Update on this:
I found that by changing insecure = very to insecure = invite, adding
the second trunk no longer stopped calls working.
I've read the documentation on this switch and still don't see how it
applies/is meant to get used.
Anyway, with this change in place, the following may help:
asterisk*CLI> sip show registry
Host Username Refresh State
Reg.Time
gw02.mytel.net.au:5060 11111 120 Request Sent
gw02.mytel.net.au:5060 22222 105 Registered
Thu, 13 Sep 2007 23:33:47
I have set a dial plan so that some handsets use the 2222 (not the real
number) extension (which work) and now I only need to determine why
11111 never seems to register.
If I remove all traces of the 2222 connection from my config, 11111
registers fine.
Joshua Small | Senior Network Engineer | VisiNet | P. +61 1300 887 959 |
www.visinet.com.au <http://www.visinet.com.au/>
This e-mail is intended for use by the named recipients only and
contains confidential information. Opinions and other information in
this message that pertain to the sender's employer and its products and
services represent the opinion of the sender and not necessarily those
of the employer.
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Joshua
Small
Sent: Thursday, 13 September 2007 10:44 AM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] Problems with two trunks
Hi,
I am attempting to setup an asterisk server, current specs:
CentOS release 5 (Final)
Asterisk 1.4.11
Asterisk-gui checked out from SVN last week
I started with a fairly basic setup involving one VOIP provider who
provided one dial in number, and a couple of handsets. Config files are
below. It was pretty much totally built by Asterisk-gui, except for the
fact I had to add "insecure=very" into users.conf in order to stop the
dialin from our provider presenting an authentication error. Advice on
any more correct approach would be appreciated, but is not the focus of
this post:
Users.conf
;several handsets setup like this...
[6001]
callwaiting = yes
context = numberplan-custom-1
email = jsmall at visinet.com.au
fullname = Joshua Small
hasagent = yes
hasdirectory = yes
hasiax = no
hasmanager = no
hassip = yes
hasvoicemail = no
host = dynamic
mailbox = 6001
secret = XXXXX
threewaycalling = yes
registeriax = no
registersip = yes
canreinvite = no
nat = no
dtmfmode = rfc2833
vmsecret = 1234
;some PSTNS
[trunk_2]
callerid = asreceived
context = DID_trunk_2
group = 2
hasexten = no
hasiax = no
hassip = no
trunkname = Ports 1,2,3,4
trunkstyle = analog
zapchan = 1,2,3,4
;my IP trunk
[trunk_3]
allow = all
context = DID_trunk_3
dialformat = ${EXTEN:1}
hasexten = no
hasiax = no
hassip = yes
host = gw02.mytel.net.au
port = 5060
registeriax = no
registersip = yes
secret = XXXXXXXX
trunkname = Custom - MyTel2
trunkstyle = customvoip
username = XXXXXXXX
type = friend
nat = yes
;extensions.conf
[numberplan-custom-1]
plancomment = DialPlan1
include = default
include = parkedcalls
exten = _0XXXXX!,1,Macro(trunkdial,${trunk_3}/${EXTEN:1})
comment = _0XXXXX!,1,First,standard
;a failover to PSTN, not yet enabled
;exten = _0XXXXX!,2,Macro(trunkdial,${trunk_2}/0${EXTEN:1})
;comment = _0XXXXX!,1,First,standard
At this point, everything appears to work fine. We can make calls from
our several handsets using our voip link no problems.
We have two different accounts with our provider, the goal being certain
handsets will connect to this account and therefore be billed
separately. I haven't gotten as far as to add the extra handsets and set
an appropriate dialplan, all I did was add this to users.conf:
[trunk_extra]
allow = all
context = DID_trunk_3
dialformat = ${EXTEN:1}
hasexten = no
hasiax = no
hassip = yes
host = gw02.mytel.net.au
port = 5060
registeriax = no
registersip = yes
secret = XXXXXXXX
trunkname = Custom - MyTel Two
trunkstyle = customvoip
username = XXXXXXXXXX
type = friend
nat = yes
You can ignore this. I mistyped the password, and once it was fixed, and
registered correctly, both links failed to work again.
I have some extended information from sip debug. Again, this shows up as
soon as I try to register two connections.
<--- SIP read from 203.166.103.242:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP
192.168.107.4:5060;branch=z9hG4bK454ad99d;received=59.167.248.154;rport53487
From: "Joshua Small" <sip:8001 at 192.168.107.4>;tag=as3d465ba3
To: <sip:phonnumber at gw02.mytel.net.au>;tag=as5937f41d
Call-ID: 2f9f21865185cb9103ef86f438a79835 at 192.168.107.4
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
Joshua Small | Senior Network Engineer | VisiNet | P. +61 1300 887 959 |
www.visinet.com.au <http://www.visinet.com.au/>
This e-mail is intended for use by the named recipients only and
contains confidential information. Opinions and other information in
this message that pertain to the sender's employer and its products and
services represent the opinion of the sender and not necessarily those
of the employer.
From: Joshua Small
Sent: Thursday, 13 September 2007 1:38 PM
To: 'asterisk-users at lists.digium.com'
Subject: FW: [asterisk-users] Problems with two trunks
Update on this:
I found that by changing insecure = very to insecure = invite, adding
the second trunk no longer stopped calls working.
I've read the documentation on this switch and still don't see how it
applies/is meant to get used.
Anyway, with this change in place, the following may help:
asterisk*CLI> sip show registry
Host Username Refresh State
Reg.Time
gw02.mytel.net.au:5060 11111 120 Request Sent
gw02.mytel.net.au:5060 22222 105 Registered
Thu, 13 Sep 2007 23:33:47
I have set a dial plan so that some handsets use the 2222 (not the real
number) extension (which work) and now I only need to determine why
11111 never seems to register.
If I remove all traces of the 2222 connection from my config, 11111
registers fine.
Joshua Small | Senior Network Engineer | VisiNet | P. +61 1300 887 959 |
www.visinet.com.au <http://www.visinet.com.au/>
This e-mail is intended for use by the named recipients only and
contains confidential information. Opinions and other information in
this message that pertain to the sender's employer and its products and
services represent the opinion of the sender and not necessarily those
of the employer.
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Joshua
Small
Sent: Thursday, 13 September 2007 10:44 AM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] Problems with two trunks
Hi,
I am attempting to setup an asterisk server, current specs:
CentOS release 5 (Final)
Asterisk 1.4.11
Asterisk-gui checked out from SVN last week
I started with a fairly basic setup involving one VOIP provider who
provided one dial in number, and a couple of handsets. Config files are
below. It was pretty much totally built by Asterisk-gui, except for the
fact I had to add "insecure=very" into users.conf in order to stop the
dialin from our provider presenting an authentication error. Advice on
any more correct approach would be appreciated, but is not the focus of
this post:
Users.conf
;several handsets setup like this...
[6001]
callwaiting = yes
context = numberplan-custom-1
email = jsmall at visinet.com.au
fullname = Joshua Small
hasagent = yes
hasdirectory = yes
hasiax = no
hasmanager = no
hassip = yes
hasvoicemail = no
host = dynamic
mailbox = 6001
secret = XXXXX
threewaycalling = yes
registeriax = no
registersip = yes
canreinvite = no
nat = no
dtmfmode = rfc2833
vmsecret = 1234
;some PSTNS
[trunk_2]
callerid = asreceived
context = DID_trunk_2
group = 2
hasexten = no
hasiax = no
hassip = no
trunkname = Ports 1,2,3,4
trunkstyle = analog
zapchan = 1,2,3,4
;my IP trunk
[trunk_3]
allow = all
context = DID_trunk_3
dialformat = ${EXTEN:1}
hasexten = no
hasiax = no
hassip = yes
host = gw02.mytel.net.au
port = 5060
registeriax = no
registersip = yes
secret = XXXXXXXX
trunkname = Custom - MyTel2
trunkstyle = customvoip
username = XXXXXXXX
type = friend
nat = yes
;extensions.conf
[numberplan-custom-1]
plancomment = DialPlan1
include = default
include = parkedcalls
exten = _0XXXXX!,1,Macro(trunkdial,${trunk_3}/${EXTEN:1})
comment = _0XXXXX!,1,First,standard
;a failover to PSTN, not yet enabled
;exten = _0XXXXX!,2,Macro(trunkdial,${trunk_2}/0${EXTEN:1})
;comment = _0XXXXX!,1,First,standard
At this point, everything appears to work fine. We can make calls from
our several handsets using our voip link no problems.
We have two different accounts with our provider, the goal being certain
handsets will connect to this account and therefore be billed
separately. I haven't gotten as far as to add the extra handsets and set
an appropriate dialplan, all I did was add this to users.conf:
[trunk_extra]
allow = all
context = DID_trunk_3
dialformat = ${EXTEN:1}
hasexten = no
hasiax = no
hassip = yes
host = gw02.mytel.net.au
port = 5060
registeriax = no
registersip = yes
secret = XXXXXXXX
trunkname = Custom - MyTel Two
trunkstyle = customvoip
username = XXXXXXXXXX
type = friend
nat = yes
One more spam from yours truely..
Got it sorted out. My configuration never had a:
Fromuser = 1111111
In it. Worked fine for any individual trunk. But after sticking that
command in for each individual connection, I was able to register two of
them (and use them both).
Thanks for the suggestions, it looks like this was all I needed.
Joshua Small | Senior Network Engineer | VisiNet | P. +61 1300 887 959 |
www.visinet.com.au <http://www.visinet.com.au/>
This e-mail is intended for use by the named recipients only and
contains confidential information. Opinions and other information in
this message that pertain to the sender's employer and its products and
services represent the opinion of the sender and not necessarily those
of the employer.
From: Joshua Small
Sent: Thursday, 13 September 2007 3:07 PM
To: 'asterisk-users at lists.digium.com'
Subject: FW: [asterisk-users] Problems with two trunks
You can ignore this. I mistyped the password, and once it was fixed, and
registered correctly, both links failed to work again.
I have some extended information from sip debug. Again, this shows up as
soon as I try to register two connections.
<--- SIP read from 203.166.103.242:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP
192.168.107.4:5060;branch=z9hG4bK454ad99d;received=59.167.248.154;rport53487
From: "Joshua Small" <sip:8001 at 192.168.107.4>;tag=as3d465ba3
To: <sip:phonnumber at gw02.mytel.net.au>;tag=as5937f41d
Call-ID: 2f9f21865185cb9103ef86f438a79835 at 192.168.107.4
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
Joshua Small | Senior Network Engineer | VisiNet | P. +61 1300 887 959 |
www.visinet.com.au <http://www.visinet.com.au/>
This e-mail is intended for use by the named recipients only and
contains confidential information. Opinions and other information in
this message that pertain to the sender's employer and its products and
services represent the opinion of the sender and not necessarily those
of the employer.
From: Joshua Small
Sent: Thursday, 13 September 2007 1:38 PM
To: 'asterisk-users at lists.digium.com'
Subject: FW: [asterisk-users] Problems with two trunks
Update on this:
I found that by changing insecure = very to insecure = invite, adding
the second trunk no longer stopped calls working.
I've read the documentation on this switch and still don't see how it
applies/is meant to get used.
Anyway, with this change in place, the following may help:
asterisk*CLI> sip show registry
Host Username Refresh State
Reg.Time
gw02.mytel.net.au:5060 11111 120 Request Sent
gw02.mytel.net.au:5060 22222 105 Registered
Thu, 13 Sep 2007 23:33:47
I have set a dial plan so that some handsets use the 2222 (not the real
number) extension (which work) and now I only need to determine why
11111 never seems to register.
If I remove all traces of the 2222 connection from my config, 11111
registers fine.
Joshua Small | Senior Network Engineer | VisiNet | P. +61 1300 887 959 |
www.visinet.com.au <http://www.visinet.com.au/>
This e-mail is intended for use by the named recipients only and
contains confidential information. Opinions and other information in
this message that pertain to the sender's employer and its products and
services represent the opinion of the sender and not necessarily those
of the employer.
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Joshua
Small
Sent: Thursday, 13 September 2007 10:44 AM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] Problems with two trunks
Hi,
I am attempting to setup an asterisk server, current specs:
CentOS release 5 (Final)
Asterisk 1.4.11
Asterisk-gui checked out from SVN last week
I started with a fairly basic setup involving one VOIP provider who
provided one dial in number, and a couple of handsets. Config files are
below. It was pretty much totally built by Asterisk-gui, except for the
fact I had to add "insecure=very" into users.conf in order to stop the
dialin from our provider presenting an authentication error. Advice on
any more correct approach would be appreciated, but is not the focus of
this post:
Users.conf
;several handsets setup like this...
[6001]
callwaiting = yes
context = numberplan-custom-1
email = jsmall at visinet.com.au
fullname = Joshua Small
hasagent = yes
hasdirectory = yes
hasiax = no
hasmanager = no
hassip = yes
hasvoicemail = no
host = dynamic
mailbox = 6001
secret = XXXXX
threewaycalling = yes
registeriax = no
registersip = yes
canreinvite = no
nat = no
dtmfmode = rfc2833
vmsecret = 1234
;some PSTNS
[trunk_2]
callerid = asreceived
context = DID_trunk_2
group = 2
hasexten = no
hasiax = no
hassip = no
trunkname = Ports 1,2,3,4
trunkstyle = analog
zapchan = 1,2,3,4
;my IP trunk
[trunk_3]
allow = all
context = DID_trunk_3
dialformat = ${EXTEN:1}
hasexten = no
hasiax = no
hassip = yes
host = gw02.mytel.net.au
port = 5060
registeriax = no
registersip = yes
secret = XXXXXXXX
trunkname = Custom - MyTel2
trunkstyle = customvoip
username = XXXXXXXX
type = friend
nat = yes
;extensions.conf
[numberplan-custom-1]
plancomment = DialPlan1
include = default
include = parkedcalls
exten = _0XXXXX!,1,Macro(trunkdial,${trunk_3}/${EXTEN:1})
comment = _0XXXXX!,1,First,standard
;a failover to PSTN, not yet enabled
;exten = _0XXXXX!,2,Macro(trunkdial,${trunk_2}/0${EXTEN:1})
;comment = _0XXXXX!,1,First,standard
At this point, everything appears to work fine. We can make calls from
our several handsets using our voip link no problems.
We have two different accounts with our provider, the goal being certain
handsets will connect to this account and therefore be billed
separately. I haven't gotten as far as to add the extra handsets and set
an appropriate dialplan, all I did was add this to users.conf:
[trunk_extra]
allow = all
context = DID_trunk_3
dialformat = ${EXTEN:1}
hasexten = no
hasiax = no
hassip = yes
host = gw02.mytel.net.au
port = 5060
registeriax = no
registersip = yes
secret = XXXXXXXX
trunkname = Custom - MyTel Two
trunkstyle = customvoip
username = XXXXXXXXXX
type = friend
nat = yes