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[asterisk-users] PJSIP configuration question


 
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dan at amtelco.com
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PostPosted: Tue Dec 15, 2015 11:19 am    Post subject: [asterisk-users] PJSIP configuration question Reply with quote

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" <sip:didassignedbybluip at myinternalipaddress>;tag=4fda8a53-8831-45bd-9b29-15eb276ceafb
To: <sip:callphonenumber at bluipaddress >
Contact: <sip:d2a29c8c-d960-411a-aeb0-391313efe60e@myinternalipaddress:5060>
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" <sip:didassignedbybluip at myinternalipaddress >;tag=4fda8a53-8831-45bd-9b29-15eb276ceafb
To: <sip:callphonenumber at bluipaddress>
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" <sip:didassignedbybluip at myinternalipaddress >;tag=4fda8a53-8831-45bd-9b29-15eb276ceafb
To: <sip:callphonenumber at bluipaddress>;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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jcolp at digium.com
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PostPosted: Tue Dec 15, 2015 11:26 am    Post subject: [asterisk-users] PJSIP configuration question Reply with quote

Dan Cropp wrote:
Quote:
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.

<snip>

Quote:

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...

--
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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dan at amtelco.com
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PostPosted: Tue Dec 15, 2015 11:58 am    Post subject: [asterisk-users] PJSIP configuration question Reply with quote

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: <sip:callphonenumber@companyname.com>
From: <sip: didassignedbybluip@companyname.com>;tag=JEDtnWPm
Contact: <sip:didassignedbybluip@ipaddress:5060>
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: <sip:callphonenumber@companyname.com>
From: <sip:didassignedbybluip@companyname.com>;tag=JEDtnWPm
Call-ID: kD5sJntD5-0001-@ ipaddress
CSeq: 1808 INVITE

401 Unauthorized
Via: SIP/2.0/UDP ipaddress:5060;branch=z9hG4bKQFD28QYY8412c000
To: <sip: callphonenumber@companyname.com>;tag=1164309609-1450128136967
From: <sip: didassignedbybluip@companyname.com>;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: <sip:callphonenumbe @companyname.com>;tag=1164309609-1450128136967
From: <sip: didassignedbybluip @companyname.com>;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=z9hG4bKYcoOR2md84130000
To: <sip:callphonenumber@companyname.com>
From: <sip: didassignedbybluip @companyname.com>;tag=pcjBsnMZ
Contact: <sip: didassignedbybluip @ ipaddress:5060>
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=z9hG4bKYcoOR2md84130000
To: <sip:callphonenumber@companyname.com>
From: <sip: didassignedbybluip @companyname.com>;tag=pcjBsnMZ
Call-ID: kD5sJntD5-0001-@ipaddress
CSeq: 1809 INVITE

180 Ringing
Via: SIP/2.0/UDP ipaddress:5060;branch=z9hG4bKYcoOR2md84130000
To: <sip:callphonenumber@companyname.com>;tag=878577519-1450128140899
From: <sip: didassignedbybluip @companyname.com>;tag=pcjBsnMZ
Call-ID: kD5sJntD5-0001-@ipaddress
CSeq: 1809 INVITE
Supported:
Contact: <sip:callphonenumber@bluipaddress:5060;transport=udp>
P-Asserted-Identity: <sip:callphonenumber@199.168.176.135;user=phone>
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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jcolp at digium.com
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PostPosted: Tue Dec 15, 2015 12:17 pm    Post subject: [asterisk-users] PJSIP configuration question Reply with quote

Dan Cropp wrote:
Quote:
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

--
_____________________________________________________________________
-- 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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dan at amtelco.com
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PostPosted: Tue Dec 15, 2015 1:05 pm    Post subject: [asterisk-users] PJSIP configuration question Reply with quote

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

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

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

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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