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Message |
chatter8712 at gmail.com
Guest
|
 Posted: Mon Jan 07, 2008 8:02 pm Post subject: [asterisk-users] FWD and IPCall |
 |
|
Okay. What do you mean in step 4/5 (I don't remember which) where you
write something about "Use your IPKall number as the sip number" I am
signing up for IPKall... Right?
On 1/7/08, Shane D <chatter8712 at gmail.com> wrote:
| Quote: | no-ip.org appears to want to charge me money... Is there a free alternative?
On 1/7/08, Huw Richards <huw.richards at oprig.com> wrote:
| Quote: |
If you want to forward your ipkall number directly to your asterisk
server:
1. If your asterisk server is on a private LAN and is connected to the
internet via a router, enable the router to port forward UDP/5060 &
UDP/10000-20000 to your asterisk server (assuming you have not changed
rtp config parameters in rtp.conf).
2. Check that the firewall (if any) on your asterisk server allows
connections on UDP/5060 & UDP/10000-20000
3a. Static public IP address - use the fully qualified domain name
assigned to the IP address (or setup an account on www.no-ip.org with a
name of your choice)
3b. Dynamic public IP address - setup an account on www.no-ip.org with a
name of your choice - install the dynamic ip address update client to
monitor any change of your ip address (downloads & instructions on
no-ip.org website)
4. Goto www.ipkall.com and login to your account. Use your ipkall number
as the SIP Phone Number and then the name you selected in 3a or 3b as
the SIP Proxy.
5. Wait 60 minutes for changes to take affect (!)
6. Edit asterisk sip configuration to allow calls from ipkall:
vi /etc/asterisk/sip.conf and find the section beginning [general]
Add/replace the following:
externhost=the name you setup in 3a. or 3b.
localnet=your private LAN e.g. 192.168.2.0/255.255.255.0
Add a new section at the bottom of the file:
[ipkall.com]
host=voiper.ipkall.com
context=from-ipkall
dtmfmode=rfc2833
insecure=invite
type=friend
canreinvite=no
disallow=all
allow=ulaw ; you can add other codecs if you want once the setup works
Save the file. The section you added tells asterisk to accept calls from
voiper.ipkall.com and to place them in the "from-ipkall" context. This
context can be whatever you want. You may need to change the insecure=
line if you are using asterisk 1.2
7. Edit asterisk dialplan configuration to handle calls from ipkall:
vi /etc/asterisk/extensions.conf and add at the bottom:
[from-ipkall]
exten => <IPKALL-NUMBER>,1,NoOp(from-ipkall)
exten => <IPKALL-NUMBER>,2,NoOp(${CALLERIDNAME}/${CALLERIDNUM})
exten => <IPKALL-NUMBER>,3,Dial(Local/200 at internal)
Save the file. The section you added tells asterisk what to do with
calls that are received in the "from-ipkall" context. Replace the
<IPKALL-NUMBER> with whatever you entered in the SIP Phone number field
on the ipkall website (I recommended your ipkall number).
In the "from-ipkall" section:
1: display "from-ipkall" on the console
2: display the caller id & name
3. phone the local extension 200 in context "local" - replace this line
with your personal requirements.
Connect to the asterisk console (asterisk -R on my server) and "sip
reload" followed by "dialplan reload" (asterisk 1.4) or "extensions
reload" (asterisk 1.2). "sip reload" will re-read the sip.conf file &
"dialplan reload"/"extensions reload" will re-read the extensions.conf
file.
Phone your ipkall number and see if anything is displayed on the console
and/or your phone rings.
If nothing on the console when you phone, try "sip set debug peer
ipkall.com" (asterisk 1.4 - not sure of the command for asterisk 1.2)
and phone again.
Post back your results.
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Shane D
Sent: Monday, January 07, 2008 17:32
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FWD and IPCall
Okay... That was kind of confusing. Would you contact me off-list to
help me specifically?
I've double-checked everything for the IAX, and it's a no-go. Maybe
I'll try this SIP thing. But then again, if I can just hook IPKall to
the server directly, I don't need FWD...
On 1/7/08, Huw Richards <huw.richards at oprig.com> wrote:
| Quote: | My config is as follows
Excerpt of sip.conf:
[general]
externhost=fully.qualified.domain.name
localnet=192.168.2.0/255.255.255.0
srvlookup=no
defaultexpiry=3600
dtmfmode=rfc2833
register => <fwd-id>:<fwd-pwd>@fwd.pulver.com/<fwd-id>
[sipfwd]
type=peer
secret=<fwd-pwd>
username=<fwd-id>
fromdomain=fwd.pulver.com
host=fwd.pulver.com
disallow=all
allow=ulaw
canreinvite=yes
insecure=invite
qualify=yes
context=from-fwd
Excerpt of extensions.conf:
[from-fwd]
exten => <fwd-id>,1,NoOp(from-fwd)
exten => <fwd-id>,n,Dial(whatever)
I have a dynamic public IP address, so I use http://www.no-ip.org to
|
map
| Quote: | my IP address to name. My router port forwards UDP/5060 &
UDP/10000-20000 to the internal asterisk server.
However, I do not have ipkall forwarding to my fwd account. I have it
forwarding directly to my asterisk server using the no-ip.org address
|
I
| Quote: | registered.
e.g. forward to sip:xxx at fully.qualified.domain.name on ipkall website
and then in sip.conf:
[ipkall.com]
host=voiper.ipkall.com
context=from-ipkall
dtmfmode=rfc2833
insecure=invite
type=friend
canreinvite=no
disallow=all
allow=ulaw
And in extensions.conf:
[from-ipkall]
exten => xxx,1,NoOp(from-ipkall)
exten => xxx,n,Dial(whatever)
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Shane D
Sent: Monday, January 07, 2008 12:09
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FWD and IPCall
It's Iax2. Is there a way of using amore reliable sip
connectoin/something slightly different?
If so, how would I go about that.
On 1/7/08, Huw Richards <huw.richards at oprig.com> wrote:
| Quote: | You haven't said if your connection to fwd is SIP or IAX2 but I have
found IAX2 connections to fwd to be unreliable. Other people may
|
|
have
| Quote: | | Quote: | different results.
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Shane
|
|
D
| Quote: | | Quote: | Sent: Monday, January 07, 2008 10:17
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] FWD and IPCall
Hello All,
I have a problem. I have tried everything that is in the book "The
Future of Telephony" as well as on the FWD (freeworlddialup)
|
|
website,
| Quote: | | Quote: | and there is still a problem. My asterisk box is not able to
|
|
associate
| Quote: | | Quote: | with the FWD server. I get:
Registration Rejected by [insert IP], and I can't use my IPCall
|
|
number
--
-Shane
Blog: http://blind-geek.com/blog/
CoOwner: http://sjtechzone.com
AIM: inhaddict
Skype: chatter8712
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
|
--
-Shane
Blog: http://blind-geek.com/blog/
CoOwner: http://sjtechzone.com
AIM: inhaddict
Skype: chatter8712
|
--
-Shane
Blog: http://blind-geek.com/blog/
CoOwner: http://sjtechzone.com
AIM: inhaddict
Skype: chatter8712 |
|
| Back to top |
|
 |
matt at venturevoip.com
Guest
|
| Posted: Mon Jan 07, 2008 8:10 pm Post subject: [asterisk-users] FWD and IPCall |
|
|
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Shane D wrote:
| Quote: | no-ip.org appears to want to charge me money... Is there a free alternative?
|
Dyndns.org
- --
Kind Regards,
Matt Riddell
Director
_______________________________________________
http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss)
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.7 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org
iD8DBQFHgs2NDQNt8rg0Kp4RAvJIAJ9ZXBIHESvIggx/SebD/fepyJr2xgCfeChs
3dc37G8IEH3qSQejV1UKATM=
=jBOo
-----END PGP SIGNATURE----- |
|
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|
|
chatter8712 at gmail.com
Guest
|
| Posted: Mon Jan 07, 2008 8:22 pm Post subject: [asterisk-users] FWD and IPCall |
|
|
I did everything, and when I dial, nothing comes up in the consol,
nothing rings, and the phone says "I'm sorry, but the person you are
trying to call has a mailbox that has not been configured yet. Good
bye."
What's wrong?
On 1/7/08, Shane D <chatter8712 at gmail.com> wrote:
| Quote: | Okay. What do you mean in step 4/5 (I don't remember which) where you
write something about "Use your IPKall number as the sip number" I am
signing up for IPKall... Right?
On 1/7/08, Shane D <chatter8712 at gmail.com> wrote:
| Quote: | no-ip.org appears to want to charge me money... Is there a free
|
alternative?
| Quote: |
On 1/7/08, Huw Richards <huw.richards at oprig.com> wrote:
| Quote: |
If you want to forward your ipkall number directly to your asterisk
server:
1. If your asterisk server is on a private LAN and is connected to the
internet via a router, enable the router to port forward UDP/5060 &
UDP/10000-20000 to your asterisk server (assuming you have not changed
rtp config parameters in rtp.conf).
2. Check that the firewall (if any) on your asterisk server allows
connections on UDP/5060 & UDP/10000-20000
3a. Static public IP address - use the fully qualified domain name
assigned to the IP address (or setup an account on www.no-ip.org with a
name of your choice)
3b. Dynamic public IP address - setup an account on www.no-ip.org with a
name of your choice - install the dynamic ip address update client to
monitor any change of your ip address (downloads & instructions on
no-ip.org website)
4. Goto www.ipkall.com and login to your account. Use your ipkall number
as the SIP Phone Number and then the name you selected in 3a or 3b as
the SIP Proxy.
5. Wait 60 minutes for changes to take affect (!)
6. Edit asterisk sip configuration to allow calls from ipkall:
vi /etc/asterisk/sip.conf and find the section beginning [general]
Add/replace the following:
externhost=the name you setup in 3a. or 3b.
localnet=your private LAN e.g. 192.168.2.0/255.255.255.0
Add a new section at the bottom of the file:
[ipkall.com]
host=voiper.ipkall.com
context=from-ipkall
dtmfmode=rfc2833
insecure=invite
type=friend
canreinvite=no
disallow=all
allow=ulaw ; you can add other codecs if you want once the setup works
Save the file. The section you added tells asterisk to accept calls from
voiper.ipkall.com and to place them in the "from-ipkall" context. This
context can be whatever you want. You may need to change the insecure=
line if you are using asterisk 1.2
7. Edit asterisk dialplan configuration to handle calls from ipkall:
vi /etc/asterisk/extensions.conf and add at the bottom:
[from-ipkall]
exten => <IPKALL-NUMBER>,1,NoOp(from-ipkall)
exten => <IPKALL-NUMBER>,2,NoOp(${CALLERIDNAME}/${CALLERIDNUM})
exten => <IPKALL-NUMBER>,3,Dial(Local/200 at internal)
Save the file. The section you added tells asterisk what to do with
calls that are received in the "from-ipkall" context. Replace the
<IPKALL-NUMBER> with whatever you entered in the SIP Phone number field
on the ipkall website (I recommended your ipkall number).
In the "from-ipkall" section:
1: display "from-ipkall" on the console
2: display the caller id & name
3. phone the local extension 200 in context "local" - replace this line
with your personal requirements.
Connect to the asterisk console (asterisk -R on my server) and "sip
reload" followed by "dialplan reload" (asterisk 1.4) or "extensions
reload" (asterisk 1.2). "sip reload" will re-read the sip.conf file &
"dialplan reload"/"extensions reload" will re-read the extensions.conf
file.
Phone your ipkall number and see if anything is displayed on the console
and/or your phone rings.
If nothing on the console when you phone, try "sip set debug peer
ipkall.com" (asterisk 1.4 - not sure of the command for asterisk 1.2)
and phone again.
Post back your results.
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Shane D
Sent: Monday, January 07, 2008 17:32
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FWD and IPCall
Okay... That was kind of confusing. Would you contact me off-list to
help me specifically?
I've double-checked everything for the IAX, and it's a no-go. Maybe
I'll try this SIP thing. But then again, if I can just hook IPKall to
the server directly, I don't need FWD...
On 1/7/08, Huw Richards <huw.richards at oprig.com> wrote:
| Quote: | My config is as follows
Excerpt of sip.conf:
[general]
externhost=fully.qualified.domain.name
localnet=192.168.2.0/255.255.255.0
srvlookup=no
defaultexpiry=3600
dtmfmode=rfc2833
register => <fwd-id>:<fwd-pwd>@fwd.pulver.com/<fwd-id>
[sipfwd]
type=peer
secret=<fwd-pwd>
username=<fwd-id>
fromdomain=fwd.pulver.com
host=fwd.pulver.com
disallow=all
allow=ulaw
canreinvite=yes
insecure=invite
qualify=yes
context=from-fwd
Excerpt of extensions.conf:
[from-fwd]
exten => <fwd-id>,1,NoOp(from-fwd)
exten => <fwd-id>,n,Dial(whatever)
I have a dynamic public IP address, so I use http://www.no-ip.org to
|
map
| Quote: | my IP address to name. My router port forwards UDP/5060 &
UDP/10000-20000 to the internal asterisk server.
However, I do not have ipkall forwarding to my fwd account. I have it
forwarding directly to my asterisk server using the no-ip.org address
|
I
| Quote: | registered.
e.g. forward to sip:xxx at fully.qualified.domain.name on ipkall website
and then in sip.conf:
[ipkall.com]
host=voiper.ipkall.com
context=from-ipkall
dtmfmode=rfc2833
insecure=invite
type=friend
canreinvite=no
disallow=all
allow=ulaw
And in extensions.conf:
[from-ipkall]
exten => xxx,1,NoOp(from-ipkall)
exten => xxx,n,Dial(whatever)
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Shane D
Sent: Monday, January 07, 2008 12:09
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FWD and IPCall
It's Iax2. Is there a way of using amore reliable sip
connectoin/something slightly different?
If so, how would I go about that.
On 1/7/08, Huw Richards <huw.richards at oprig.com> wrote:
| Quote: | You haven't said if your connection to fwd is SIP or IAX2 but I have
found IAX2 connections to fwd to be unreliable. Other people may
|
|
have
| Quote: | | Quote: | different results.
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Shane
|
|
D
| Quote: | | Quote: | Sent: Monday, January 07, 2008 10:17
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] FWD and IPCall
Hello All,
I have a problem. I have tried everything that is in the book "The
Future of Telephony" as well as on the FWD (freeworlddialup)
|
|
website,
| Quote: | | Quote: | and there is still a problem. My asterisk box is not able to
|
|
associate
| Quote: | | Quote: | with the FWD server. I get:
Registration Rejected by [insert IP], and I can't use my IPCall
|
|
number
--
-Shane
Blog: http://blind-geek.com/blog/
CoOwner: http://sjtechzone.com
AIM: inhaddict
Skype: chatter8712
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
|
--
-Shane
Blog: http://blind-geek.com/blog/
CoOwner: http://sjtechzone.com
AIM: inhaddict
Skype: chatter8712
|
--
-Shane
Blog: http://blind-geek.com/blog/
CoOwner: http://sjtechzone.com
AIM: inhaddict
Skype: chatter8712
|
--
-Shane
Blog: http://blind-geek.com/blog/
CoOwner: http://sjtechzone.com
AIM: inhaddict
Skype: chatter8712 |
|
| Back to top |
|
|
huw.richards at oprig.com
Guest
|
| Posted: Mon Jan 07, 2008 8:26 pm Post subject: [asterisk-users] FWD and IPCall |
|
|
I think you said that you already had an ipkall account? If so, logon to
the account and on the resulting screen there are 2 fields that you need
to change:
The first is the "SIP Phone Number" - if you have already tried to
forward the IPKALL number to your fwd account, this field will contain
your fwd number. You can change this to be anything - my convention is
to use the actual phone number that IPKALL assigned me.
The second is "SIP Proxy" - again, if you have tried to forward the
IPKALL number to your fwd account, this field will contain
"fwd.pulver.net" (or something similar). Change this value to the
hostname you setup at no-ip.org.
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Shane D
Sent: Monday, January 07, 2008 20:03
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FWD and IPCall
Okay. What do you mean in step 4/5 (I don't remember which) where you
write something about "Use your IPKall number as the sip number" I am
signing up for IPKall... Right?
On 1/7/08, Shane D <chatter8712 at gmail.com> wrote:
| Quote: | no-ip.org appears to want to charge me money... Is there a free
|
alternative?
| Quote: |
On 1/7/08, Huw Richards <huw.richards at oprig.com> wrote:
| Quote: |
If you want to forward your ipkall number directly to your asterisk
server:
1. If your asterisk server is on a private LAN and is connected to
|
|
the
| Quote: | | Quote: | internet via a router, enable the router to port forward UDP/5060 &
UDP/10000-20000 to your asterisk server (assuming you have not
|
|
changed
| Quote: | | Quote: | rtp config parameters in rtp.conf).
2. Check that the firewall (if any) on your asterisk server allows
connections on UDP/5060 & UDP/10000-20000
3a. Static public IP address - use the fully qualified domain name
assigned to the IP address (or setup an account on www.no-ip.org
|
|
with a
| Quote: | | Quote: | name of your choice)
3b. Dynamic public IP address - setup an account on www.no-ip.org
|
|
with a
| Quote: | | Quote: | name of your choice - install the dynamic ip address update client
|
|
to
| Quote: | | Quote: | monitor any change of your ip address (downloads & instructions on
no-ip.org website)
4. Goto www.ipkall.com and login to your account. Use your ipkall
|
|
number
| Quote: | | Quote: | as the SIP Phone Number and then the name you selected in 3a or 3b
|
|
as
| Quote: | | Quote: | the SIP Proxy.
5. Wait 60 minutes for changes to take affect (!)
6. Edit asterisk sip configuration to allow calls from ipkall:
vi /etc/asterisk/sip.conf and find the section beginning [general]
Add/replace the following:
externhost=the name you setup in 3a. or 3b.
localnet=your private LAN e.g. 192.168.2.0/255.255.255.0
Add a new section at the bottom of the file:
[ipkall.com]
host=voiper.ipkall.com
context=from-ipkall
dtmfmode=rfc2833
insecure=invite
type=friend
canreinvite=no
disallow=all
allow=ulaw ; you can add other codecs if you want once the setup
|
|
works
| Quote: | | Quote: |
Save the file. The section you added tells asterisk to accept calls
|
|
from
| Quote: | | Quote: | voiper.ipkall.com and to place them in the "from-ipkall" context.
|
|
This
| Quote: | | Quote: | context can be whatever you want. You may need to change the
|
|
insecure=
| Quote: | | Quote: | line if you are using asterisk 1.2
7. Edit asterisk dialplan configuration to handle calls from ipkall:
vi /etc/asterisk/extensions.conf and add at the bottom:
[from-ipkall]
exten => <IPKALL-NUMBER>,1,NoOp(from-ipkall)
exten => <IPKALL-NUMBER>,2,NoOp(${CALLERIDNAME}/${CALLERIDNUM})
exten => <IPKALL-NUMBER>,3,Dial(Local/200 at internal)
Save the file. The section you added tells asterisk what to do with
calls that are received in the "from-ipkall" context. Replace the
<IPKALL-NUMBER> with whatever you entered in the SIP Phone number
|
|
field
| Quote: | | Quote: | on the ipkall website (I recommended your ipkall number).
In the "from-ipkall" section:
1: display "from-ipkall" on the console
2: display the caller id & name
3. phone the local extension 200 in context "local" - replace this
|
|
line
| Quote: | | Quote: | with your personal requirements.
Connect to the asterisk console (asterisk -R on my server) and "sip
reload" followed by "dialplan reload" (asterisk 1.4) or "extensions
reload" (asterisk 1.2). "sip reload" will re-read the sip.conf file
|
|
&
| Quote: | | Quote: | "dialplan reload"/"extensions reload" will re-read the
|
|
extensions.conf
| Quote: | | Quote: | file.
Phone your ipkall number and see if anything is displayed on the
|
|
console
| Quote: | | Quote: | and/or your phone rings.
If nothing on the console when you phone, try "sip set debug peer
ipkall.com" (asterisk 1.4 - not sure of the command for asterisk
|
|
1.2)
| Quote: | | Quote: | and phone again.
Post back your results.
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Shane
|
|
D
| Quote: | | Quote: | Sent: Monday, January 07, 2008 17:32
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FWD and IPCall
Okay... That was kind of confusing. Would you contact me off-list to
help me specifically?
I've double-checked everything for the IAX, and it's a no-go. Maybe
I'll try this SIP thing. But then again, if I can just hook IPKall
|
|
to
| Quote: | | Quote: | the server directly, I don't need FWD...
On 1/7/08, Huw Richards <huw.richards at oprig.com> wrote:
| Quote: | My config is as follows
Excerpt of sip.conf:
[general]
externhost=fully.qualified.domain.name
localnet=192.168.2.0/255.255.255.0
srvlookup=no
defaultexpiry=3600
dtmfmode=rfc2833
register => <fwd-id>:<fwd-pwd>@fwd.pulver.com/<fwd-id>
[sipfwd]
type=peer
secret=<fwd-pwd>
username=<fwd-id>
fromdomain=fwd.pulver.com
host=fwd.pulver.com
disallow=all
allow=ulaw
canreinvite=yes
insecure=invite
qualify=yes
context=from-fwd
Excerpt of extensions.conf:
[from-fwd]
exten => <fwd-id>,1,NoOp(from-fwd)
exten => <fwd-id>,n,Dial(whatever)
I have a dynamic public IP address, so I use http://www.no-ip.org
|
|
|
to
| Quote: | | Quote: | map
| Quote: | my IP address to name. My router port forwards UDP/5060 &
UDP/10000-20000 to the internal asterisk server.
However, I do not have ipkall forwarding to my fwd account. I have
|
|
|
it
| Quote: | | Quote: | | Quote: | forwarding directly to my asterisk server using the no-ip.org
|
|
|
address
| Quote: | | Quote: | I
| Quote: | registered.
e.g. forward to sip:xxx at fully.qualified.domain.name on ipkall
|
|
|
website
| Quote: | | Quote: | | Quote: | and then in sip.conf:
[ipkall.com]
host=voiper.ipkall.com
context=from-ipkall
dtmfmode=rfc2833
insecure=invite
type=friend
canreinvite=no
disallow=all
allow=ulaw
And in extensions.conf:
[from-ipkall]
exten => xxx,1,NoOp(from-ipkall)
exten => xxx,n,Dial(whatever)
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of
|
|
|
Shane D
| Quote: | | Quote: | | Quote: | Sent: Monday, January 07, 2008 12:09
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FWD and IPCall
It's Iax2. Is there a way of using amore reliable sip
connectoin/something slightly different?
If so, how would I go about that.
On 1/7/08, Huw Richards <huw.richards at oprig.com> wrote:
| Quote: | You haven't said if your connection to fwd is SIP or IAX2 but I
|
|
|
|
have
| Quote: | | Quote: | | Quote: | | Quote: | found IAX2 connections to fwd to be unreliable. Other people may
|
|
have
| Quote: | | Quote: | different results.
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of
|
|
|
|
Shane
| Quote: | | Quote: | D
| Quote: | | Quote: | Sent: Monday, January 07, 2008 10:17
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] FWD and IPCall
Hello All,
I have a problem. I have tried everything that is in the book
|
|
|
|
"The
| Quote: | | Quote: | | Quote: | | Quote: | Future of Telephony" as well as on the FWD (freeworlddialup)
|
|
website,
| Quote: | | Quote: | and there is still a problem. My asterisk box is not able to
|
|
associate
| Quote: | | Quote: | with the FWD server. I get:
Registration Rejected by [insert IP], and I can't use my IPCall
|
|
number
| Quote: | | Quote: | to reach my Asterisk box.
Any suggestions?
--
-Shane
Blog: http://blind-geek.com/blog/
CoOwner: http://sjtechzone.com
AIM: inhaddict
Skype: chatter8712
_______________________________________________
--Bandwidth and Colocation Provided by
|
|
|
|
http://www.api-digital.com--
http://www.api-digital.com--
http://www.api-digital.com--
http://www.api-digital.com--
--
-Shane
Blog: http://blind-geek.com/blog/
CoOwner: http://sjtechzone.com
AIM: inhaddict
Skype: chatter8712
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
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|
|
chatter8712 at gmail.com
Guest
|
| Posted: Mon Jan 07, 2008 8:34 pm Post subject: [asterisk-users] FWD and IPCall |
|
|
I'm an idiot... I dialled wrong on my phone... I changed it, and was
able to use the Echo application. Dialling for a call to my softphone
as we speak!
On 1/7/08, Huw Richards <huw.richards at oprig.com> wrote:
| Quote: | I think you said that you already had an ipkall account? If so, logon to
the account and on the resulting screen there are 2 fields that you need
to change:
The first is the "SIP Phone Number" - if you have already tried to
forward the IPKALL number to your fwd account, this field will contain
your fwd number. You can change this to be anything - my convention is
to use the actual phone number that IPKALL assigned me.
The second is "SIP Proxy" - again, if you have tried to forward the
IPKALL number to your fwd account, this field will contain
"fwd.pulver.net" (or something similar). Change this value to the
hostname you setup at no-ip.org.
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Shane D
Sent: Monday, January 07, 2008 20:03
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FWD and IPCall
Okay. What do you mean in step 4/5 (I don't remember which) where you
write something about "Use your IPKall number as the sip number" I am
signing up for IPKall... Right?
On 1/7/08, Shane D <chatter8712 at gmail.com> wrote:
| Quote: | no-ip.org appears to want to charge me money... Is there a free
|
alternative?
| Quote: |
On 1/7/08, Huw Richards <huw.richards at oprig.com> wrote:
| Quote: |
If you want to forward your ipkall number directly to your asterisk
server:
1. If your asterisk server is on a private LAN and is connected to
|
|
the
| Quote: | | Quote: | internet via a router, enable the router to port forward UDP/5060 &
UDP/10000-20000 to your asterisk server (assuming you have not
|
|
changed
| Quote: | | Quote: | rtp config parameters in rtp.conf).
2. Check that the firewall (if any) on your asterisk server allows
connections on UDP/5060 & UDP/10000-20000
3a. Static public IP address - use the fully qualified domain name
assigned to the IP address (or setup an account on www.no-ip.org
|
|
with a
| Quote: | | Quote: | name of your choice)
3b. Dynamic public IP address - setup an account on www.no-ip.org
|
|
with a
| Quote: | | Quote: | name of your choice - install the dynamic ip address update client
|
|
to
| Quote: | | Quote: | monitor any change of your ip address (downloads & instructions on
no-ip.org website)
4. Goto www.ipkall.com and login to your account. Use your ipkall
|
|
number
| Quote: | | Quote: | as the SIP Phone Number and then the name you selected in 3a or 3b
|
|
as
| Quote: | | Quote: | the SIP Proxy.
5. Wait 60 minutes for changes to take affect (!)
6. Edit asterisk sip configuration to allow calls from ipkall:
vi /etc/asterisk/sip.conf and find the section beginning [general]
Add/replace the following:
externhost=the name you setup in 3a. or 3b.
localnet=your private LAN e.g. 192.168.2.0/255.255.255.0
Add a new section at the bottom of the file:
[ipkall.com]
host=voiper.ipkall.com
context=from-ipkall
dtmfmode=rfc2833
insecure=invite
type=friend
canreinvite=no
disallow=all
allow=ulaw ; you can add other codecs if you want once the setup
|
|
works
| Quote: | | Quote: |
Save the file. The section you added tells asterisk to accept calls
|
|
from
| Quote: | | Quote: | voiper.ipkall.com and to place them in the "from-ipkall" context.
|
|
This
| Quote: | | Quote: | context can be whatever you want. You may need to change the
|
|
insecure=
| Quote: | | Quote: | line if you are using asterisk 1.2
7. Edit asterisk dialplan configuration to handle calls from ipkall:
vi /etc/asterisk/extensions.conf and add at the bottom:
[from-ipkall]
exten => <IPKALL-NUMBER>,1,NoOp(from-ipkall)
exten => <IPKALL-NUMBER>,2,NoOp(${CALLERIDNAME}/${CALLERIDNUM})
exten => <IPKALL-NUMBER>,3,Dial(Local/200 at internal)
Save the file. The section you added tells asterisk what to do with
calls that are received in the "from-ipkall" context. Replace the
<IPKALL-NUMBER> with whatever you entered in the SIP Phone number
|
|
field
| Quote: | | Quote: | on the ipkall website (I recommended your ipkall number).
In the "from-ipkall" section:
1: display "from-ipkall" on the console
2: display the caller id & name
3. phone the local extension 200 in context "local" - replace this
|
|
line
| Quote: | | Quote: | with your personal requirements.
Connect to the asterisk console (asterisk -R on my server) and "sip
reload" followed by "dialplan reload" (asterisk 1.4) or "extensions
reload" (asterisk 1.2). "sip reload" will re-read the sip.conf file
|
|
&
| Quote: | | Quote: | "dialplan reload"/"extensions reload" will re-read the
|
|
extensions.conf
| Quote: | | Quote: | file.
Phone your ipkall number and see if anything is displayed on the
|
|
console
| Quote: | | Quote: | and/or your phone rings.
If nothing on the console when you phone, try "sip set debug peer
ipkall.com" (asterisk 1.4 - not sure of the command for asterisk
|
|
1.2)
| Quote: | | Quote: | and phone again.
Post back your results.
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Shane
|
|
D
| Quote: | | Quote: | Sent: Monday, January 07, 2008 17:32
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FWD and IPCall
Okay... That was kind of confusing. Would you contact me off-list to
help me specifically?
I've double-checked everything for the IAX, and it's a no-go. Maybe
I'll try this SIP thing. But then again, if I can just hook IPKall
|
|
to
| Quote: | | Quote: | the server directly, I don't need FWD...
On 1/7/08, Huw Richards <huw.richards at oprig.com> wrote:
| Quote: | My config is as follows
Excerpt of sip.conf:
[general]
externhost=fully.qualified.domain.name
localnet=192.168.2.0/255.255.255.0
srvlookup=no
defaultexpiry=3600
dtmfmode=rfc2833
register => <fwd-id>:<fwd-pwd>@fwd.pulver.com/<fwd-id>
[sipfwd]
type=peer
secret=<fwd-pwd>
username=<fwd-id>
fromdomain=fwd.pulver.com
host=fwd.pulver.com
disallow=all
allow=ulaw
canreinvite=yes
insecure=invite
qualify=yes
context=from-fwd
Excerpt of extensions.conf:
[from-fwd]
exten => <fwd-id>,1,NoOp(from-fwd)
exten => <fwd-id>,n,Dial(whatever)
I have a dynamic public IP address, so I use http://www.no-ip.org
|
|
|
to
| Quote: | | Quote: | map
| Quote: | my IP address to name. My router port forwards UDP/5060 &
UDP/10000-20000 to the internal asterisk server.
However, I do not have ipkall forwarding to my fwd account. I have
|
|
|
it
| Quote: | | Quote: | | Quote: | forwarding directly to my asterisk server using the no-ip.org
|
|
|
address
| Quote: | | Quote: | I
| Quote: | registered.
e.g. forward to sip:xxx at fully.qualified.domain.name on ipkall
|
|
|
website
| Quote: | | Quote: | | Quote: | and then in sip.conf:
[ipkall.com]
host=voiper.ipkall.com
context=from-ipkall
dtmfmode=rfc2833
insecure=invite
type=friend
canreinvite=no
disallow=all
allow=ulaw
And in extensions.conf:
[from-ipkall]
exten => xxx,1,NoOp(from-ipkall)
exten => xxx,n,Dial(whatever)
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of
|
|
|
Shane D
| Quote: | | Quote: | | Quote: | Sent: Monday, January 07, 2008 12:09
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FWD and IPCall
It's Iax2. Is there a way of using amore reliable sip
connectoin/something slightly different?
If so, how would I go about that.
On 1/7/08, Huw Richards <huw.richards at oprig.com> wrote:
| Quote: | You haven't said if your connection to fwd is SIP or IAX2 but I
|
|
|
|
have
| Quote: | | Quote: | | Quote: | | Quote: | found IAX2 connections to fwd to be unreliable. Other people may
|
|
have
| Quote: | | Quote: | different results.
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of
|
|
|
|
Shane
| Quote: | | Quote: | D
| Quote: | | Quote: | Sent: Monday, January 07, 2008 10:17
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] FWD and IPCall
Hello All,
I have a problem. I have tried everything that is in the book
|
|
|
|
"The
| Quote: | | Quote: | | Quote: | | Quote: | Future of Telephony" as well as on the FWD (freeworlddialup)
|
|
website,
| Quote: | | Quote: | and there is still a problem. My asterisk box is not able to
|
|
associate
| Quote: | | Quote: | with the FWD server. I get:
Registration Rejected by [insert IP], and I can't use my IPCall
|
|
number
| Quote: | | Quote: | to reach my Asterisk box.
Any suggestions?
--
-Shane
Blog: http://blind-geek.com/blog/
CoOwner: http://sjtechzone.com
AIM: inhaddict
Skype: chatter8712
_______________________________________________
--Bandwidth and Colocation Provided by
|
|
|
|
http://www.api-digital.com--
http://www.api-digital.com--
http://www.api-digital.com--
http://www.api-digital.com--
--
-Shane
Blog: http://blind-geek.com/blog/
CoOwner: http://sjtechzone.com
AIM: inhaddict
Skype: chatter8712
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
|
--
-Shane
Blog: http://blind-geek.com/blog/
CoOwner: http://sjtechzone.com
AIM: inhaddict
Skype: chatter8712 |
|
| Back to top |
|
|
huw.richards at oprig.com
Guest
|
| Posted: Mon Jan 07, 2008 8:35 pm Post subject: [asterisk-users] FWD and IPCall |
|
|
You're right that dyndns.org offers the same type of services as
no-ip.org.
However, when I first setup ipkall forwarding directly to my asterisk
server (about a year ago), it would not work with a dyndns.org account -
I forget the reason why. Maybe ipkall works with dyndns now - I haven't
tried it in over a year.
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Matt
Riddell
Sent: Monday, January 07, 2008 20:11
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FWD and IPCall
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Shane D wrote:
| Quote: | no-ip.org appears to want to charge me money... Is there a free
|
alternative?
Dyndns.org
- --
Kind Regards,
Matt Riddell
Director
_______________________________________________
http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss)
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.7 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org
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3dc37G8IEH3qSQejV1UKATM=
=jBOo
-----END PGP SIGNATURE-----
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asterisk-users mailing list
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|
chatter8712 at gmail.com
Guest
|
| Posted: Mon Jan 07, 2008 8:46 pm Post subject: [asterisk-users] FWD and IPCall |
|
|
It works! It works! It works!
I am able to talk to myself. Now all I have to do is write my dialplan...
Say, would I have to use the [ipkallnumber] extention? Could I specify
the "s" extension instead to catch multiple numbers?
--
-Shane
Blog: http://blind-geek.com/blog/
CoOwner: http://sjtechzone.com
AIM: inhaddict
Skype: chatter8712 |
|
| Back to top |
|
|
matt at venturevoip.com
Guest
|
| Posted: Mon Jan 07, 2008 8:50 pm Post subject: [asterisk-users] FWD and IPCall |
|
|
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Huw Richards wrote:
| Quote: | You're right that dyndns.org offers the same type of services as
no-ip.org.
However, when I first setup ipkall forwarding directly to my asterisk
server (about a year ago), it would not work with a dyndns.org account -
I forget the reason why. Maybe ipkall works with dyndns now - I haven't
tried it in over a year.
|
Must have not been registered.
I have both the free and commercial services set up on around 200 Linux
boxes distributed around the world 
- --
Kind Regards,
Matt Riddell
Director
_______________________________________________
http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss)
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.7 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org
iD8DBQFHgtbtDQNt8rg0Kp4RAu26AJ9QgqYk7TkHajOj7i18uXSGdI7CfQCfUzAS
SL7gJyqd9ZvvOBhZ5MPXZQI=
=qB/r
-----END PGP SIGNATURE----- |
|
| Back to top |
|
|
huw.richards at oprig.com
Guest
|
| Posted: Mon Jan 07, 2008 9:07 pm Post subject: [asterisk-users] FWD and IPCall |
|
|
I seem to recall the problem was related to DNS A vs SRV records. I
believe that dyndns at that time did not register SRV records on host
(i.e. free) accounts and ipkall was looking for an SRV record.
I know that an SRV record can be added on a paid account, but I still
don't think that you can add an SRV record to a free account - unless
all "A" records are automatically "SRV" records also.
Or mayble ipkall gracefully falls back to search for an "A" record if an
"SRV" lookup fails?
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Matt
Riddell
Sent: Monday, January 07, 2008 20:51
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FWD and IPCall
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Huw Richards wrote:
| Quote: | You're right that dyndns.org offers the same type of services as
no-ip.org.
However, when I first setup ipkall forwarding directly to my asterisk
server (about a year ago), it would not work with a dyndns.org account
|
-
| Quote: | I forget the reason why. Maybe ipkall works with dyndns now - I
|
haven't
| Quote: | tried it in over a year.
|
Must have not been registered.
I have both the free and commercial services set up on around 200 Linux
boxes distributed around the world
- --
Kind Regards,
Matt Riddell
Director
_______________________________________________
http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss)
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.7 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org
iD8DBQFHgtbtDQNt8rg0Kp4RAu26AJ9QgqYk7TkHajOj7i18uXSGdI7CfQCfUzAS
SL7gJyqd9ZvvOBhZ5MPXZQI=
=qB/r
-----END PGP SIGNATURE-----
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|
huw.richards at oprig.com
Guest
|
| Posted: Mon Jan 07, 2008 9:10 pm Post subject: [asterisk-users] FWD and IPCall |
|
|
You would have to try out the "s" extension yourself.
I tend to have different contexts for each incoming number (as a home
user, only one number at a provider) so I can potentially handle them
differently i.e. time of day check.
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Shane D
Sent: Monday, January 07, 2008 20:47
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FWD and IPCall
It works! It works! It works!
I am able to talk to myself. Now all I have to do is write my
dialplan...
Say, would I have to use the [ipkallnumber] extention? Could I specify
the "s" extension instead to catch multiple numbers?
--
-Shane
Blog: http://blind-geek.com/blog/
CoOwner: http://sjtechzone.com
AIM: inhaddict
Skype: chatter8712
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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|
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|
|
chatter8712 at gmail.com
Guest
|
| Posted: Mon Jan 07, 2008 9:42 pm Post subject: [asterisk-users] FWD and IPCall |
|
|
Oh. Well, I want to design a dialplan, and I don't care what number it
is, as long as you get my starting menu... So I'll try out the S
extension.
On 1/7/08, Huw Richards <huw.richards at oprig.com> wrote:
| Quote: | You would have to try out the "s" extension yourself.
I tend to have different contexts for each incoming number (as a home
user, only one number at a provider) so I can potentially handle them
differently i.e. time of day check.
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Shane D
Sent: Monday, January 07, 2008 20:47
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FWD and IPCall
It works! It works! It works!
I am able to talk to myself. Now all I have to do is write my
dialplan...
Say, would I have to use the [ipkallnumber] extention? Could I specify
the "s" extension instead to catch multiple numbers?
--
-Shane
Blog: http://blind-geek.com/blog/
CoOwner: http://sjtechzone.com
AIM: inhaddict
Skype: chatter8712
_______________________________________________
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Blog: http://blind-geek.com/blog/
CoOwner: http://sjtechzone.com
AIM: inhaddict
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