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[asterisk-users] PJSIP outbound register and inbound calls


 
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jleed at me.com
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PostPosted: Wed Jul 16, 2014 10:46 am    Post subject: [asterisk-users] PJSIP outbound register and inbound calls Reply with quote

Hi all,
In my case I using realtime,
here is how it looks in plant

[10001]
type=registration
transport=upd_static
outbound_auth=10001
server_uri=sip:600@192.168.1.1:5060
client_uri=sip:600@192.168.1.4:5060
[10001]
type=auth
auth_type=userpass
password=600
username=600
[10001]
type=aor
contact=sip:192.168.1.4:5060
[10001]
type=endpoint
transport=upd_static
context=dialmap
disallow=all
allow=ulaw
outbound_auth=10001
aors=10001
[10001]
type=identify
endpoint=10001
match=192.168.1.1
when I call 600 from other pbx I getting an notice

NOTICE[10202]: res_pjsip/pjsip_distributor.c:246 log_unidentified_request: Request from '"Ilya" <sip:502@192.168.1.1>' failed for '192.168.1.1:5060' (callid: ZTNhYjU4ZjU5ZmUxNjM5M2FlYjBlYTE3YzgwZTU4MGY.) - No matching endpoint found
and "Not Accessable" on phone

let's imagine that 600 its external number of voip operator, and I wanna accept all incoming calls from it (no matter what caller id it has)
what I doing wrong?


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jcolp at digium.com
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PostPosted: Wed Jul 16, 2014 10:55 am    Post subject: [asterisk-users] PJSIP outbound register and inbound calls Reply with quote

Nick Awesome wrote:
Quote:
Hi all, In my case I using realtime, here is how it looks in plant

[10001] type=registration transport=upd_static outbound_auth=10001
server_uri=sip:600@192.168.1.1:5060
client_uri=sip:600@192.168.1.4:5060 [10001] type=auth
auth_type=userpass password=600 username=600 [10001] type=aor
contact=sip:192.168.1.4:5060 [10001] type=endpoint
transport=upd_static context=dialmap disallow=all allow=ulaw
outbound_auth=10001 aors=10001 [10001] type=identify endpoint=10001
match=192.168.1.1 when I call 600 from other pbx I getting an notice

NOTICE[10202]: res_pjsip/pjsip_distributor.c:246
log_unidentified_request: Request from '"Ilya"<sip:502@192.168.1.1>'
failed for '192.168.1.1:5060' (callid:
ZTNhYjU4ZjU5ZmUxNjM5M2FlYjBlYTE3YzgwZTU4MGY.) - No matching endpoint
found and "Not Accessable" on phone

let's imagine that 600 its external number of voip operator, and I
wanna accept all incoming calls from it (no matter what caller id it
has) what I doing wrong?

When receiving calls from a VoIP provider you have to match using the
source IP address. You also don't authenticate as the provider will
refuse to do so.

When you control both ends it's really up to you whether to do the
matching based on the source IP address OR use a user account with
authentication. If using the user account the user portion of the From
header has to be set to the username (from_user in pjsip, fromuser in
chan_sip).

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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jleed at me.com
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PostPosted: Wed Jul 16, 2014 11:47 am    Post subject: [asterisk-users] PJSIP outbound register and inbound calls Reply with quote

I thought that
Quote:
Quote:
type=identify
will match an IP address and accept it,

well, in my example I can control both sides and able to configure it without registration.
in real life I have a provider that requires username/password authentication

provider gives me
- Username
- Password
- DomainName

I have configure it like I showed before and have exactly the same notice

[Jul 16 20:32:23] NOTICE[21926]: res_pjsip/pjsip_distributor.c:246 log_unidentified_request: Request from '"cb5069" <sip:asterisk@85.195.98.178>' failed for '85.195.98.178:5060' (callid: 173995aa2e25283807700d65055c9214@85.195.98.178) - No matching endpoint found
85.195.98.178 is an operator,

so what I should add to my config to be able accept calls from Registered peer ?


On Jul 16, 2014, at 7:55 PM, Joshua Colp <jcolp@digium.com> wrote:

Quote:
Nick Awesome wrote:
Quote:
Hi all, In my case I using realtime, here is how it looks in plant

[10001] type=registration transport=upd_static outbound_auth=10001
server_uri=sip:600@192.168.1.1:5060
client_uri=sip:600@192.168.1.4:5060 [10001] type=auth
auth_type=userpass password=600 username=600 [10001] type=aor
contact=sip:192.168.1.4:5060 [10001] type=endpoint
transport=upd_static context=dialmap disallow=all allow=ulaw
outbound_auth=10001 aors=10001 [10001] type=identify endpoint=10001
match=192.168.1.1 when I call 600 from other pbx I getting an notice

NOTICE[10202]: res_pjsip/pjsip_distributor.c:246
log_unidentified_request: Request from '"Ilya"<sip:502@192.168.1.1>'
failed for '192.168.1.1:5060' (callid:
ZTNhYjU4ZjU5ZmUxNjM5M2FlYjBlYTE3YzgwZTU4MGY.) - No matching endpoint
found and "Not Accessable" on phone

let's imagine that 600 its external number of voip operator, and I
wanna accept all incoming calls from it (no matter what caller id it
has) what I doing wrong?

When receiving calls from a VoIP provider you have to match using the source IP address. You also don't authenticate as the provider will refuse to do so.

When you control both ends it's really up to you whether to do the matching based on the source IP address OR use a user account with authentication. If using the user account the user portion of the From header has to be set to the username (from_user in pjsip, fromuser in chan_sip).

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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jcolp at digium.com
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PostPosted: Wed Jul 16, 2014 11:53 am    Post subject: [asterisk-users] PJSIP outbound register and inbound calls Reply with quote

Nick Awesome wrote:
Quote:
I thought that
Quote:
Quote:
type=identify
will match an IP address and accept it,

well, in my example I can control both sides and able to configure it
without registration. in real life I have a provider that requires
username/password authentication

provider gives me - Username - Password - DomainName

They may require it for *outgoing* calls to them but for incoming I
highly doubt they'd want you to authenticate them. It's usually always
IP authentication.

Quote:
I have configure it like I showed before and have exactly the same
notice

[Jul 16 20:32:23] NOTICE[21926]: res_pjsip/pjsip_distributor.c:246
log_unidentified_request: Request from
'"cb5069"<sip:asterisk@85.195.98.178>' failed for
'85.195.98.178:5060' (callid:
173995aa2e25283807700d65055c9214@85.195.98.178) - No matching
endpoint found 85.195.98.178 is an operator,

so what I should add to my config to be able accept calls from
Registered peer ?

The PJSIP functionality does not currently allow using the dynamic IP of
a registration to match an incoming call. You either have to explicitly
use the identify section or match as I previously described.

Without further details of your setup (IP addresses, who are calling
who) and how you want it to work I can't answer.

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Back to top
jleed at me.com
Guest





PostPosted: Wed Jul 16, 2014 12:13 pm    Post subject: [asterisk-users] PJSIP outbound register and inbound calls Reply with quote

Ok there is my test account from sipiko.net

username: cb5069
password: sqv664yqtp
domain: callme.sipiko.net


its using username/password authentication.
because its just website widget I need only inbound calls from this peer,
test call can be done from url:
http://callme.sipiko.net/callme.php?id=5069&call_id=210&tunnel=yes


on my side I have an asterisk 12 using pjsip


Have configured IVR with number 5000 on context "dialmap", so I need forward all calls from this provider to number 5000 over "dialmap" context


help if you can please:)


On Jul 16, 2014, at 8:53 PM, Joshua Colp <jcolp@digium.com (jcolp@digium.com)> wrote:
Quote:
Nick Awesome wrote:
Quote:
I thought that
Quote:
Quote:
type=identify
will match an IP address and accept it,

well, in my example I can control both sides and able to configure it
without registration. in real life I have a provider that requires
username/password authentication

provider gives me - Username - Password - DomainName

They may require it for *outgoing* calls to them but for incoming I
highly doubt they'd want you to authenticate them. It's usually always
IP authentication.

Quote:
I have configure it like I showed before and have exactly the same
notice

[Jul 16 20:32:23] NOTICE[21926]: res_pjsip/pjsip_distributor.c:246
log_unidentified_request: Request from
'"cb5069"<[url=sip:asterisk@85.195.98.178]sip:asterisk@85.195.98.178[/url]>' failed for
'85.195.98.178:5060' (callid:
173995aa2e25283807700d65055c9214@85.195.98.178) - No matching
endpoint found 85.195.98.178 is an operator,

so what I should add to my config to be able accept calls from
Registered peer ?

The PJSIP functionality does not currently allow using the dynamic IP of a registration to match an incoming call. You either have to explicitly use the identify section or match as I previously described.

Without further details of your setup (IP addresses, who are calling who) and how you want it to work I can't answer.

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Back to top
jleed at me.com
Guest





PostPosted: Thu Jul 17, 2014 2:06 am    Post subject: [asterisk-users] PJSIP outbound register and inbound calls Reply with quote

New information, as I said I’m using realtime,thats the problem!


I just tested using static config file and it is working perfect.
after some research I figured out that problem with “ps_endpoint_id_ips" for some reason asterisk ignoring matches in this table,


I have string in sorcery.conf


identify = realtime,ps_endpoint_id_ips


also have string in extconfig.conf


ps_endpoint_id_ips => odbc,asterisk,pbx_endpoint_id_ips


and ofc I have table


CREATE TABLE `pbx_endpoint_id_ips` (
`id` varchar(40) NOT NULL,
`endpoint` varchar(40) DEFAULT NULL,
`match` varchar(80) DEFAULT NULL,
UNIQUE KEY `id` (`id`),
KEY `ps_endpoint_id_ips_id` (`id`)
) ENGINE=InnoDB DEFAULT CHARSET=latin1;



with entry


10001 | 10001 | 85.195.98.178


but thats just didn’t works(


is this a bug and should I open ticket ?


On 16 Jul 2014, at 21:13, Nick Awesome <jleed@me.com (jleed@me.com)> wrote:
Quote:
Ok there is my test account from sipiko.net

username: cb5069
password: sqv664yqtp
domain: callme.sipiko.net


its using username/password authentication.
because its just website widget I need only inbound calls from this peer,
test call can be done from url:
http://callme.sipiko.net/callme.php?id=5069&call_id=210&tunnel=yes


on my side I have an asterisk 12 using pjsip


Have configured IVR with number 5000 on context "dialmap", so I need forward all calls from this provider to number 5000 over "dialmap" context


help if you can please:)


On Jul 16, 2014, at 8:53 PM, Joshua Colp <jcolp@digium.com (jcolp@digium.com)> wrote:
Quote:
Nick Awesome wrote:
Quote:
I thought that
Quote:
Quote:
type=identify
will match an IP address and accept it,

well, in my example I can control both sides and able to configure it
without registration. in real life I have a provider that requires
username/password authentication

provider gives me - Username - Password - DomainName

They may require it for *outgoing* calls to them but for incoming I
highly doubt they'd want you to authenticate them. It's usually always
IP authentication.

Quote:
I have configure it like I showed before and have exactly the same
notice

[Jul 16 20:32:23] NOTICE[21926]: res_pjsip/pjsip_distributor.c:246
log_unidentified_request: Request from
'"cb5069"<[url=sip:asterisk@85.195.98.178]sip:asterisk@85.195.98.178[/url]>' failed for
'85.195.98.178:5060' (callid:
173995aa2e25283807700d65055c9214@85.195.98.178 (173995aa2e25283807700d65055c9214@85.195.98.178)) - No matching
endpoint found 85.195.98.178 is an operator,

so what I should add to my config to be able accept calls from
Registered peer ?

The PJSIP functionality does not currently allow using the dynamic IP of a registration to match an incoming call. You either have to explicitly use the identify section or match as I previously described.

Without further details of your setup (IP addresses, who are calling who) and how you want it to work I can't answer.

--
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:
http://lists.digium.com/mailman/listinfo/asterisk-users
Back to top
jleed at me.com
Guest





PostPosted: Thu Jul 17, 2014 2:29 am    Post subject: [asterisk-users] PJSIP outbound register and inbound calls Reply with quote

oh.. its simple.

"[res_pjsip_endpoint_identifier_ip]" should be before "identify=realtime,ps_endpoint_id_ips”, not "[res_pjsip]”


Thanks all for help Smile


On 17 Jul 2014, at 11:05, Nick Awesome <jleed@me.com (jleed@me.com)> wrote:
Quote:
New information, as I said I’m using realtime,thats the problem!


I just tested using static config file and it is working perfect.
after some research I figured out that problem with “ps_endpoint_id_ips" for some reason asterisk ignoring matches in this table,


I have string in sorcery.conf


identify = realtime,ps_endpoint_id_ips


also have string in extconfig.conf


ps_endpoint_id_ips => odbc,asterisk,pbx_endpoint_id_ips


and ofc I have table


CREATE TABLE `pbx_endpoint_id_ips` (
`id` varchar(40) NOT NULL,
`endpoint` varchar(40) DEFAULT NULL,
`match` varchar(80) DEFAULT NULL,
UNIQUE KEY `id` (`id`),
KEY `ps_endpoint_id_ips_id` (`id`)
) ENGINE=InnoDB DEFAULT CHARSET=latin1;



with entry


10001 | 10001 | 85.195.98.178


but thats just didn’t works(


is this a bug and should I open ticket ?


On 16 Jul 2014, at 21:13, Nick Awesome <jleed@me.com (jleed@me.com)> wrote:
Quote:
Ok there is my test account from sipiko.net

username: cb5069
password: sqv664yqtp
domain: callme.sipiko.net


its using username/password authentication.
because its just website widget I need only inbound calls from this peer,
test call can be done from url:
http://callme.sipiko.net/callme.php?id=5069&call_id=210&tunnel=yes


on my side I have an asterisk 12 using pjsip


Have configured IVR with number 5000 on context "dialmap", so I need forward all calls from this provider to number 5000 over "dialmap" context


help if you can please:)


On Jul 16, 2014, at 8:53 PM, Joshua Colp <jcolp@digium.com (jcolp@digium.com)> wrote:
Quote:
Nick Awesome wrote:
Quote:
I thought that
Quote:
Quote:
type=identify
will match an IP address and accept it,

well, in my example I can control both sides and able to configure it
without registration. in real life I have a provider that requires
username/password authentication

provider gives me - Username - Password - DomainName

They may require it for *outgoing* calls to them but for incoming I
highly doubt they'd want you to authenticate them. It's usually always
IP authentication.

Quote:
I have configure it like I showed before and have exactly the same
notice

[Jul 16 20:32:23] NOTICE[21926]: res_pjsip/pjsip_distributor.c:246
log_unidentified_request: Request from
'"cb5069"<[url=sip:asterisk@85.195.98.178]sip:asterisk@85.195.98.178[/url]>' failed for
'85.195.98.178:5060' (callid:
173995aa2e25283807700d65055c9214@85.195.98.178 (173995aa2e25283807700d65055c9214@85.195.98.178)) - No matching
endpoint found 85.195.98.178 is an operator,

so what I should add to my config to be able accept calls from
Registered peer ?

The PJSIP functionality does not currently allow using the dynamic IP of a registration to match an incoming call. You either have to explicitly use the identify section or match as I previously described.

Without further details of your setup (IP addresses, who are calling who) and how you want it to work I can't answer.

--
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:
http://lists.digium.com/mailman/listinfo/asterisk-users




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

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