25 Nov
2004
25 Nov
'04
7:47 p.m.
I am not sure I understand your description and the problem. If SER
gets the INVITE then registration with the PSTN gateway works (I think
that was the original problem).
Jan.
On 22-11 09:58, E. Versaevel wrote:
I have to authenticate on each outgoing call, so I've setup SER+Asterisk,
however I still have some problems.
I already asked this on the asterisk mailing list, but I'm not getting any
replies there.
Ser forwards outgoing calls to asterisk, which authenticates itself to the
carrier proxy, so far so good. However, asterisk has to forward incoming
calls from the carrier to SER for routing to the UA's, however, asterisk
alters the from header in the sip messages, this way when someone adds the
incoming call to the address book, the sip uri is wrong and the call won't
get anywere.
Take a look at this sip msg:
INVITE sip:erik@localphone:5061 SIP/2.0
Max-Forwards: 10
Record-Route: <sip:3400009521@ser.box;ftag=as3f718642;lr=on>
Via: SIP/2.0/UDP ser.box;branch=z9hG4bK93dc.71ad80b3.0
Via: SIP/2.0/UDP ser.box:5065;branch=z9hG4bK513b584d
From: "3400009525" <sip:asterisk@ser.box:5065>;tag=as3f718642
To: <sip:3400009521@sip.ser.box>
Contact: <sip:asterisk@ser.box:5065>
Call-ID: 533cb84a48058ebb71fbd7bf7557c0f0(a)ser.box
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Thu, 18 Nov 2004 12:46:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 220
P-hint: USRLOC
As you can see the from user is not correct, this should be
3400009525(a)iptel.org. If a user adds this entry to a phonebook, the contact
info will be wrong.
For outgoing calls this won't be a problem as I only send calls to the pstn
thru asterisk, incoming is a bit troublesome.
-----Oorspronkelijk bericht-----
Van: Jan Janak [mailto:jan@iptel.org]
Verzonden: zondag 14 november 2004 13:14
Aan: E. Versaevel
CC: serusers(a)lists.iptel.org
Onderwerp: Re: [Serusers] SER Proxy auth
If you are using asterisk just for the purpose of registering with the
gateway then you could also consider using sipsak for that (that should
be simpler).
sipsak can send the registration and you can configured it to put the IP
address of your proxy server into the Contact.
Jan.
On 12-11 15:03, E. Versaevel wrote:
Hello,
I'm currently trying to setup a SIP environment for VoIP calling for my
final school project, so I'm just working with VoIP/SIP for 2 weeks.
I'm using SER as a SIP proxy server, but the carrier/gateway I am using
for
calling to/from PSTN is requiring me to register
at their server and
authorize outgoing calls, which is something SER won't do. So I got the
idea
to use asterisk between the PSTN carrier and SER
for the authorization,
since Asterisk can register and auth itself.
SIP --------- SIP --------- SIP --------- PSTN
-----| |-----------| |-----------| |------
--------- --------- ---------
SER Asterisk Carrier
<-- auth stuff -->
<-- sip relay -->
Has anyone here ever tried a similar setup or is this an impossibility ?
Kind regards,
E. Versaevel
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