Re: [asterisk-users] PJSIP configuration question

[Dan Cropp]

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

[BLUIPIN]
type = aor
remove_existing = yes
contact = sip:bluipaddress 

[auth7]
type = auth
username = didassignedbybluip
password = password

[didassignedbybluip]
type = endpoint
context = TestApp
transport = transport1
outbound_auth = auth7
aors = BLUIPIN
accountcode = 16
dtmf_mode = rfc4733
device_state_busy_at = 2
force_rport = no
rtp_symmetric = no
rewrite_contact = no
identify_by = username
outbound_proxy = chi-sbc3-iad.bluip.com
trust_id_outbound = yes
from_domain = amtelco.com
disallow = all
allow = ulaw

[identify72]
type = identify
endpoint = didassignedbybluip
match = bluipaddress

[registration2]
type = registration
transport = transport1
client_uri = sip:didassignedbybluip at companyname dot com
server_uri = sip:chi-sbc3-iad.bluip.com
contact_user = sip:didassignedbybluip at companyname dot com
outbound_auth = auth7

Here is a trace of another system where it works.

IINVITE sip:callphonenumber@bluipaddress:5060 SIP/2.0
Via: SIP/2.0/UDP ipaddress:5060;branch=z9hG4bKQFD28QYY8412c000
To: 
From: ;tag=JEDtnWPm
Contact: 
Alert-Info: Classic-1
Call-ID: kD5sJntD5-0001-@ipaddress
CSeq: 1808 INVITE
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,OPTIONS,REFER,NOTIFY

100 Trying
Via: SIP/2.0/UDP ipaddress:5060;branch=z9hG4bKQFD28QYY8412c000
To: 
From: ;tag=JEDtnWPm
Call-ID: kD5sJntD5-0001-@ ipaddress
CSeq: 1808 INVITE

401 Unauthorized
Via: SIP/2.0/UDP ipaddress:5060;branch=z9hG4bKQFD28QYY8412c000
To: ;tag=1164309609-1450128136967
From: ;tag=JEDtnWPm
Call-ID: kD5sJntD5-0001-@ ipaddress
CSeq: 1808 INVITE
WWW-Authenticate: DIGEST 
qop="auth",nonce="BroadWorksXii6gunlzTrqhij5BW",realm="BroadWorks",algorithm=MD5
Content-Length: 0

ACK sip:callphonenumber@bluipaddress:5060 SIP/2.0
Via: SIP/2.0/UDP ipaddress:5060;branch=z9hG4bKQFD28QYY8412c000
To: ;tag=1164309609-1450128136967
From: ;tag=JEDtnWPm
Max-Forwards: 70
Call-ID: kD5sJntD5-0001-@ ipaddress
CSeq: 1808 ACK

INVITE sip:callphonenumber@bluipaddress:5060 SIP/2.0
Via: SIP/2.0/UDP ipaddress:5060;branch=z9hG4bKYcoOR2md8413
To: 
From: ;tag=pcjBsnMZ
Contact: 
Alert-Info: Classic-1
Call-ID: kD5sJntD5-0001-@ ipaddress
CSeq: 1809 INVITE
Authorization: Digest username=" didassignedbybluip 
",realm="BroadWorks",nonce="BroadWorksXii6gunlzTrqhij5BW",uri="sip:callphonenumber@bluipaddress:5060",response="6d3f8dea20a6173a9cbde2ce912696b7",algorithm=MD5
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,OPTIONS,REFER,NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 335

100 Trying
Via: SIP/2.0/UDP ipaddress:5060;branch=z9hG4bKYcoOR2md8413
To: 
From: ;tag=pcjBsnMZ
Call-ID: kD5sJntD5-0001-@ipaddress
CSeq: 1809 INVITE

180 Ringing
Via: SIP/2.0/UDP ipaddress:5060;branch=z9hG4bKYcoOR2md8413
To: ;tag=878577519-1450128140899
From: ;tag=pcjBsnMZ
Call-ID: kD5sJntD5-0001-@ipaddress
CSeq: 1809 INVITE
Supported: 
Contact: 
P-Asserted-Identity: 
Privacy: none
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 223

...

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Re: [asterisk-users] PJSIP configuration question

[Joshua Colp]

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 Contact tells the server where to send incoming 
INVITEs in the case of a REGISTER. If it wasn't your address then the 
server wouldn't really know where to send incoming calls. If you need it 
to be a public IP address then NAT settings can be set on the transport.






I can Originate (using AMI) to my Vitelity trunk (IP based authentication).

However, when I Originate to my BluIP, it is being rejected.

When I compare this with another system that does the same behavior, the
one difference I notice is that the other system does not pass the
internal ip address for the From domain portion. Instead, it is passing
companyname.com.


The domain in the From header can be set using from_domain on the endpoint.

It would also be useful to provide the configuration, minus passwords, 
and an example of a working SIP INVITE from another machine...


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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

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Re: [asterisk-users] PJSIP configuration question

[Dan Cropp]

Thanks Joshua.

Commenting out the outbound_proxy line did fixed my problem.
Both inbound and outbound calls are now working.

Have a great day!
Dan

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp
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

--
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

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[asterisk-users] PJSIP configuration question

[Dan Cropp]

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 Vitelity trunk (IP based authentication).
However, when I Originate to my BluIP, it is being rejected.

When I compare this with another system that does the same behavior, the one 
difference I notice is that the other system does not pass the internal ip 
address for the From domain portion.  Instead, it is passing companyname.com.

For the outbound Registration, I noticed the REGISTER Contact field is always 
my internal ip address.  I programmed the contact for the registration portion 
of pjsip.conf to attempt to force the companyname instead of the internal ip 
address.  However, this didn't change the REGISTER packet.
Is there a different setting that I need to enter to force the REGISTER to use 
my companyname dot com instead of the ip address?

contact = sip:didassignedbybluip at companyname.com

The REGISTER succeeds (which explains internal calls working).  However, they 
are always using my internal ip address.

BluIP
Xmt
INVITE sip:18005551212 @bluipaddress SIP/2.0
Via: SIP/2.0/UDP 
myinternalipaddress:5060;rport;branch=z9hG4bKPj32298b5f-ed09-4de4-9649-da4c4b78fafb
From: "Name" ;tag=4fda8a53-8831-45bd-9b29-15eb276ceafb
To: 
Contact: 

Call-ID: ebc6049a-d700-43a4-b230-b06cd6289eea
CSeq: 12733 INVITE
...

Rcv
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 
myinternalipaddress:5060;received=myinternalipaddress;branch=z9hG4bKPj32298b5f-ed09-4de4-9649-da4c4b78fafb;rport=5060
From: "Name" ;tag=4fda8a53-8831-45bd-9b29-15eb276ceafb
To: 
Call-ID: ebc6049a-d700-43a4-b230-b06cd6289eea
CSeq: 12733 INVITE


Rcv
SIP/2.0 604 Does not exist anywhere
Via: SIP/2.0/UDP 
myinternalipaddress:5060;received=myinternalipaddress;branch=z9hG4bKPj32298b5f-ed09-4de4-9649-da4c4b78fafb;rport=5060
From: "Name" ;tag=4fda8a53-8831-45bd-9b29-15eb276ceafb
To: ;tag=1389319045-1450195105789
Call-ID: ebc6049a-d700-43a4-b230-b06cd6289eea
CSeq: 12733 INVITE
Content-Length: 0

Any suggestions of what I am doing wrong?

Have a great day!
Dan
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Re: [asterisk-users] PJSIP configuration question

[Joshua Colp]

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

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Re: [asterisk-users] PJSIP configuration question

[Dan Cropp]

Thanks George.
I will give it a try.

Have a great day!
Dan


From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@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.
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Re: [asterisk-users] PJSIP configuration question

[Dan Cropp]

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-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@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 
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 you can actually specify anything, it just has to be populated with 
something other than a sub-account username.


[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.jos...@fairview5.commailto:george.jos...@fairview5.com wrote:


On Mon, Dec 15, 2014 at 7:34 PM, Dan Cropp 
d...@amtelco.commailto:d...@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 192.168.0.1/255.255.255.0http://192.168.0.1/255.255.255.0 
then the correct entry would be 192.168.0.0/24http://192.168.0.0/24.

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.




Have a great day!

Da

From: 
asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com
 
[mailto:asterisk-users-boun...@lists.digium.commailto: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 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
external_media_address=your public ip address
external_signaling_address=your public address

[outbound.vitelity.nethttp://outbound.vitelity.net]
type = aor
remove_existing = yes
qualify_frequency = 60
contact = sip:64.2.142.93
[outbound.vitelity.nethttp://outbound.vitelity.net]
type = endpoint
context = TestApp
transport = transport1
aors = outbound.vitelity.nethttp://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
from_user=your main vitelity account name  ; Not subaccount
[outbound.vitelity.nethttp://outbound.vitelity.net]
type = identify
endpoint = outbound.vitelity.nethttp://outbound.vitelity.net
match = 64.2.142.93

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Re: [asterisk-users] PJSIP configuration question

[George Joseph]

On Tue, Dec 16, 2014 at 9:00 AM, Dan Cropp d...@amtelco.com 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).

 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?






 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@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 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 you can actually specify anything, it just has to be populated
 with something other than a sub-account username.





 [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.jos...@fairview5.com
 wrote:





 On Mon, Dec 15, 2014 at 7:34 PM, Dan Cropp d...@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.



 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.









 Have a great day!



 Da



 *From:* asterisk-users-boun...@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 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 public ip
 addressexternal_signaling_address=your public address*
 [outbound.vitelity.net]
 type = aor
 remove_existing = yes
 qualify_frequency = 60
 contact = sip:64.2.142.93

 [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

 *from_user=your main vitelity account name  ; Not subaccount*

 [outbound.vitelity.net]
 type = identify
 endpoint = outbound.vitelity.net
 match = 64.2.142.93


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Re: [asterisk-users] PJSIP configuration question

[Joshua Colp]

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 required to be in the ACK.



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



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

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Re: [asterisk-users] PJSIP configuration question

[Dan Cropp]

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.
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Re: [asterisk-users] PJSIP configuration question

[Dan Cropp]

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

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Re: [asterisk-users] PJSIP configuration question

[George Joseph]

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 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!


 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

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Re: [asterisk-users] PJSIP configuration question

[Dan Cropp]

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.  Otherwise, he's told me to go back to using chan_sip and wait a year 
or two for PJSIP to be in the field more.

Have a great day!

Dan
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Re: [asterisk-users] PJSIP configuration question

[George Joseph]

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 is giving me until tomorrow to get the PJSIP support working with
 Vitelity.  Otherwise, he’s told me to go back to using chan_sip and wait a
 year or two for PJSIP to be in the field more.


I have a Vitelity account I can try.  Re-post your pjsip config and I'll
try it now.





 Have a great day!



 Dan

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Re: [asterisk-users] PJSIP configuration question

[Dan Cropp]

Hi George,

Thank you for looking into this.
This is behind a nat…

[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:64.2.142.93

[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

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

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?

Anyone have any suggestions for what I can try?

My boss is giving me until tomorrow to get the PJSIP support working with 
Vitelity.  Otherwise, he’s told me to go back to using chan_sip and wait a year 
or two for PJSIP to be in the field more.

I have a Vitelity account I can try.  Re-post your pjsip config and I'll try it 
now.



Have a great day!

Dan

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Re: [asterisk-users] PJSIP configuration question

[George Joseph]

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 the pbx and local endpoints are behind the same NAT?



 [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:64.2.142.93



 [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



 *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



 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 is giving me until tomorrow to get the PJSIP support working with
 Vitelity.  Otherwise, he’s told me to go back to using chan_sip and wait a
 year or two for PJSIP to be in the field more.



 I have a Vitelity account I can try.  Re-post your pjsip config and I'll
 try it now.







 Have a great day!



 Dan


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Re: [asterisk-users] PJSIP configuration question

[Dan Cropp]

Yes, everything is behind the same NAT.

For the application I’m working on, the only endpoint is the endpoint to 
Vitelity.
We use AMI to Originate calls from Asterisk endpoint through Vitelity to phones.
After that, we control the call through AMI to perform the work we need.



From: asterisk-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, Dan Cropp 
d...@amtelco.commailto:d...@amtelco.com wrote:
Hi George,

Thank you for looking into this.
This is behind a nat…


Just to be clear...both the pbx and local endpoints are behind the same NAT?


[global]
type = global
debug = yes

[transport1]
type = transport
bind = 0.0.0.0
protocol = udp

[outbound.vitelity.nethttp://outbound.vitelity.net]
type = aor
remove_existing = yes
qualify_frequency = 60
contact = sip:64.2.142.93

[outbound.vitelity.nethttp://outbound.vitelity.net]
type = endpoint
context = TestApp
transport = transport1
aors = outbound.vitelity.nethttp://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.nethttp://outbound.vitelity.net]
type = identify
endpoint = outbound.vitelity.nethttp://outbound.vitelity.net
match = 64.2.142.93

Have a great day!

Dan

From: 
asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com
 
[mailto:asterisk-users-boun...@lists.digium.commailto: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

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?

Anyone have any suggestions for what I can try?

My boss is giving me until tomorrow to get the PJSIP support working with 
Vitelity.  Otherwise, he’s told me to go back to using chan_sip and wait a year 
or two for PJSIP to be in the field more.

I have a Vitelity account I can try.  Re-post your pjsip config and I'll try it 
now.



Have a great day!

Dan

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Re: [asterisk-users] PJSIP configuration question

[George Joseph]

On Mon, Dec 15, 2014 at 3:54 PM, Dan Cropp d...@amtelco.com wrote:

 Yes, everything is behind the same NAT.



 For the application I’m working on, the only endpoint is the endpoint to
 Vitelity.

 We use AMI to Originate calls from Asterisk endpoint through Vitelity to
 phones.

 After that, we control the call through AMI to perform the work we need.




And it's outbound calls that aren't working right?






 *From:* asterisk-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, Dan Cropp d...@amtelco.com wrote:

 Hi George,



 Thank you for looking into this.

 This is behind a nat…





 Just to be clear...both the pbx and local endpoints are behind the same
 NAT?





 [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:64.2.142.93



 [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



 *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



 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 is giving me until tomorrow to get the PJSIP support working with
 Vitelity.  Otherwise, he’s told me to go back to using chan_sip and wait a
 year or two for PJSIP to be in the field more.



 I have a Vitelity account I can try.  Re-post your pjsip config and I'll
 try it now.







 Have a great day!



 Dan


 --
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Re: [asterisk-users] PJSIP configuration question

[Dan Cropp]

Yes, outbound calls are the only ones I’m trying.


From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of George Joseph
Sent: Monday, December 15, 2014 4:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PJSIP configuration question



On Mon, Dec 15, 2014 at 3:54 PM, Dan Cropp 
d...@amtelco.commailto:d...@amtelco.com wrote:
Yes, everything is behind the same NAT.

For the application I’m working on, the only endpoint is the endpoint to 
Vitelity.
We use AMI to Originate calls from Asterisk endpoint through Vitelity to phones.
After that, we control the call through AMI to perform the work we need.


And it's outbound calls that aren't working right?



From: 
asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com
 
[mailto:asterisk-users-boun...@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...@amtelco.commailto:d...@amtelco.com wrote:
Hi George,

Thank you for looking into this.
This is behind a nat…


Just to be clear...both the pbx and local endpoints are behind the same NAT?


[global]
type = global
debug = yes

[transport1]
type = transport
bind = 0.0.0.0
protocol = udp

[outbound.vitelity.nethttp://outbound.vitelity.net]
type = aor
remove_existing = yes
qualify_frequency = 60
contact = sip:64.2.142.93

[outbound.vitelity.nethttp://outbound.vitelity.net]
type = endpoint
context = TestApp
transport = transport1
aors = outbound.vitelity.nethttp://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.nethttp://outbound.vitelity.net]
type = identify
endpoint = outbound.vitelity.nethttp://outbound.vitelity.net
match = 64.2.142.93

Have a great day!

Dan

From: 
asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com
 
[mailto:asterisk-users-boun...@lists.digium.commailto: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

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?

Anyone have any suggestions for what I can try?

My boss is giving me until tomorrow to get the PJSIP support working with 
Vitelity.  Otherwise, he’s told me to go back to using chan_sip and wait a year 
or two for PJSIP to be in the field more.

I have a Vitelity account I can try.  Re-post your pjsip config and I'll try it 
now.



Have a great day!

Dan

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Re: [asterisk-users] PJSIP configuration question

[George Joseph]

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 public ip
addressexternal_signaling_address=your public address*
[outbound.vitelity.net]
type = aor
remove_existing = yes
qualify_frequency = 60
contact = sip:64.2.142.93

[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

*from_user=your main vitelity account name  ; Not subaccount*
[outbound.vitelity.net]
type = identify
endpoint = outbound.vitelity.net
match = 64.2.142.93
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Re: [asterisk-users] PJSIP configuration question

[Dan Cropp]

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 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
external_media_address=your public ip address
external_signaling_address=your public address

[outbound.vitelity.nethttp://outbound.vitelity.net]
type = aor
remove_existing = yes
qualify_frequency = 60
contact = sip:64.2.142.93

[outbound.vitelity.nethttp://outbound.vitelity.net]
type = endpoint
context = TestApp
transport = transport1
aors = outbound.vitelity.nethttp://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
from_user=your main vitelity account name  ; Not subaccount

[outbound.vitelity.nethttp://outbound.vitelity.net]
type = identify
endpoint = outbound.vitelity.nethttp://outbound.vitelity.net
match = 64.2.142.93

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Re: [asterisk-users] PJSIP configuration question

[Dan Cropp]

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.

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?

Have a great day!

Da

From: asterisk-users-boun...@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 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
external_media_address=your public ip address
external_signaling_address=your public address

[outbound.vitelity.nethttp://outbound.vitelity.net]
type = aor
remove_existing = yes
qualify_frequency = 60
contact = sip:64.2.142.93
[outbound.vitelity.nethttp://outbound.vitelity.net]
type = endpoint
context = TestApp
transport = transport1
aors = outbound.vitelity.nethttp://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
from_user=your main vitelity account name  ; Not subaccount
[outbound.vitelity.nethttp://outbound.vitelity.net]
type = identify
endpoint = outbound.vitelity.nethttp://outbound.vitelity.net
match = 64.2.142.93
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Re: [asterisk-users] PJSIP configuration question

[George Joseph]

On Mon, Dec 15, 2014 at 7:34 PM, Dan Cropp d...@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.


 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.






 Have a great day!



 Da



 *From:* asterisk-users-boun...@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 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 public ip
 addressexternal_signaling_address=your public address*
 [outbound.vitelity.net]
 type = aor
 remove_existing = yes
 qualify_frequency = 60
 contact = sip:64.2.142.93

 [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

 *from_user=your main vitelity account name  ; Not subaccount*

 [outbound.vitelity.net]
 type = identify
 endpoint = outbound.vitelity.net
 match = 64.2.142.93

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Re: [asterisk-users] PJSIP configuration question

[Dan Cropp]

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.jos...@fairview5.commailto:george.jos...@fairview5.com wrote:



On Mon, Dec 15, 2014 at 7:34 PM, Dan Cropp 
d...@amtelco.commailto:d...@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 192.168.0.1/255.255.255.0http://192.168.0.1/255.255.255.0 
then the correct entry would be 192.168.0.0/24http://192.168.0.0/24.

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.




Have a great day!

Da

From: 
asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com
 
[mailto:asterisk-users-boun...@lists.digium.commailto: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 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
external_media_address=your public ip address
external_signaling_address=your public address

[outbound.vitelity.nethttp://outbound.vitelity.net]
type = aor
remove_existing = yes
qualify_frequency = 60
contact = sip:64.2.142.93
[outbound.vitelity.nethttp://outbound.vitelity.net]
type = endpoint
context = TestApp
transport = transport1
aors = outbound.vitelity.nethttp://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
from_user=your main vitelity account name  ; Not subaccount
[outbound.vitelity.nethttp://outbound.vitelity.net]
type = identify
endpoint = outbound.vitelity.nethttp://outbound.vitelity.net
match = 64.2.142.93

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Re: [asterisk-users] PJSIP configuration question

[George Joseph]

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 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.



 [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.jos...@fairview5.com
 wrote:



 On Mon, Dec 15, 2014 at 7:34 PM, Dan Cropp d...@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.


 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.






 Have a great day!



 Da



 *From:* asterisk-users-boun...@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 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 public ip
 addressexternal_signaling_address=your public address*
 [outbound.vitelity.net]
 type = aor
 remove_existing = yes
 qualify_frequency = 60
 contact = sip:64.2.142.93

 [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

 *from_user=your main vitelity account name  ; Not subaccount*

 [outbound.vitelity.net]
 type = identify
 endpoint = outbound.vitelity.net
 match = 64.2.142.93

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[asterisk-users] PJSIP configuration question

[Dan Cropp]

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
Context: TestApp
Priority: 1
Timeout: 6
CallerID: John Doe 1234
Variable: CALLERID(num-pres)=allowed_passed_screen
Async: true

sip.conf
[HVout]
type=friend
dtmfmode=auto
host=64.2.142.93
disallow=all
allow=ulaw
canreinvite=no
trustrpid=yes
sendrpid=yes
nat=yes
context=TestApp


== Using SIP RTP CoS mark 5
Audio is at 18226
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 64.2.142.189:5060:
INVITE sip:800...@outbound.vitelity.net SIP/2.0
Via: SIP/2.0/UDP 192.168.11.166:5060;branch=z9hG4bK40275183
Max-Forwards: 70
From: John Doe sip:1234@192.168.11.166;tag=as466267de
To: sip:800...@outbound.vitelity.net
Contact: sip:1234@192.168.11.166:5060
Call-ID: 59e9eff8339e32af271c23541298135d@192.168.11.166:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.0.0
Date: Sun, 21 Dec 2014 20:06:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, 
PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 1422632184 1422632184 IN IP4 192.168.11.166
s=Asterisk PBX 13.0.0
c=IN IP4 192.168.11.166
t=0 0
m=audio 18226 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

---
-- Called 800...@outbound.vitelity.net

--- SIP read from UDP:64.2.142.189:5060 ---
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 
192.168.11.166:5060;branch=z9hG4bK40275183;received=192.168.11.166
From: John Doe sip:1234@192.168.11.166;tag=as466267de
To: sip:800...@outbound.vitelity.net
Call-ID: 59e9eff8339e32af271c23541298135d@192.168.11.166:5060
CSeq: 102 INVITE
User-Agent: packetrino
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: sip:800555@64.2.142.189
Content-Length: 0

-
--- (11 headers 0 lines) ---

--- SIP read from UDP:64.2.142.189:5060 ---
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 
192.168.11.166:5060;branch=z9hG4bK40275183;received=192.168.11.166
From: John Doe sip:1234@192.168.11.166;tag=as466267de
To: sip:800...@outbound.vitelity.net;tag=as5458ca04
Call-ID: 59e9eff8339e32af271c23541298135d@192.168.11.166:5060
CSeq: 102 INVITE
User-Agent: packetrino
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: sip:800555@64.2.142.189
Content-Type: application/sdp
Content-Length: 287

v=0
o=root 21997 21997 IN IP4 64.2.142.189
s=session
c=IN IP4 64.2.142.189
t=0 0
m=audio 19282 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
-
--- (12 headers 14 lines) ---
sip_route_dump: route/path hop: sip:800555@64.2.142.189
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format GSM for ID 3
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|gsm|h263), peer - 
audio=(ulaw|gsm|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|gsm)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 
(telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 64.2.142.189:19282
-- SIP/outbound.vitelity.net- is making progress
0x483cdb0 -- Probation passed - setting RTP source address to 
64.2.142.189:19282

--- SIP read from UDP:64.2.142.189:5060 ---
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
192.168.11.166:5060;branch=z9hG4bK40275183;received=192.168.11.166
From: John Doe sip:1234@192.168.11.166;tag=as466267de
To: sip:800...@outbound.vitelity.net;tag=as5458ca04
Call-ID: 59e9eff8339e32af271c23541298135d@192.168.11.166:5060
CSeq: 102 INVITE
User-Agent: packetrino
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: sip:800555@64.2.142.189
Content-Type: application/sdp
Content-Length: 287

v=0
o=root 21997 21998 IN IP4 64.2.142.189
s=session
c=IN IP4 64.2.142.189
t=0 0
m=audio 19282 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
-
--- (12 headers 14 lines) ---
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description 

Re: [asterisk-users] PJSIP configuration question

[Dan Cropp]

Thank you Joshua.

I will make the modifications this morning and give it a try.

Have a great day!

Dan



-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp
Sent: Wednesday, December 10, 2014 7:27 PM
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
 remove_existing = yes
 contact = sip:64.2.142.93@5060

This is incorrect. The contact should be:

contact = sip:64.2.142.93

It will use a default port of 5060.

I also believe I've covered your origination issue in a separate email. 
Your dial string should be:

PJSIP/800...@outbound.vitelity.net

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

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Re: [asterisk-users] PJSIP configuration question

[Dan Cropp]

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 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 
something configured incorrectly….

--- Transmitting SIP request (483 bytes) to 
UDP:0.0.19.196:5060http://0.0.19.196:5060 ---

Well, THAT's not right.  Did you obfuscate the 0.0.19.196 or is that how it 
really is?  Are you NATed?


OPTIONS sip:64.2.142.93@5060 SIP/2.0
Via: SIP/2.0/UDP 
xxx.xxx.xx.xxx:5060;rport;branch=z9hG4bKPjcea63914-b8d1-483d-96db-11968abab704
From: 
sip:e31d5809-f26a-4219-8365-709314280...@xxx.xxx.xx.xxx;tag=7cfab3ba-73de-4243-9967-d1e6a5e7b0b4
To: sip:64.2.142.93@5060
Contact: sip:e31d5809-f26a-4219-8365-709314280...@xxx.xxx.xx.xxx:5060
Call-ID: 7ba766bf-363b-47d0-a388-62a58d1df88d
CSeq: 33778 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 13.0.0
Content-Length:  0


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Re: [asterisk-users] PJSIP configuration question

[Dan Cropp]

This fixed the problem.

Developer before me wrote some code to build up the dial string.
Always thought that string appeared off, but it worked so I left it alone.

Thanks Joshua and George for helping with this.

Have a great day!

Dan

-Original Message-
From: 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 will make the modifications this morning and give it a try.

Have a great day!

Dan



-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp
Sent: Wednesday, December 10, 2014 7:27 PM
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
 remove_existing = yes
 contact = sip:64.2.142.93@5060

This is incorrect. The contact should be:

contact = sip:64.2.142.93

It will use a default port of 5060.

I also believe I've covered your origination issue in a separate email. 
Your dial string should be:

PJSIP/800...@outbound.vitelity.net

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

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To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

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To UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] PJSIP configuration question

[Dan Cropp]

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.
However, a short while later, Vitelity tears down that call and Asterisk is 
never notified about it.

I tell Asterisk to hang up the call (via AMI) and it is removed from Asterisk.

I gather the pjsip trace.  Then, I shut down that VM, fired up another running 
chan_sip.  Did the same behavior and gathered the sip trace.
Using chan_sip, the call worked flawlessly.


Vitelity sends Asterisk the ACK (for the answer).
Asterisk send an ACK in response.  For the sip.conf system, the ACK includes 
the Contact for the response.  For PJSIP, the Contact field is not in the ACK

Is there a setting to indicate whether the Contact field should be sent as part 
of the ACK (response to the OK)?

Have a great day!
Dan


-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Cropp
Sent: Thursday, December 11, 2014 9:58 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PJSIP configuration question

This fixed the problem.

Developer before me wrote some code to build up the dial string.
Always thought that string appeared off, but it worked so I left it alone.

Thanks Joshua and George for helping with this.

Have a great day!

Dan

-Original Message-
From: 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 will make the modifications this morning and give it a try.

Have a great day!

Dan



-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp
Sent: Wednesday, December 10, 2014 7:27 PM
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
 remove_existing = yes
 contact = sip:64.2.142.93@5060

This is incorrect. The contact should be:

contact = sip:64.2.142.93

It will use a default port of 5060.

I also believe I've covered your origination issue in a separate email. 
Your dial string should be:

PJSIP/800...@outbound.vitelity.net

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

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To UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] PJSIP configuration question

[Dan Cropp]

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

When using the chan_sip, it does not include that field in the ACK response to 
the OK.

(Been a long couple weeks)

Have a great day!

Dan

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Cropp
Sent: Thursday, December 11, 2014 2:24 PM
To: Asterisk Users 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.
However, a short while later, Vitelity tears down that call and Asterisk is 
never notified about it.

I tell Asterisk to hang up the call (via AMI) and it is removed from Asterisk.

I gather the pjsip trace.  Then, I shut down that VM, fired up another running 
chan_sip.  Did the same behavior and gathered the sip trace.
Using chan_sip, the call worked flawlessly.


Vitelity sends Asterisk the ACK (for the answer).
Asterisk send an ACK in response.  For the sip.conf system, the ACK includes 
the Contact for the response.  For PJSIP, the Contact field is not in the ACK

Is there a setting to indicate whether the Contact field should be sent as part 
of the ACK (response to the OK)?

Have a great day!
Dan


-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Cropp
Sent: Thursday, December 11, 2014 9:58 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PJSIP configuration question

This fixed the problem.

Developer before me wrote some code to build up the dial string.
Always thought that string appeared off, but it worked so I left it alone.

Thanks Joshua and George for helping with this.

Have a great day!

Dan

-Original Message-
From: 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 will make the modifications this morning and give it a try.

Have a great day!

Dan



-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp
Sent: Wednesday, December 10, 2014 7:27 PM
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
 remove_existing = yes
 contact = sip:64.2.142.93@5060

This is incorrect. The contact should be:

contact = sip:64.2.142.93

It will use a default port of 5060.

I also believe I've covered your origination issue in a separate email. 
Your dial string should be:

PJSIP/800...@outbound.vitelity.net

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

--
_
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To UNSUBSCRIBE

Re: [asterisk-users] PJSIP configuration question

[Joshua Colp]

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 to configure this behavior. What is the 
full SIP signaling?


--
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

Re: [asterisk-users] PJSIP configuration question

[Dan Cropp]

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 bytes) to UDP:64.2.142.93:5060 ---
INVITE sip:800555@64.2.142.93 SIP/2.0
Via: SIP/2.0/UDP 
XXX.XXX.XXX.XXX:5060;rport;branch=z9hG4bKPj61911226-4812-44da-b682-7ec02e8c821a
From: Dan sip:2...@xxx.xxx.xxx.xxx;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8
To: sip:800555@64.2.142.93
Contact: sip:15062fef-986e-4fcf-a93e-06b28da02...@xxx.xxx.xxx.xxx:5060
Call-ID: 309ec892-56a8-46d2-95ba-a6b1d65d0c43
CSeq: 23191 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, 
PRACK, REFER, REGISTER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Remote-Party-ID: Dan sip:2...@xxx.xxx.xxx.xxx;privacy=off;screen=no
Max-Forwards: 70
User-Agent: Asterisk PBX 13.0.0
Content-Type: application/sdp
Content-Length:   239

v=0
o=- 540555224 540555224 IN IP4 XXX.XXX.XXX.XXX
s=Asterisk
c=IN IP4 XXX.XXX.XXX.XXX
t=0 0
m=audio 10030 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

--- Received SIP response (378 bytes) from UDP:64.2.142.93:5060 ---
SIP/2.0 100 Giving a try
Via: SIP/2.0/UDP 
XXX.XXX.XXX.XXX:5060;rport=5060;branch=z9hG4bKPj61911226-4812-44da-b682-7ec02e8c821a
From: Dan sip:2...@xxx.xxx.xxx.xxx;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8
To: sip:800555@64.2.142.93
Call-ID: 309ec892-56a8-46d2-95ba-a6b1d65d0c43
CSeq: 23191 INVITE
Server: OpenSIPS (1.6.4-2-notls (x86_64/linux))
Content-Length: 0





--- Received SIP response (844 bytes) from UDP:64.2.142.93:5060 ---
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 
XXX.XXX.XXX.XXX:5060;received=XXX.XXX.XXX.XXX;rport=5060;branch=z9hG4bKPj61911226-4812-44da-b682-7ec02e8c821a
Record-Route: sip:64.2.142.93;lr=on
From: Dan sip:2...@xxx.xxx.xxx.xxx;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8
To: sip:800555@64.2.142.93;tag=as7aad862a
Call-ID: 309ec892-56a8-46d2-95ba-a6b1d65d0c43
CSeq: 23191 INVITE
User-Agent: packetrino
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: sip:1800555@64.2.142.192
Content-Type: application/sdp
Content-Length: 240

v=0
o=root 32312 32312 IN IP4 64.2.142.192
s=session
c=IN IP4 64.2.142.192
t=0 0
m=audio 17494 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
 
Phone is ringing.
Next, I answer my cell phone


--- Received SIP response (830 bytes) from UDP:64.2.142.93:5060 ---
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
XXX.XXX.XXX.XXX:5060;received=XXX.XXX.XXX.XXX;rport=5060;branch=z9hG4bKPj61911226-4812-44da-b682-7ec02e8c821a
Record-Route: sip:64.2.142.93;lr=on
From: Dan sip:2...@xxx.xxx.xxx.xxx;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8
To: sip:800555@64.2.142.93;tag=as7aad862a
Call-ID: 309ec892-56a8-46d2-95ba-a6b1d65d0c43
CSeq: 23191 INVITE
User-Agent: packetrino
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: sip:1800555@64.2.142.192
Content-Type: application/sdp
Content-Length: 240

v=0
o=root 32312 32313 IN IP4 64.2.142.192
s=session
c=IN IP4 64.2.142.192
t=0 0
m=audio 17494 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

--- Transmitting SIP request (437 bytes) to UDP:64.2.142.93:5060 ---
ACK sip:1800555@64.2.142.93:5060 SIP/2.0
Via: SIP/2.0/UDP 
XXX.XXX.XXX.XXX:5060;rport;branch=z9hG4bKPj744f71bf-b90a-4e49-9dfb-42ff4aa3a641
From: Dan sip:2...@xxx.xxx.xxx.xxx;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8
To: sip:800555@64.2.142.93;tag=as7aad862a
Call-ID: 309ec892-56a8-46d2-95ba-a6b1d65d0c43
CSeq: 23191 ACK
Route: sip:64.2.142.93;lr
Max-Forwards: 70
User-Agent: Asterisk PBX 13.0.0
Content-Length:  0


--- Received SIP response (830 bytes) from UDP:64.2.142.93:5060 ---
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
XXX.XXX.XXX.XXX:5060;received=XXX.XXX.XXX.XXX;rport=5060;branch=z9hG4bKPj61911226-4812-44da-b682-7ec02e8c821a
Record-Route: sip:64.2.142.93;lr=on
From: Dan sip:2...@xxx.xxx.xxx.xxx;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8
To: sip:800555@64.2.142.93;tag=as7aad862a
Call-ID: 309ec892-56a8-46d2-95ba-a6b1d65d0c43
CSeq: 23191 INVITE
User-Agent: packetrino
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: sip:1800555@64.2.142.192
Content-Type: application/sdp
Content-Length: 240

v=0
o=root 32312 32313 IN IP4 64.2.142.192
s=session
c=IN IP4 64.2.142.192
t=0 0
m=audio 17494 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

--- Transmitting SIP request (437 bytes) to UDP:64.2.142.93:5060 ---
ACK 

Re: [asterisk-users] PJSIP configuration question

[Dan Cropp]

Ugh.

I'm having a bad day.  The two traces were swapped.

The one on Asterisk 13 is PJSIP.
The one on Asterisk 12 is using chan_sip.


-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Cropp
Sent: Thursday, 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 for the pjsip.conf which it seems Vitelity 
isn't accepting the ACK in response to the OK

 SIP ---

--- Transmitting SIP request (1004 bytes) to UDP:64.2.142.93:5060 --- INVITE 
sip:800555@64.2.142.93 SIP/2.0
Via: SIP/2.0/UDP 
XXX.XXX.XXX.XXX:5060;rport;branch=z9hG4bKPj61911226-4812-44da-b682-7ec02e8c821a
From: Dan sip:2...@xxx.xxx.xxx.xxx;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8
To: sip:800555@64.2.142.93
Contact: sip:15062fef-986e-4fcf-a93e-06b28da02...@xxx.xxx.xxx.xxx:5060
Call-ID: 309ec892-56a8-46d2-95ba-a6b1d65d0c43
CSeq: 23191 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, 
PRACK, REFER, REGISTER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Remote-Party-ID: Dan sip:2...@xxx.xxx.xxx.xxx;privacy=off;screen=no
Max-Forwards: 70
User-Agent: Asterisk PBX 13.0.0
Content-Type: application/sdp
Content-Length:   239

v=0
o=- 540555224 540555224 IN IP4 XXX.XXX.XXX.XXX s=Asterisk c=IN IP4 
XXX.XXX.XXX.XXX
t=0 0
m=audio 10030 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

--- Received SIP response (378 bytes) from UDP:64.2.142.93:5060 ---
SIP/2.0 100 Giving a try
Via: SIP/2.0/UDP 
XXX.XXX.XXX.XXX:5060;rport=5060;branch=z9hG4bKPj61911226-4812-44da-b682-7ec02e8c821a
From: Dan sip:2...@xxx.xxx.xxx.xxx;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8
To: sip:800555@64.2.142.93
Call-ID: 309ec892-56a8-46d2-95ba-a6b1d65d0c43
CSeq: 23191 INVITE
Server: OpenSIPS (1.6.4-2-notls (x86_64/linux))
Content-Length: 0





--- Received SIP response (844 bytes) from UDP:64.2.142.93:5060 ---
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 
XXX.XXX.XXX.XXX:5060;received=XXX.XXX.XXX.XXX;rport=5060;branch=z9hG4bKPj61911226-4812-44da-b682-7ec02e8c821a
Record-Route: sip:64.2.142.93;lr=on
From: Dan sip:2...@xxx.xxx.xxx.xxx;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8
To: sip:800555@64.2.142.93;tag=as7aad862a
Call-ID: 309ec892-56a8-46d2-95ba-a6b1d65d0c43
CSeq: 23191 INVITE
User-Agent: packetrino
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: sip:1800555@64.2.142.192
Content-Type: application/sdp
Content-Length: 240

v=0
o=root 32312 32312 IN IP4 64.2.142.192
s=session
c=IN IP4 64.2.142.192
t=0 0
m=audio 17494 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
 
Phone is ringing.
Next, I answer my cell phone


--- Received SIP response (830 bytes) from UDP:64.2.142.93:5060 ---
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
XXX.XXX.XXX.XXX:5060;received=XXX.XXX.XXX.XXX;rport=5060;branch=z9hG4bKPj61911226-4812-44da-b682-7ec02e8c821a
Record-Route: sip:64.2.142.93;lr=on
From: Dan sip:2...@xxx.xxx.xxx.xxx;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8
To: sip:800555@64.2.142.93;tag=as7aad862a
Call-ID: 309ec892-56a8-46d2-95ba-a6b1d65d0c43
CSeq: 23191 INVITE
User-Agent: packetrino
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: sip:1800555@64.2.142.192
Content-Type: application/sdp
Content-Length: 240

v=0
o=root 32312 32313 IN IP4 64.2.142.192
s=session
c=IN IP4 64.2.142.192
t=0 0
m=audio 17494 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

--- Transmitting SIP request (437 bytes) to UDP:64.2.142.93:5060 --- ACK 
sip:1800555@64.2.142.93:5060 SIP/2.0
Via: SIP/2.0/UDP 
XXX.XXX.XXX.XXX:5060;rport;branch=z9hG4bKPj744f71bf-b90a-4e49-9dfb-42ff4aa3a641
From: Dan sip:2...@xxx.xxx.xxx.xxx;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8
To: sip:800555@64.2.142.93;tag=as7aad862a
Call-ID: 309ec892-56a8-46d2-95ba-a6b1d65d0c43
CSeq: 23191 ACK
Route: sip:64.2.142.93;lr
Max-Forwards: 70
User-Agent: Asterisk PBX 13.0.0
Content-Length:  0


--- Received SIP response (830 bytes) from UDP:64.2.142.93:5060 ---
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
XXX.XXX.XXX.XXX:5060;received=XXX.XXX.XXX.XXX;rport=5060;branch=z9hG4bKPj61911226-4812-44da-b682-7ec02e8c821a
Record-Route: sip:64.2.142.93;lr=on
From: Dan sip:2...@xxx.xxx.xxx.xxx;tag=781b4b63-0df9-47ad-9fa1-52a8729359c8
To: sip:800555@64.2.142.93;tag=as7aad862a
Call-ID: 309ec892-56a8-46d2-95ba-a6b1d65d0c43
CSeq: 23191 INVITE
User-Agent: packetrino
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE

[asterisk-users] PJSIP configuration question

[Dan Cropp]

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 suggestions as to what I am doing wrong?
The SIP provider says the latest version of Asterisk they have anyone using is 
Asterisk 11, so they have no PJSIP configuration experience.

The only setting that I believe I haven't found a PJSIP settting for is the 
insecure=invite from sip.conf
I thought that would be the equivalent of no authentication object, so I tried 
that.  However, that did not work either.

I tried changing the endpoint to have no auth and outbound_auth settings.
I tried changing the endpoint to use the auth instead of the outbound_auth.

The SIP provider even changed the username and passwords to blank.  I followed 
suit and changed the pjsip.conf user and password related settings to blank.


Our sip.conf (running in a different VM on Asterisk 13.0.0) settings look like 
this...
[x]
type = friend
qualify = no
nat = yes
host = x
defaultuser = y
secret = z
incominglimit = 4
accountcode = 9
port = 5060
context = TestApp
dtmfmode = auto
insecure = invite
fromdomain = x
fromuser = y
sendrpid = yes
trustrpid = yes
canreinvite = no

For the pjsip.conf settings (Asterisk 13.0.0), I have
[transport1]
type = transport
bind = 0.0.0.0
protocol = udp

[x]
type = aor
max_contacts = 1
remove_existing = yes
contact = sip:y@x:5060

[auth9]
type = auth
username = y
password = z

[x]
type = endpoint
context = TestApp
transport = transport1
outbound_auth = auth9
aors = x
accountcode = 9
dtmf_mode = rfc4733
device_state_busy_at = 4
;force_rport = yes   ; also tried with this setting, 
but it still didn't help
rtp_symmetric = yes
rewrite_contact = yes
from_domain = xx
from_user = y
send_rpid = yes
trust_id_inbound = yes
direct_media = no


Have a great day!
Dan
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Re: [asterisk-users] PJSIP configuration question

[Joshua Colp]

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 Originate?


I’m currently running Asterisk 13.0.0.

Anyone have any suggestions as to what I am doing wrong?

The SIP provider says the latest version of Asterisk they have anyone
using is Asterisk 11, so they have no PJSIP configuration experience.

The only setting that I believe I haven’t found a PJSIP settting for is
the “insecure=invite” from sip.conf


That functionality exists in the form of the identify object. It does 
IP based matching of incoming traffic and to associate it with an endpoint.




I thought that would be the equivalent of no authentication object, so I
tried that. However, that did not work either.


Authentication controls authentication, it doesn't control how PJSIP 
associates traffic with a specific endpoint. They are separate things.


I think before we get into config we need to see the dial string for 
your origination.


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

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Re: [asterisk-users] PJSIP configuration question

[Dan Cropp]

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 based solution, it's...
SIP/outbound.vitelity.net/1234567890

Have a great day!

Dan

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp
Sent: Wednesday, December 10, 2014 9:35 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
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 any PJSIP messages.

What dial string are you providing to Originate?

 I'm currently running Asterisk 13.0.0.

 Anyone have any suggestions as to what I am doing wrong?

 The SIP provider says the latest version of Asterisk they have anyone 
 using is Asterisk 11, so they have no PJSIP configuration experience.

 The only setting that I believe I haven't found a PJSIP settting for 
 is the insecure=invite from sip.conf

That functionality exists in the form of the identify object. It does IP 
based matching of incoming traffic and to associate it with an endpoint.


 I thought that would be the equivalent of no authentication object, so 
 I tried that. However, that did not work either.

Authentication controls authentication, it doesn't control how PJSIP associates 
traffic with a specific endpoint. They are separate things.

I think before we get into config we need to see the dial string for your 
origination.

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

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Re: [asterisk-users] PJSIP configuration question

[Dan Cropp]

I should mention, I am actually sending this via AMI in both the chan_sip and 
the pjsip case.

Pjsip originate...

Action: Originate
ActionID: S8
Channel: PJSIP/outbound.vitelity.net/1234567890
Exten: createcall
Context: TestApp
Priority: 1
Timeout: 6
CallerID: Dan Cropp1234
Variable: CALLERID(num-pres)=allowed_passed_screened
Async: true


Chan_sip based originate...

Action: Originate
ActionID: S8
Channel: SIP/outbound.vitelity.net/1234567890
Exten: createcall
Context: TestApp
Priority: 1
Timeout: 6
CallerID: Dan Cropp1234
Variable: CALLERID(num-pres)=allowed_passed_screened
Async: true

Have a great day!

Dan

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Cropp
Sent: Wednesday, December 10, 2014 10:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [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 based solution, it's...
SIP/outbound.vitelity.net/1234567890

Have a great day!

Dan

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp
Sent: Wednesday, December 10, 2014 9:35 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
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 any PJSIP messages.

What dial string are you providing to Originate?

 I'm currently running Asterisk 13.0.0.

 Anyone have any suggestions as to what I am doing wrong?

 The SIP provider says the latest version of Asterisk they have anyone 
 using is Asterisk 11, so they have no PJSIP configuration experience.

 The only setting that I believe I haven't found a PJSIP settting for 
 is the insecure=invite from sip.conf

That functionality exists in the form of the identify object. It does IP 
based matching of incoming traffic and to associate it with an endpoint.


 I thought that would be the equivalent of no authentication object, so 
 I tried that. However, that did not work either.

Authentication controls authentication, it doesn't control how PJSIP associates 
traffic with a specific endpoint. They are separate things.

I think before we get into config we need to see the dial string for your 
origination.

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

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[asterisk-users] PJSIP configuration question

[Dan Cropp]

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 sip.conf they sent me, 
everything works.

Action: Originate
ActionID: S8
Channel: SIP/outbound.vitelity.net/800555
Exten: createcall
Context: TestApp
Priority: 1
Timeout: 6
CallerID: John Doe 1234
Variable: CALLERID(num-pres)=allowed_passed_screened
Async: true


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

[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
allow = all
direct_media = no

[identify1]
type = identify
endpoint = outbound.vitelity.net
match = 64.2.142.93

When I attempt to use AMI Originate, it's failing.  I am not seeing anything 
with pjsip logging turned on, so it seems to be something with the settings.

Action: Originate
ActionID: S8
Channel: PJSIP/outbound.vitelity.net/800555
Exten: createcall
Context: TestApp
Priority: 1
Timeout: 6
CallerID: John Doe 1234
Variable: CALLERID(num-pres)=allowed_passed_screened
Async: true

NOTE: I am able to use AMI Originate to other PJSIP endpoints.

Action: Originate
ActionID: S9
Channel: PJSIP/1003/1003
Exten: createcall
Context: TestApp
Priority: 1
Timeout: 6
CallerID: John Doe 1234
Variable: CALLERID(num-pres)=allowed_passed_screened
Async: true

Anyone have any suggestions as to what I am doing wrong?

Have a great day!

Dan

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Re: [asterisk-users] PJSIP configuration question

[George Joseph]

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
 sendrpid=yes

 When I use these settings to originate calls using the sip.conf they sent
 me, everything works.

 Action: Originate
 ActionID: S8
 Channel: SIP/outbound.vitelity.net/800555
 Exten: createcall
 Context: TestApp
 Priority: 1
 Timeout: 6
 CallerID: John Doe 1234
 Variable: CALLERID(num-pres)=allowed_passed_screened
 Async: true


 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


You might want to set a qualify_frequency here  to see if the server
responds to OPTIONS messages.  Also 64.2.142.93 isn't currently one of
their outbound servers.  Are you using one of their inbound* servers as
outbound?  IIRC unless you ask them, they don't allow it.


 [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
 allow = all
 direct_media = no

 [identify1]
 type = identify
 endpoint = outbound.vitelity.net
 match = 64.2.142.93

 When I attempt to use AMI Originate, it's failing.  I am not seeing
 anything with pjsip logging turned on, so it seems to be something with the
 settings.

 Action: Originate
 ActionID: S8
 Channel: PJSIP/outbound.vitelity.net/800555
 Exten: createcall
 Context: TestApp
 Priority: 1
 Timeout: 6
 CallerID: John Doe 1234
 Variable: CALLERID(num-pres)=allowed_passed_screened
 Async: true

 NOTE: I am able to use AMI Originate to other PJSIP endpoints.

 Action: Originate
 ActionID: S9
 Channel: PJSIP/1003/1003
 Exten: createcall
 Context: TestApp
 Priority: 1
 Timeout: 6
 CallerID: John Doe 1234
 Variable: CALLERID(num-pres)=allowed_passed_screened
 Async: true

 Anyone have any suggestions as to what I am doing wrong?

 Have a great day!

 Dan

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Re: [asterisk-users] PJSIP configuration question

[Dan Cropp]

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 
something configured incorrectly….

--- Transmitting SIP request (483 bytes) to UDP:0.0.19.196:5060 ---
OPTIONS sip:64.2.142.93@5060 SIP/2.0
Via: SIP/2.0/UDP 
xxx.xxx.xx.xxx:5060;rport;branch=z9hG4bKPjcea63914-b8d1-483d-96db-11968abab704
From: 
sip:e31d5809-f26a-4219-8365-709314280...@xxx.xxx.xx.xxx;tag=7cfab3ba-73de-4243-9967-d1e6a5e7b0b4
To: sip:64.2.142.93@5060
Contact: sip:e31d5809-f26a-4219-8365-709314280...@xxx.xxx.xx.xxx:5060
Call-ID: 7ba766bf-363b-47d0-a388-62a58d1df88d
CSeq: 33778 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 13.0.0
Content-Length:  0


[Dec 17 19:22:31] WARNING[49476]: pjsip:0 ?:tsx0x3c501e8 .Failed to send 
Request msg OPTIONS/cseq=33778 (tdta0x32c7c90)! err=120022 (Invalid argument)
[Dec 17 19:22:31] ERROR[49476]: res_pjsip.c:2532 endpt_send_request: Error 
120022 'Invalid argument' sending OPTIONS request to endpoint unknown


The 64.2.142.93 is the exact value I was given to use for the outbound trunk 
(works with sip.conf)
host=64.2.142.93
Any thoughts?
I was really hoping they had worked with the PJSIP, but apparently the latest 
Asterisk version any of their customers are using is Asterisk 11.

Have a great day!

Dan

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] 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, 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 sip.conf they sent me, 
everything works.

Action: Originate
ActionID: S8
Channel: 
SIP/outbound.vitelity.net/800555http://outbound.vitelity.net/800555
Exten: createcall
Context: TestApp
Priority: 1
Timeout: 6
CallerID: John Doe 1234
Variable: CALLERID(num-pres)=allowed_passed_screened
Async: true


I translated those settings to the following for pjsip.conf...

[transport1]
type = transport
bind = 0.0.0.0
protocol = udp

[outbound.vitelity.nethttp://outbound.vitelity.net]
type = aor
remove_existing = yes
contact = sip:64.2.142.93@5060

You might want to set a qualify_frequency here  to see if the server responds 
to OPTIONS messages.  Also 64.2.142.93 isn't currently one of their outbound 
servers.  Are you using one of their inbound* servers as outbound?  IIRC unless 
you ask them, they don't allow it.

[outbound.vitelity.nethttp://outbound.vitelity.net]
type = endpoint
context = TestApp
transport = transport1
aors = outbound.vitelity.nethttp://outbound.vitelity.net
dtmf_mode = rfc4733
force_rport = yes
rtp_symmetric = yes
rewrite_contact = yes
send_rpid = yes
trust_id_inbound = yes
allow = all
direct_media = no

[identify1]
type = identify
endpoint = outbound.vitelity.nethttp://outbound.vitelity.net
match = 64.2.142.93

When I attempt to use AMI Originate, it's failing.  I am not seeing anything 
with pjsip logging turned on, so it seems to be something with the settings.

Action: Originate
ActionID: S8
Channel: 
PJSIP/outbound.vitelity.net/800555http://outbound.vitelity.net/800555
Exten: createcall
Context: TestApp
Priority: 1
Timeout: 6
CallerID: John Doe 1234
Variable: CALLERID(num-pres)=allowed_passed_screened
Async: true

NOTE: I am able to use AMI Originate to other PJSIP endpoints.

Action: Originate
ActionID: S9
Channel: PJSIP/1003/1003
Exten: createcall
Context: TestApp
Priority: 1
Timeout: 6
CallerID: John Doe 1234
Variable: CALLERID(num-pres)=allowed_passed_screened
Async: true

Anyone have any suggestions as to what I am doing wrong?

Have a great day!

Dan

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Re: [asterisk-users] PJSIP configuration question

[George Joseph]

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 something configured incorrectly….



 --- Transmitting SIP request (483 bytes) to UDP:0.0.19.196:5060 ---


Well, THAT's not right.  Did you obfuscate the 0.0.19.196 or is that how it
really is?  Are you NATed?



 OPTIONS sip:64.2.142.93@5060 SIP/2.0

 Via: SIP/2.0/UDP
 xxx.xxx.xx.xxx:5060;rport;branch=z9hG4bKPjcea63914-b8d1-483d-96db-11968abab704

 From: sip:e31d5809-f26a-4219-8365-709314280...@xxx.xxx.xx.xxx
 ;tag=7cfab3ba-73de-4243-9967-d1e6a5e7b0b4

 To: sip:64.2.142.93@5060

 Contact: sip:e31d5809-f26a-4219-8365-709314280...@xxx.xxx.xx.xxx:5060

 Call-ID: 7ba766bf-363b-47d0-a388-62a58d1df88d

 CSeq: 33778 OPTIONS

 Max-Forwards: 70

 User-Agent: Asterisk PBX 13.0.0

 Content-Length:  0





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Re: [asterisk-users] PJSIP configuration question

[Joshua Colp]

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 will use a default port of 5060.

I also believe I've covered your origination issue in a separate email. 
Your dial string should be:


PJSIP/800...@outbound.vitelity.net

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

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