VoIP Mailing List Archives

[dan at amtelco.com]

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…
[xxxxx]
type = friend
qualify = no
nat = yes
host = xxxxx
defaultuser = yyyyy
secret = zzzzz
incominglimit = 4
accountcode = 9
port = 5060
context = TestApp
dtmfmode = auto
insecure = invite
fromdomain = xxxxx
fromuser = yyyyy
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

[xxxxx]
type = aor
max_contacts = 1
remove_existing = yes
contact = [url=sip:yyyyy@xxxxx:5060]sip:yyyyy@xxxxx:5060[/url]

[auth9]
type = auth
username = yyyyy
password = zzzzz

[xxxxx]
type = endpoint
context = TestApp
transport = transport1
outbound_auth = auth9
aors = xxxxx
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 = xxxxxx
from_user = yyyyy
send_rpid = yes
trust_id_inbound = yes
direct_media = no


Have a great day!
Dan

[jcolp at digium.com]

Kia ora,

Dan Cropp wrote:

[dan at amtelco.com]

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: 60000
CallerID: Dan Cropp<1234>
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: 60000
CallerID: Dan Cropp<1234>
Variable: CALLERID(num-pres)=allowed_passed_screened
Async: true

Have a great day!

Dan

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@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 xxxxx 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-bounces@lists.digium.com [mailto:asterisk-users-bounces@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:

[dan at amtelco.com]

Thank you for the speedy reply.

My originate string is something like the following where
xxxxx 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-bounces@lists.digium.com [mailto:asterisk-users-bounces@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:

[dan at amtelco.com]

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/8005555555
Exten: createcall
Context: TestApp
Priority: 1
Timeout: 60000
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/8005555555
Exten: createcall
Context: TestApp
Priority: 1
Timeout: 60000
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: 60000
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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[george.joseph at fairv...]

On Wed, Dec 10, 2014 at 1:27 PM, Dan Cropp <dan@amtelco.com (dan@amtelco.com)> wrote:

[dan at amtelco.com]

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-4219-8365-70931428072b@xxx.xxx.xx.xxx>;tag=7cfab3ba-73de-4243-9967-d1e6a5e7b0b4
To: <sip:64.2.142.93@5060>
Contact: <sip:e31d5809-f26a-4219-8365-70931428072b@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-bounces@lists.digium.com [mailto:asterisk-users-bounces@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 <dan@amtelco.com (dan@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/8005555555
Exten: createcall
Context: TestApp
Priority: 1
Timeout: 60000
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 = [url=sip:64.2.142.93@5060]sip:64.2.142.93@5060[/url]


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.

[george.joseph at fairv...]

On Wed, Dec 10, 2014 at 2:03 PM, Dan Cropp <dan@amtelco.com (dan@amtelco.com)> wrote:

[jcolp at digium.com]

<snip>

[dan at amtelco.com]

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-bounces@lists.digium.com [mailto:asterisk-users-bounces@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>

[dan at amtelco.com]

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 <dan@amtelco.com (dan@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?

[dan at amtelco.com]

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-bounces@lists.digium.com [mailto:asterisk-users-bounces@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-bounces@lists.digium.com [mailto:asterisk-users-bounces@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>

[dan at amtelco.com]

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-bounces@lists.digium.com [mailto:asterisk-users-bounces@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-bounces@lists.digium.com [mailto:asterisk-users-bounces@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-bounces@lists.digium.com [mailto:asterisk-users-bounces@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>

[dan at amtelco.com]

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:1234@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-bounces@lists.digium.com [mailto:asterisk-users-bounces@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-bounces@lists.digium.com [mailto:asterisk-users-bounces@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-bounces@lists.digium.com [mailto:asterisk-users-bounces@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-bounces@lists.digium.com [mailto:asterisk-users-bounces@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>

[jcolp at digium.com]

Dan Cropp wrote: