VoIP Mailing List Archives
Mailing list archives for the VoIP community |
|
View previous topic :: View next topic |
Author |
Message |
dan at amtelco.com Guest
|
Posted: Mon Dec 15, 2014 11:48 pm Post subject: [asterisk-users] PJSIP configuration question |
|
|
Thanks George.
I will correct my local_net in the morning.
Vitelity chan_sip settings I have working, do not have a fromuser.
sip.conf settings...
[HVout]
type=friend
dtmfmode=auto
host=64.2.142.93
disallow=all
allow=ulaw
canreinvite=no
trustrpid=yes
sendrpid=yes
nat=yes
context=TestApp
On Dec 15, 2014, at 9:32 PM, George Joseph <george.joseph@fairview5.com (george.joseph@fairview5.com)> wrote:
Quote: |
On Mon, Dec 15, 2014 at 7:34 PM, Dan Cropp <dan@amtelco.com (dan@amtelco.com)> wrote: Quote: |
I am not sure if I entered the correct settings for the transport information.
For the local_net, I entered my local ip address, but no mask. I will check with the network admin so he can verify the settings I entered.
| You need the network and mask. For example if the ip address and mask of the test machine is 192.168.0.1/255.255.255.0 then the correct entry would be 192.168.0.0/24.
Quote: |
One minor detail, we are using ip authentication. When Vitelity changed my account from user based authentication to IP based authentication, they stopped including a user for the account.
Should these settings work without the from_user (IP based authentication) or do I need to get the account name from Vitelity?
|
You definitely need the master account login username. If you has this working with chan_sip, then try the 'fromuser' from sip.conf and user is from_user.
--
_____________________________________________________________________
-- 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
|
|
|
Back to top |
|
|
george.joseph at fairv... Guest
|
Posted: Tue Dec 16, 2014 12:14 am Post subject: [asterisk-users] PJSIP configuration question |
|
|
On Mon, Dec 15, 2014 at 9:48 PM, Dan Cropp <dan@amtelco.com (dan@amtelco.com)> wrote: Quote: | Thanks George.
I will correct my local_net in the morning.
Vitelity chan_sip settings I have working, do not have a fromuser.
sip.conf settings...
| I think you can actually specify anything, it just has to be populated with something other than a sub-account username.
Quote: |
[HVout]
type=friend
dtmfmode=auto
host=64.2.142.93
disallow=all
allow=ulaw
canreinvite=no
trustrpid=yes
sendrpid=yes
nat=yes
context=TestApp
On Dec 15, 2014, at 9:32 PM, George Joseph <george.joseph@fairview5.com (george.joseph@fairview5.com)> wrote:
Quote: |
On Mon, Dec 15, 2014 at 7:34 PM, Dan Cropp <dan@amtelco.com (dan@amtelco.com)> wrote: Quote: |
I am not sure if I entered the correct settings for the transport information.
For the local_net, I entered my local ip address, but no mask. I will check with the network admin so he can verify the settings I entered.
| You need the network and mask. For example if the ip address and mask of the test machine is 192.168.0.1/255.255.255.0 then the correct entry would be 192.168.0.0/24.
Quote: |
One minor detail, we are using ip authentication. When Vitelity changed my account from user based authentication to IP based authentication, they stopped including a user for the account.
Should these settings work without the from_user (IP based authentication) or do I need to get the account name from Vitelity?
|
You definitely need the master account login username. If you has this working with chan_sip, then try the 'fromuser' from sip.conf and user is from_user.
|
--
_____________________________________________________________________
-- 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
|
|
|
Back to top |
|
|
dan at amtelco.com Guest
|
Posted: Tue Dec 16, 2014 9:16 am Post subject: [asterisk-users] PJSIP configuration question |
|
|
Thanks George.
I will give it a try.
Have a great day!
Dan
From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of George Joseph
Sent: Monday, December 15, 2014 11:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PJSIP configuration question
I think you can actually specify anything, it just has to be populated with something other than a sub-account username. |
|
Back to top |
|
|
dan at amtelco.com Guest
|
Posted: Tue Dec 16, 2014 11:01 am Post subject: [asterisk-users] PJSIP configuration question |
|
|
I corrected my local_net setting (based on advice from network admin).
I have tried several different values for the from_user and still have the same problem.
Asterisk receives the OK from Vitelity.
Asterisk sends the ACK (without a Contact header).
Vitelity doesn’t seem to process it, so they send an OK again.
The OK receive, Transmit ACK occurs 4 times.
A short while later, Vitelity hangs up on my cell phone.
Asterisk is never told the call is gone.
If I hangup the call from Asterisk side,
Asterisk sends the BYE message.
Vitelity responds with a “SIP/2.0 481 Call leg/transaction does not exist”
Again, the trace indicates the ACK message is missing the Contact header.
Additional note: the network admin is asking why the local_net, external_media_address, and external_signalling_address are needed. He wrote me…“You should NOT have to know your public IP. The firewall should be doing fixup commands to change the values in the packet”
From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of George Joseph
Sent: Monday, December 15, 2014 11:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PJSIP configuration question
On Mon, Dec 15, 2014 at 9:48 PM, Dan Cropp <dan@amtelco.com (dan@amtelco.com)> wrote:
Thanks George.
I will correct my local_net in the morning.
Vitelity chan_sip settings I have working, do not have a fromuser.
sip.conf settings...
I think you can actually specify anything, it just has to be populated with something other than a sub-account username.
Quote: |
[HVout]
type=friend
dtmfmode=auto
host=64.2.142.93
disallow=all
allow=ulaw
canreinvite=no
trustrpid=yes
sendrpid=yes
nat=yes
context=TestApp
On Dec 15, 2014, at 9:32 PM, George Joseph <george.joseph@fairview5.com (george.joseph@fairview5.com)> wrote:
Quote: |
On Mon, Dec 15, 2014 at 7:34 PM, Dan Cropp <dan@amtelco.com (dan@amtelco.com)> wrote:
I am not sure if I entered the correct settings for the transport information.
For the local_net, I entered my local ip address, but no mask. I will check with the network admin so he can verify the settings I entered.
You need the network and mask. For example if the ip address and mask of the test machine is 192.168.0.1/255.255.255.0 then the correct entry would be 192.168.0.0/24.
Quote: |
One minor detail, we are using ip authentication. When Vitelity changed my account from user based authentication to IP based authentication, they stopped including a user for the account.
Should these settings work without the from_user (IP based authentication) or do I need to get the account name from Vitelity?
|
You definitely need the master account login username. If you has this working with chan_sip, then try the 'fromuser' from sip.conf and user is from_user.
--
_____________________________________________________________________
-- 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
|
--
_____________________________________________________________________
-- 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 |
|
|
Back to top |
|
|
george.joseph at fairv... Guest
|
Posted: Tue Dec 16, 2014 11:11 am Post subject: [asterisk-users] PJSIP configuration question |
|
|
On Tue, Dec 16, 2014 at 9:00 AM, Dan Cropp <dan@amtelco.com (dan@amtelco.com)> wrote: Quote: |
I corrected my local_net setting (based on advice from network admin).
I have tried several different values for the from_user and still have the same problem.
Asterisk receives the OK from Vitelity.
Asterisk sends the ACK (without a Contact header).
Vitelity doesn’t seem to process it, so they send an OK again.
The OK receive, Transmit ACK occurs 4 times.
A short while later, Vitelity hangs up on my cell phone.
Asterisk is never told the call is gone.
If I hangup the call from Asterisk side,
Asterisk sends the BYE message.
Vitelity responds with a “SIP/2.0 481 Call leg/transaction does not exist”
Again, the trace indicates the ACK message is missing the Contact header.
Additional note: the network admin is asking why the local_net, external_media_address, and external_signalling_address are needed. He wrote me…“You should NOT have to know your public IP. The firewall should be doing fixup commands to change the values in the packet”
|
First...
"The firewall should be doing fixup commands to change the values in the packet”
The firewall should NOT be changing values in the packet. If it is, all bets are off.
Second.
Can you try making a call from a phone instead of from an AMI originate?
Quote: |
From: asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@lists.digium.com) [mailto:asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@lists.digium.com)] On Behalf Of George Joseph
Sent: Monday, December 15, 2014 11:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PJSIP configuration question
On Mon, Dec 15, 2014 at 9:48 PM, Dan Cropp <dan@amtelco.com (dan@amtelco.com)> wrote:
Thanks George.
I will correct my local_net in the morning.
Vitelity chan_sip settings I have working, do not have a fromuser.
sip.conf settings...
I think you can actually specify anything, it just has to be populated with something other than a sub-account username.
Quote: |
[HVout]
type=friend
dtmfmode=auto
host=64.2.142.93
disallow=all
allow=ulaw
canreinvite=no
trustrpid=yes
sendrpid=yes
nat=yes
context=TestApp
On Dec 15, 2014, at 9:32 PM, George Joseph <george.joseph@fairview5.com (george.joseph@fairview5.com)> wrote:
Quote: |
On Mon, Dec 15, 2014 at 7:34 PM, Dan Cropp <dan@amtelco.com (dan@amtelco.com)> wrote:
I am not sure if I entered the correct settings for the transport information.
For the local_net, I entered my local ip address, but no mask. I will check with the network admin so he can verify the settings I entered.
You need the network and mask. For example if the ip address and mask of the test machine is 192.168.0.1/255.255.255.0 then the correct entry would be 192.168.0.0/24.
Quote: |
One minor detail, we are using ip authentication. When Vitelity changed my account from user based authentication to IP based authentication, they stopped including a user for the account.
Should these settings work without the from_user (IP based authentication) or do I need to get the account name from Vitelity?
|
You definitely need the master account login username. If you has this working with chan_sip, then try the 'fromuser' from sip.conf and user is from_user.
|
--
_____________________________________________________________________
-- 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 |
--
_____________________________________________________________________
-- 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
|
|
|
Back to top |
|
|
jcolp at digium.com Guest
|
Posted: Tue Dec 16, 2014 11:13 am Post subject: [asterisk-users] PJSIP configuration question |
|
|
Dan Cropp wrote:
Quote: | I corrected my local_net setting (based on advice from network admin).
I have tried several different values for the from_user and still have
the same problem.
Asterisk receives the OK from Vitelity.
Asterisk sends the ACK (without a Contact header).
|
A Contact header is not required to be in the ACK.
Quote: |
Vitelity doesn’t seem to process it, so they send an OK again.
|
I'd try to isolate this further as there's two possible things:
1. The ACK never got to them
2. They didn't process it
Quote: |
The OK receive, Transmit ACK occurs 4 times.
A short while later, Vitelity hangs up on my cell phone.
Asterisk is never told the call is gone.
If I hangup the call from Asterisk side,
Asterisk sends the BYE message.
Vitelity responds with a “SIP/2.0 481 Call leg/transaction does not exist”
Again, the trace indicates the ACK message is missing the Contact header.
Additional note: the network admin is asking why the local_net,
external_media_address, and external_signalling_address are needed. He
wrote me…“You should NOT have to know your public IP. The firewall
should be doing fixup commands to change the values in the packet”
|
This can cause major problems. I've rarely (if ever) come across an ALG
(that's what that is) that didn't break something.
Cheers,
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
--
_____________________________________________________________________
-- 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 |
|
Back to top |
|
|
dan at amtelco.com Guest
|
Posted: Tue Dec 16, 2014 11:43 am Post subject: [asterisk-users] PJSIP configuration question |
|
|
Thank you George and Joshua.
"This can cause major problems. I've rarely (if ever) come across an ALG (that's what that is) that didn't break something."
I am contacting the network admin and seeing if he can modify the firewall.
I'm a lifelong programmer. Code and programming, I understand.
When it comes to the network, I'm clueless.
--
_____________________________________________________________________
-- 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 |
|
Back to top |
|
|
dan at amtelco.com Guest
|
Posted: Tue Dec 16, 2014 1:45 pm Post subject: [asterisk-users] PJSIP configuration question |
|
|
Here's an update...
My network admin would not turn off the ALG because it would cause several other problems to other phone systems we have.
He looked at the sip trace. What he found is the PJSIP trace showed a different IP address than the older chan_sip so he had me change the aor contact to outbound.vitelity.net
At this point, it seems to be working (and this is going through a Cisco ALG).
I will run more tests, but here is the pjsip.conf I have.
[global]
type = global
debug = yes
[transport1]
type = transport
bind = 0.0.0.0
protocol = udp
[outbound.vitelity.net]
type = aor
remove_existing = yes
qualify_frequency = 60
contact = sip:outbound.vitelity.net
[outbound.vitelity.net]
type = endpoint
context = TestApp
transport = transport1
aors = outbound.vitelity.net
dtmf_mode = rfc4733
force_rport = yes
rtp_symmetric = yes
rewrite_contact = yes
send_rpid = yes
trust_id_inbound = yes
disallow = all
allow = ulaw
direct_media = no
[outbound.vitelity.net]
type = identify
endpoint = outbound.vitelity.net
match = 64.2.142.93
Have a great day!
Dan
--
_____________________________________________________________________
-- 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 |
|
Back to top |
|
|
george.joseph at fairv... Guest
|
Posted: Tue Dec 16, 2014 3:39 pm Post subject: [asterisk-users] PJSIP configuration question |
|
|
On Tue, Dec 16, 2014 at 11:45 AM, Dan Cropp <dan@amtelco.com (dan@amtelco.com)> wrote: Quote: | Here's an update...
My network admin would not turn off the ALG because it would cause several other problems to other phone systems we have.
He looked at the sip trace. What he found is the PJSIP trace showed a different IP address than the older chan_sip so he had me change the aor contact to outbound.vitelity.net
At this point, it seems to be working (and this is going through a Cisco ALG).
|
Glad you got it working!
|
|
Back to top |
|
|
|
|
|
You cannot post new topics in this forum You cannot reply to topics in this forum You cannot edit your posts in this forum You cannot delete your posts in this forum You cannot vote in polls in this forum
|
Powered by phpBB © 2001, 2005 phpBB Group
|