Thank you Joshua.
I tried setting the from_domain for the endpoint, but it still sends the
internal ip address for the INVITE's From field
[acl1]
type = acl
deny = 0.0.0.0/0.0.0.0
permit = variousaddress
permit = bluipaddress
[transport1]
type = transport
bind = 0.0.0.0
protocol = udp
Dan Cropp wrote:
I am trying to configure a connection to BluIP. I am able to make
incoming calls work. However outgoing calls are not working.
For the Outbound Registration, I noticed the contact field is always the
internal IP address of my pc instead of mycompany dot com
This is fine. The
Sent: Tuesday, December 15, 2015 11:17 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PJSIP configuration question
Dan Cropp wrote:
> outbound_proxy = chi-sbc3-iad.bluip.com
Try setting this to: outbound_proxy = chi-sbc3-iad.bluip.com\;lr
--
Jos
I am trying to configure a connection to BluIP. I am able to make incoming
calls work. However outgoing calls are not working.
For the Outbound Registration, I noticed the contact field is always the
internal IP address of my pc instead of mycompany dot com
I can Originate (using AMI) to my
Dan Cropp wrote:
outbound_proxy = chi-sbc3-iad.bluip.com
Try setting this to: outbound_proxy = chi-sbc3-iad.bluip.com\;lr
--
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
--
: [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.
--
_
-- Bandwidth and Colocation Provided by http://www.api
: Re: [asterisk-users] PJSIP configuration question
On Mon, Dec 15, 2014 at 9:48 PM, Dan Cropp
d...@amtelco.commailto:d...@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
: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 d...@amtelco.com wrote:
Thanks George.
I will correct my local_net in the morning.
Vitelity chan_sip settings I
Dan Cropp wrote:
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
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.
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
On Tue, Dec 16, 2014 at 11:45 AM, Dan Cropp d...@amtelco.com wrote:
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
Today, I tried the same behavior on Asterisk 13.1.0 and Asterisk 12.2.0.
Same problem is happening with both of them.
Could this be caused by PJPROJECT 2.3?
Anyone have any suggestions for what I can try?
My boss is giving me until tomorrow to get the PJSIP support working with
Vitelity.
On Mon, Dec 15, 2014 at 2:08 PM, Dan Cropp d...@amtelco.com wrote:
Today, I tried the same behavior on Asterisk 13.1.0 and Asterisk 12.2.0.
Same problem is happening with both of them.
Could this be caused by PJPROJECT 2.3?
Anyone have any suggestions for what I can try?
My boss
: Re: [asterisk-users] PJSIP configuration question
On Mon, Dec 15, 2014 at 2:08 PM, Dan Cropp
d...@amtelco.commailto:d...@amtelco.com wrote:
Today, I tried the same behavior on Asterisk 13.1.0 and Asterisk 12.2.0.
Same problem is happening with both of them.
Could this be caused by PJPROJECT 2.3
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *George Joseph
*Sent:* Monday, December 15, 2014 3:40 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] PJSIP configuration question
-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of George Joseph
Sent: Monday, December 15, 2014 4:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PJSIP configuration question
On Mon, Dec 15, 2014 at 3:33 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] PJSIP configuration question
On Mon, Dec 15, 2014 at 3:33 PM, Dan Cropp d...@amtelco.com wrote:
Hi George,
Thank you for looking into this.
This is behind a nat…
Just to be clear...both
...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com]
On Behalf Of George Joseph
Sent: Monday, December 15, 2014 4:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PJSIP configuration question
On Mon, Dec 15, 2014 at 3:33 PM, Dan Cropp
d
Ok Dan, try this... I was able to get this to work behind a NAT and with
ip address authentication.
[global]
type = global
debug = yes
[transport1]
type = transport
bind = 0.0.0.0
protocol = udp
*local_net=yourlocalnet I.E. 10.10.10.10/24
http://10.10.10.10/24external_media_address=your
Thanks George.
I will remote into and give this a try.
Have a great evening!
Dan
On Dec 15, 2014, at 7:27 PM, George Joseph
george.jos...@fairview5.commailto:george.jos...@fairview5.com wrote:
Ok Dan, try this... I was able to get this to work behind a NAT and with ip
address
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of George Joseph
Sent: Monday, December 15, 2014 7:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PJSIP configuration question
Ok Dan, try this... I was able to get
-Commercial Discussion
*Subject:* Re: [asterisk-users] PJSIP configuration question
Ok Dan, try this... I was able to get this to work behind a NAT and with
ip address authentication.
[global]
type = global
debug = yes
[transport1]
type = transport
bind = 0.0.0.0
protocol = udp
-users] PJSIP configuration question
Ok Dan, try this... I was able to get this to work behind a NAT and with ip
address authentication.
[global]
type = global
debug = yes
[transport1]
type = transport
bind = 0.0.0.0
protocol = udp
local_net=yourlocalnet I.E. 10.10.10.10/24http://10.10.10.10/24
-Commercial Discussion
*Subject:* Re: [asterisk-users] PJSIP configuration question
Ok Dan, try this... I was able to get this to work behind a NAT and with
ip address authentication.
[global]
type = global
debug = yes
[transport1]
type = transport
bind = 0.0.0.0
protocol = udp
Trying this again after my first away from work in a couple weeks.
Running Asterisk 13.0.0
IP authentication with Vitelity
I can Originate with sip, but not pjsip.
Here is the sip settings and trace.
Action: Originate
ActionID: S8
Channel: SIP/800...@outbound.vitelity.net
Exten: createcall
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PJSIP configuration question
snip
I translated those settings to the following for pjsip.conf...
[transport1]
type = transport
bind = 0.0.0.0
protocol = udp
[outbound.vitelity.net]
type = aor
Thanks George.
I am NATed.
I did not obfuscate the 0.0.19.196. That is really what is showing up.
The only portion that I hid is the IP address of my box.
Have a great day!
Dan
On Wed, Dec 10, 2014 at 2:03 PM, Dan Cropp
d...@amtelco.commailto:d...@amtelco.com wrote:
Thanks George.
That was
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Cropp
Sent: Thursday, December 11, 2014 8:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PJSIP configuration question
Thank you Joshua.
I will make the modifications
: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Cropp
Sent: Thursday, December 11, 2014 8:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PJSIP configuration question
Thank you Joshua.
I
Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PJSIP configuration question
Ok, it didn't quite solve everything.
There is one slight issue. When I answer the call on my cell phone, Asterisk
sees it as answered.
I can play audio, send dtmfs, etc and hear it on my phone
Dan Cropp wrote:
I had my screenshots flipped. Is there a way to make sure the Contact field is
NOT included in the ACK response to the OK (for the Answer)?
PJSIP is including the Contact for the ACK response to the OK.
Contact:sip:1...@xxx.xxx.xx.xxx:5060
There is no configuration option
I am not sure what you mean by the ful SIP signaling?
Here is the trace for the sip.conf which works successfully.
Below that, I will include the trace for the pjsip.conf which it seems Vitelity
isn't accepting the ACK in response to the OK
SIP ---
--- Transmitting SIP request (1004
, December 11, 2014 3:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PJSIP configuration question
I am not sure what you mean by the ful SIP signaling?
Here is the trace for the sip.conf which works successfully.
Below that, I will include the trace
I'm working with a SIP provider to try and transition our sip connection with
them to PJSIP. I thought I had transitioned the settings correctly, but
whenever I attempt an Originate it never even tries to send any PJSIP messages.
I'm currently running Asterisk 13.0.0.
Anyone have any
Kia ora,
Dan Cropp wrote:
I’m working with a SIP provider to try and transition our sip connection
with them to PJSIP. I thought I had transitioned the settings correctly,
but whenever I attempt an Originate it never even tries to send any
PJSIP messages.
What dial string are you providing to
Subject: Re: [asterisk-users] PJSIP configuration question
Kia ora,
Dan Cropp wrote:
I'm working with a SIP provider to try and transition our sip
connection with them to PJSIP. I thought I had transitioned the
settings correctly, but whenever I attempt an Originate it never even
tries to send
: [asterisk-users] PJSIP configuration question
Thank you for the speedy reply.
My originate string is something like the following where x is really the
sip provider's supplied IP address
1234567890 is really the phone number I am dialing
PJSIP/outbound.vitelity.net/1234567890
In the chan_sip
Not sure why, but Vitelity changed the settings to IP based authentication on
me. Here's the new sip.conf settings they sent me.
type=friend
dtmfmode=auto
host=64.2.142.93
allow=all
nat=yes
canreinvite=no
trustrpid=yes
sendrpid=yes
When I use these settings to originate calls using the
On Wed, Dec 10, 2014 at 1:27 PM, Dan Cropp d...@amtelco.com wrote:
Not sure why, but Vitelity changed the settings to IP based authentication
on me. Here's the new sip.conf settings they sent me.
type=friend
dtmfmode=auto
host=64.2.142.93
allow=all
nat=yes
canreinvite=no
trustrpid=yes
] On Behalf Of George Joseph
Sent: Wednesday, December 10, 2014 2:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PJSIP configuration question
On Wed, Dec 10, 2014 at 1:27 PM, Dan Cropp
d...@amtelco.commailto:d...@amtelco.com wrote:
Not sure why
On Wed, Dec 10, 2014 at 2:03 PM, Dan Cropp d...@amtelco.com wrote:
Thanks George.
That was the ip address I was given. Unfortunately, my contact at
Vitelity is gone for the day so I can’t verify it with him.
I added the qualify_frequency as you suggested and it does appear that I
have
snip
I translated those settings to the following for pjsip.conf...
[transport1]
type = transport
bind = 0.0.0.0
protocol = udp
[outbound.vitelity.net]
type = aor
remove_existing = yes
contact = sip:64.2.142.93@5060
This is incorrect. The contact should be:
contact = sip:64.2.142.93
It
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