SIP/2.0 407 Proxy Authentication Required

asmlicense · ‎06-15-2017

Hi all

I have CME having sip trunk (just dial-peer) to GSM gateway and registration to SIP provider. SIP trunk works fine, but I have problem with GSM gateway. No inbound and outbound works. In GSM gateway I see sip trunk is unreachable to CME, but they are in the same network, I can ping to GSM Gateway. I guess there authentication issue, but it could not had to happen, because in CME it is just dial-peer, in GSM gateway sip trunk type is peer-to-peer. What I see from debugs that maybe CME mixed sip registration up and sends authentication parameters to GSM gateway. I Here is debug output(I changed actual number to 0885555555 and public IP of router to 1.1.1.1):

Sent:
INVITE sip:0885555555@192.168.10.22:5060 SIP/2.0
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK2318C7
From: <sip:1006@gvgw.simonics.com>;tag=F0804-1887
To: <sip:0885555555@192.168.10.22>
Date: Fri, 16 Jun 2017 00:03:35 GMT
Call-ID: 1350ABE5-515E11E7-85BDB47B-E8A93C19@1.1.1.1
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 0303934062-1365119463-2242819195-3903405081
User-Agent: Cisco-SIPGateway/IOS-15.6.3.M
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1497571415
Contact: <sip:1006@1.1.1.1:5060>
Expires: 180
Allow-Events: telephone-event
Session-ID: 0001a7b13ec75b1987dabb8d74944200;remote=00000000000000000000000000000000
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 264

Received:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK2318C7;received=1.1.1.1
From: <sip:1006@gvgw.simonics.com>;tag=F0804-1887
To: <sip:0885555555@192.168.10.22>;tag=as63667f52
Call-ID: 1350ABE5-515E11E7-85BDB47B-E8A93C19@1.1.1.1
CSeq: 101 INVITE
Server: TG200V2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1a9ee7eb"
Content-Length: 0

As I understand CME sends 1006@gvgw.simonics.com, instead of 1006@192.168.10.1. That is why GSM gateway rejects call. Is that correct diagnose. If so, how can I prevent CME to send wrong "From" parameter to GSM gateway?

Hamzeh Alzubi · ‎06-16-2017

Hello,

Test call in debugs was placed from Cisco GW (CME) out to TG200V2 (GSM GW supposedly). Invite sent by CME has no authentication parameters, as you expect. Response received from GSM GW asks for challenge on realm "asterisk". So it is GSM GW which tries to authenticate CME, not the other way around. If you can reconfigure GSM GW not to start challenge, initial invite sent by CME should go through. Or else, you need to configure CME with "asterisk" username and Password to properly respond to this challenge with a re-invite.

HTH

Hamzeh

asmlicense · ‎06-16-2017

The problem was with sip media and control interface. I have set it outside interface for sip provider.

After adding source interface to the dial-peer going to gsm gateway, it started to work.

CSCO12245191 · ‎12-09-2020

Hi Asmilicense,

I think I got the same issue. What command solve it?

Roger Kallberg · ‎12-10-2020

You would use the bind control and media command on the dial peer. See this example.

dial-peer voice 100 voip
 description ** Inbound calls from PSTN **
 session protocol sipv2
 incoming uri via PSTN
 voice-class codec 1  
 voice-class sip early-offer forced
 voice-class sip bind control source-interface GigabitEthernet0/0/0
 voice-class sip bind media source-interface GigabitEthernet0/0/0
 dtmf-relay sip-kpml rtp-nte sip-notify
 fax-relay sg3-to-g3
 fax rate 9600
 fax nsf 000000
 fax protocol t38 version 0 ls-redundancy 2 hs-redundancy 1 fallback pass-through g711ulaw
 no vad
!
dial-peer voice 110 voip
 description ** Outbound calls to PSTN **
 translation-profile outgoing PSTN-OUT
 huntstop
 session protocol sipv2
 session server-group 2000
 destination e164-pattern-map 2000
 voice-class codec 1  
 voice-class sip profiles 10
 voice-class sip options-keepalive profile 2000
 voice-class sip bind control source-interface GigabitEthernet0/0/0
 voice-class sip bind media source-interface GigabitEthernet0/0/0
 voice-class sip audio forced
 dtmf-relay sip-kpml rtp-nte sip-notify
 fax-relay sg3-to-g3
 fax rate 9600
 fax nsf 000000
 fax protocol t38 version 0 ls-redundancy 2 hs-redundancy 1 fallback pass-through g711ulaw
 no vad
!