBSD 8.1 running as a firewall/gateway with PF. Does it work if you remove the firewall from the equation? Since Internode is an OZ company, and provided this issue turns out to be specific to that provider, you might have more luck solving the problem by asking in the Whirlpool forum: http://forums.whirlpool.net.au/forum/68 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Unable to make outgoing calls with Internode
Surely there must be someone here who can help me with this problem.
I have spent weeks trying to get this damned service to work with no
luck. I have incoming calls working, but no outgoing. If get outgoing
working then incoming don't work.
I have sent this problem to this list a couple of times with little or
no response, and I _really_ need some help to sort it out.
I have an asterisk 1.8 server running on FreeBSD 8.1, and another
FreeBSD 8.1 running as a firewall/gateway with PF.
I have a nodephone service with Internode (who have been absolutely
useless in helping me- they point blank refuse, or they say to open
everything right up to their server; which didn't wok anyway btw).
I have been running endless tests on settings changes, tcpdumps on both
the firewall and asterisk, and hours poring over SIP rfc's. I've only
managed to get a headache...
I have tried following best practices, worst practices, and still
nothing works.
My sip.conf looks like this:
[general]
context=default
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
allow=all
;allow=t140red
textsupport=yes
videosupport=yes
;allow=h263
maxcallbitrate=384
register=sip-in?phone
number:secret@sip.internode.on.net/phone number
externip= my static ip
localnet= my local subnet
canreinvite=no
hasvoicemail=no
qualify=yes
nat=no
;rtptimeout=120
rtpkeepalive=5
;ignoresdpversion=yes
;directmediapermit= my local subnet
[sip-in]
type=peer
host=sip.internode.on.net
context=internode-incoming
;externip= my static ip
;domain=internode.on.net,internode-incoming
;fromdomain=sip.internode.on.net
;fromuser= phone number
;username= phone number
;secret= secret
;auth= phone number:secret@BroadWorks
;insecure=invite,port
;register= phone number:secret@sip.internode.on.net
;nat=never
qualify=yes
canreinvite=no
;expire=240
[sip-out]
type=peer
host=sip.internode.on.net
context=internode-outgoing
externip= my static ip
;username= phone number
fromuser= phone number
;fromdomain=internode.on.net
;secret= secret
;qualify=yes
canreinvite=no
;auth= phone number:secret@BroadWorks
;nat=never
;pedantic=yes
;insecure=invite,port
;ignoresdpversion=yes
;compactheaders=yes
As you can see I've tried lots of settings. It registers and peers with
the provider, but no outgoing. The provider can call me though.
In extensions.conf:
[internode-outgoing]
exten=_X.,1,Dial(SIP/${EXTEN}@sip-out)
exten=_X.,n,Answer(2)
exten=_X.,n,Playback(ss-noservice)
With debugging enabled, verbose 9, debug 9:
SIP Debugging enabled
--- SIP read from UDP:my ata ip:5060 ---
INVITE sip:0871271201@asterisk server SIP/2.0
Via: SIP/2.0/UDP my ata ip:5060;branch=z9hG4bK-78cdde11;rport
From: my ata cid sip:my ata username@asterisk
server;tag=600053496208a4a8o1
To: sip:0871271201@asterisk server
Call-ID: e2895c9d-55b90b64@my ata ip
CSeq: 101 INVITE
Max-Forwards: 70
Contact: my ata cid sip:my ata username@my ata ip:5060
Expires: 240
User-Agent: Linksys/PAP2T-3.1.15(LS)
Content-Length: 446
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp
v=0
o=- 5330142 5330142 IN IP4 my ata ip
s=-
c=IN IP4 my ata ip
t=0 0
m=audio 16436 RTP/AVP 0 2 4 8 18 96 97 98 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/--- (14 headers 20 lines) ---
Sending to my ata ip:5060 (no NAT)
Using INVITE request as basis request - e2895c9d-55b90b64@my ata ip
Found peer 'my ata username' for 'my ata username' from my ata ip:5060
--- Reliably Transmitting (no NAT) to my ata ip:5060 ---
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP my ata ip:5060;branch=z9hG4bK-78cdde11;received=my
ata ip;rport=5060
From: Skinner's Home sip:my ata username@asterisk
server;tag=600053496208a4a8o1
To: sip:0871271201@asterisk server;tag=as6957dfb9
Call-ID: e2895c9d-55b90b64@192.168.0.196
CSeq: 101 INVITE
Server: Asterisk PBX 1.8.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=12eb6973
Content-Length: 0
Scheduling destruction of SIP dialog 'e2895c9d-55b90b64@my ata ip' in
6400 ms (Method: INVITE)
--- SIP read from UDP:my ata ip:5060 ---
ACK sip:0871271201@asterisk server SIP/2.0
Via: SIP/2.0/UDP my ata ip:5060;branch=z9hG4bK-78cdde11;rport
Re: [asterisk-users] Unable to make outgoing calls with Internode
Hi,
Under sip-out why do you have secret, fromdomain and NAT commented out ?
Also it seems like Asterisk is re-transmitting which means it seems like it
is not getting any response from your ISP. It could be a firewall issue, it
could be your ISP. If your ISP refuses to work with you you may want to go
with an ISP that will help.
Regards,
Dovid
- Original Message -
From: Da Rock asterisk-us...@herveybayaustralia.com.au
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, February 10, 2011 05:08
Subject: [asterisk-users] Unable to make outgoing calls with Internode
Surely there must be someone here who can help me with this problem.
I have spent weeks trying to get this damned service to work with no luck.
I have incoming calls working, but no outgoing. If get outgoing working
then incoming don't work.
I have sent this problem to this list a couple of times with little or no
response, and I _really_ need some help to sort it out.
I have an asterisk 1.8 server running on FreeBSD 8.1, and another FreeBSD
8.1 running as a firewall/gateway with PF.
I have a nodephone service with Internode (who have been absolutely
useless in helping me- they point blank refuse, or they say to open
everything right up to their server; which didn't wok anyway btw).
I have been running endless tests on settings changes, tcpdumps on both
the firewall and asterisk, and hours poring over SIP rfc's. I've only
managed to get a headache...
I have tried following best practices, worst practices, and still nothing
works.
My sip.conf looks like this:
[general]
context=default
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
allow=all
;allow=t140red
textsupport=yes
videosupport=yes
;allow=h263
maxcallbitrate=384
register=sip-in?phone
number:secret@sip.internode.on.net/phone number
externip= my static ip
localnet= my local subnet
canreinvite=no
hasvoicemail=no
qualify=yes
nat=no
;rtptimeout=120
rtpkeepalive=5
;ignoresdpversion=yes
;directmediapermit= my local subnet
[sip-in]
type=peer
host=sip.internode.on.net
context=internode-incoming
;externip= my static ip
;domain=internode.on.net,internode-incoming
;fromdomain=sip.internode.on.net
;fromuser= phone number
;username= phone number
;secret= secret
;auth= phone number:secret@BroadWorks
;insecure=invite,port
;register= phone number:secret@sip.internode.on.net
;nat=never
qualify=yes
canreinvite=no
;expire=240
[sip-out]
type=peer
host=sip.internode.on.net
context=internode-outgoing
externip= my static ip
;username= phone number
fromuser= phone number
;fromdomain=internode.on.net
;secret= secret
;qualify=yes
canreinvite=no
;auth= phone number:secret@BroadWorks
;nat=never
;pedantic=yes
;insecure=invite,port
;ignoresdpversion=yes
;compactheaders=yes
As you can see I've tried lots of settings. It registers and peers with
the provider, but no outgoing. The provider can call me though.
In extensions.conf:
[internode-outgoing]
exten=_X.,1,Dial(SIP/${EXTEN}@sip-out)
exten=_X.,n,Answer(2)
exten=_X.,n,Playback(ss-noservice)
With debugging enabled, verbose 9, debug 9:
SIP Debugging enabled
--- SIP read from UDP:my ata ip:5060 ---
INVITE sip:0871271201@asterisk server SIP/2.0
Via: SIP/2.0/UDP my ata ip:5060;branch=z9hG4bK-78cdde11;rport
From: my ata cid sip:my ata username@asterisk
server;tag=600053496208a4a8o1
To: sip:0871271201@asterisk server
Call-ID: e2895c9d-55b90b64@my ata ip
CSeq: 101 INVITE
Max-Forwards: 70
Contact: my ata cid sip:my ata username@my ata ip:5060
Expires: 240
User-Agent: Linksys/PAP2T-3.1.15(LS)
Content-Length: 446
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp
v=0
o=- 5330142 5330142 IN IP4 my ata ip
s=-
c=IN IP4 my ata ip
t=0 0
m=audio 16436 RTP/AVP 0 2 4 8 18 96 97 98 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/--- (14 headers 20 lines) ---
Sending to my ata ip:5060 (no NAT)
Using INVITE request as basis request - e2895c9d-55b90b64@my ata ip
Found peer 'my ata username' for 'my ata username' from my ata
ip:5060
--- Reliably Transmitting (no NAT) to my ata ip:5060 ---
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP my ata ip:5060;branch=z9hG4bK-78cdde11;received=my ata
ip;rport=5060
From: Skinner's Home sip:my
Re: [asterisk-users] Unable to make outgoing calls with Internode
On 02/10/11 14:00, Dovid Bender wrote:
Hi,
Under sip-out why do you have secret, fromdomain and NAT commented out ?
Also it seems like Asterisk is re-transmitting which means it seems
like it is not getting any response from your ISP. It could be a
firewall issue, it could be your ISP. If your ISP refuses to work with
you you may want to go with an ISP that will help.
Regards,
Dovid
Thanks Dovid. I've actually tried others and they're surprisingly worse.
I haven't stopped going through settings and logs, etc; but I just
looked at them again fresh. I noticed fragmented packets which the
firewall was dropping. I tweaked it and it came good- finally!
I suspected it was a the case, but I just could never find where. What
has me wondering now is, why is asterisk fragmenting packets? Any other
packets- from the same machine even- are ok. Is it a bug? Have I
seriously been tearing my hair out because of a bug?
The settings are commented out because I have tried them, discarded
them, tried them again differently I left them in here so you could
see I might have tried them at one stage.
One more question- what is the issue with having a single peer reference
in sip.conf? Why does it say have one incoming and one outgoing? Is it a
problem just having the one?
Thanks again Dovid, you saved my sanity. Having no one to confirm or
deny a supposition can be a pain at times. Your confirmation helped
immensely.
Cheers
- Original Message - From: Da Rock
asterisk-us...@herveybayaustralia.com.au
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, February 10, 2011 05:08
Subject: [asterisk-users] Unable to make outgoing calls with Internode
Surely there must be someone here who can help me with this problem.
I have spent weeks trying to get this damned service to work with no
luck. I have incoming calls working, but no outgoing. If get outgoing
working then incoming don't work.
I have sent this problem to this list a couple of times with little
or no response, and I _really_ need some help to sort it out.
I have an asterisk 1.8 server running on FreeBSD 8.1, and another
FreeBSD 8.1 running as a firewall/gateway with PF.
I have a nodephone service with Internode (who have been absolutely
useless in helping me- they point blank refuse, or they say to open
everything right up to their server; which didn't wok anyway btw).
I have been running endless tests on settings changes, tcpdumps on
both the firewall and asterisk, and hours poring over SIP rfc's. I've
only managed to get a headache...
I have tried following best practices, worst practices, and still
nothing works.
My sip.conf looks like this:
[general]
context=default
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
allow=all
;allow=t140red
textsupport=yes
videosupport=yes
;allow=h263
maxcallbitrate=384
register=sip-in?phone
number:secret@sip.internode.on.net/phone number
externip= my static ip
localnet= my local subnet
canreinvite=no
hasvoicemail=no
qualify=yes
nat=no
;rtptimeout=120
rtpkeepalive=5
;ignoresdpversion=yes
;directmediapermit= my local subnet
[sip-in]
type=peer
host=sip.internode.on.net
context=internode-incoming
;externip= my static ip
;domain=internode.on.net,internode-incoming
;fromdomain=sip.internode.on.net
;fromuser= phone number
;username= phone number
;secret= secret
;auth= phone number:secret@BroadWorks
;insecure=invite,port
;register= phone number:secret@sip.internode.on.net
;nat=never
qualify=yes
canreinvite=no
;expire=240
[sip-out]
type=peer
host=sip.internode.on.net
context=internode-outgoing
externip= my static ip
;username= phone number
fromuser= phone number
;fromdomain=internode.on.net
;secret= secret
;qualify=yes
canreinvite=no
;auth= phone number:secret@BroadWorks
;nat=never
;pedantic=yes
;insecure=invite,port
;ignoresdpversion=yes
;compactheaders=yes
As you can see I've tried lots of settings. It registers and peers
with the provider, but no outgoing. The provider can call me though.
In extensions.conf:
[internode-outgoing]
exten=_X.,1,Dial(SIP/${EXTEN}@sip-out)
exten=_X.,n,Answer(2)
exten=_X.,n,Playback(ss-noservice)
With debugging enabled, verbose 9, debug 9:
SIP Debugging enabled
--- SIP read from UDP:my ata ip:5060 ---
INVITE sip:0871271201@asterisk server SIP/2.0
Via: SIP/2.0/UDP my ata ip:5060;branch=z9hG4bK-78cdde11;rport
From: my ata cid sip:my ata username@asterisk
server;tag=600053496208a4a8o1
To: sip
