Problem: Asterisk is ignoring INVITE Messages

Hi!

We are experementing with SIP and Asterisk in school.
We had no big problems so far, but now we have a problem with sending an INVITE-Message to another host, because Asterisk ignores them.
And we cannot figure out where this failure comes from.

So, please help us to get this thing done.

Here is the SIP debug:
REGISTER-Part:

Code:

<--- SIP read from 192.168.100.17:5070 ---> REGISTER sip:192.168.100.32 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.17:5070;branch=z9hG4bK-I0qc From: <sip:2000@192.168.100.32>;tag=16cD To: <sip:2000@192.168.100.32> Call-ID: I0qc [!at] 192.168.100.17 (replace the [!at] with a @) CSeq: 2 REGISTER User-Agent: TwinSIP Communicator Contact: <sip:2000@192.168.100.17:5070> Expires: 3600 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.100.17 : 5070 (no NAT) <--- Transmitting (no NAT) to 192.168.100.17:5070 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.100.17:5070;branch=z9hG4bK-I0qc;received=192.168.100.17 From: <sip:2000@192.168.100.32>;tag=16cD To: <sip:2000@192.168.100.32> Call-ID: I0qc [!at] 192.168.100.17 (replace the [!at] with a @) CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:2000@192.168.100.32> Content-Length: 0 <------------> <--- Transmitting (no NAT) to 192.168.100.17:5070 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.100.17:5070;branch=z9hG4bK-I0qc;received=192.168.100.17 From: <sip:2000@192.168.100.32>;tag=16cD To: <sip:2000@192.168.100.32>;tag=as0d8811c3 Call-ID: I0qc [!at] 192.168.100.17 (replace the [!at] with a @) CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="12105d19" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'I0qc@192.168.100.17' in 32000 ms (Method: REGISTER) <--- SIP read from 192.168.100.17:5070 ---> REGISTER sip:192.168.100.32 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.17:5070;maddr=;ttl=;received=;branch=z9hG4bK6mvpR;rport= From: <sip:2000@192.168.100.32>;tag=16cD To: <sip:2000@192.168.100.32> Call-ID: I0qc [!at] 192.168.100.17 (replace the [!at] with a @) CSeq: 2 REGISTER User-Agent: TwinSIP Communicator Contact: <sip:2000@192.168.100.17:5070>;q=;expires= Expires: 3600 Content-Length: 0 Authorization: Digest username="2000", realm="asterisk", nonce="12105d19", uri="192.168.100.32", response="474418e865cc40a226cd6d54820f0bce", algorithm=MD5 <-------------> --- (11 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.100.17 : 5070 (no NAT) <--- Transmitting (no NAT) to 192.168.100.17:5070 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.100.17:5070;maddr=;ttl=;received=;branch=z9hG4bK6mvpR;rport=;received=192.168.100.17 From: <sip:2000@192.168.100.32>;tag=16cD To: <sip:2000@192.168.100.32> Call-ID: I0qc [!at] 192.168.100.17 (replace the [!at] with a @) CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:2000@192.168.100.32> Content-Length: 0 <------------>     -- Saved useragent "TwinSIP Communicator" for peer 2000 <--- Transmitting (no NAT) to 192.168.100.17:5070 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.100.17:5070;maddr=;ttl=;received=;branch=z9hG4bK6mvpR;rport=;received=192.168.100.17 From: <sip:2000@192.168.100.32>;tag=16cD To: <sip:2000@192.168.100.32>;tag=as0d8811c3 Call-ID: I0qc [!at] 192.168.100.17 (replace the [!at] with a @) CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 3600 Contact: <sip:2000@192.168.100.17:5070>;expires=3600 Date: Thu, 24 Apr 2008 15:29:15 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'I0qc@192.168.100.17' in 32000 ms (Method: REGISTER) Really destroying SIP dialog 'I0qc@192.168.100.17' Method: REGISTER <--- SIP read from 192.168.100.50:5070 ---> REGISTER sip:192.168.100.32 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.50:5070;branch=z9hG4bK-U$ÖO From: <sip:2001@192.168.100.32>;tag=YilY To: <sip:2001@192.168.100.32> Call-ID: U$ÖO@192.168.100.50 CSeq: 1 REGISTER User-Agent: TwinSIP Communicator Contact: <sip:2001@192.168.100.50:5070> Expires: 3600 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.100.50 : 5070 (no NAT) <--- Transmitting (no NAT) to 192.168.100.50:5070 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.100.50:5070;branch=z9hG4bK-U$ÖO;received=192.168.100.50 From: <sip:2001@192.168.100.32>;tag=YilY To: <sip:2001@192.168.100.32> Call-ID: U$ÖO@192.168.100.50 CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:2001@192.168.100.32> Content-Length: 0 <------------> <--- Transmitting (no NAT) to 192.168.100.50:5070 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.100.50:5070;branch=z9hG4bK-U$ÖO;received=192.168.100.50 From: <sip:2001@192.168.100.32>;tag=YilY To: <sip:2001@192.168.100.32>;tag=as77afec68 Call-ID: U$ÖO@192.168.100.50 CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="06c7a8b6" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'U$ÖO@192.168.100.50' in 32000 ms (Method: REGISTER) <--- SIP read from 192.168.100.50:5070 ---> REGISTER sip:192.168.100.32 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.50:5070;maddr=;ttl=;received=;branch=z9hG4bKß$fyT;rport= From: <sip:2001@192.168.100.32>;tag=YilY To: <sip:2001@192.168.100.32> Call-ID: U$ÖO@192.168.100.50 CSeq: 1 REGISTER User-Agent: TwinSIP Communicator Contact: <sip:2001@192.168.100.50:5070>;q=;expires= Expires: 3600 Content-Length: 0 Authorization: Digest username="2001", realm="asterisk", nonce="06c7a8b6", uri="192.168.100.32", response="0097b6ddf1cec8fc636ccb938dcb0e38", algorithm=MD5 <-------------> --- (11 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.100.50 : 5070 (no NAT) <--- Transmitting (no NAT) to 192.168.100.50:5070 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.100.50:5070;maddr=;ttl=;received=;branch=z9hG4bKß$fyT;rport=;received=192.168.100.50 From: <sip:2001@192.168.100.32>;tag=YilY To: <sip:2001@192.168.100.32> Call-ID: U$ÖO@192.168.100.50 CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: <sip:2001@192.168.100.32> Content-Length: 0 <------------>     -- Saved useragent "TwinSIP Communicator" for peer 2001 <--- Transmitting (no NAT) to 192.168.100.50:5070 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.100.50:5070;maddr=;ttl=;received=;branch=z9hG4bKß$fyT;rport=;received=192.168.100.50 From: <sip:2001@192.168.100.32>;tag=YilY To: <sip:2001@192.168.100.32>;tag=as77afec68 Call-ID: U$ÖO@192.168.100.50 CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 3600 Contact: <sip:2001@192.168.100.50:5070>;expires=3600 Date: Thu, 24 Apr 2008 15:29:51 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'U$ÖO@192.168.100.50' in 32000 ms (Method: REGISTER) Really destroying SIP dialog 'U$ÖO@192.168.100.50' Method: REGISTER

INVITE-Part:

Quote:

<--- SIP read from 192.168.100.50:5070 ---> INVITE sip:2000@192.168.100.32 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.50:5070;branch=z9hG4bK-eE6hF;rport= From: <sip:2001@192.168.100.32> To: <sip:2000@192.168.100.32> Call-ID: eE6h [!at] 192.168.100.32 (replace the [!at] with a @) CSeq: 3 INVITE Contact: <sip:2001@192.168.100.50> User-Agent: TwinSIP Communicator Content-Type: application/sdp Content-Length: 99 v=0 o=andy 2634 2634 IN IP4 192.168.100.50 s=test von alex t=0 0 m=audio 9050 udp PCM/32000/8/1 <-------------> --- (10 headers 5 lines) --- Sending to 192.168.100.50 : 5070 (no NAT) Using INVITE request as basis request - eE6h [!at] 192.168.100.32 (replace the [!at] with a @) <--- Reliably Transmitting (no NAT) to 192.168.100.50:5070 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.100.50:5070;branch=z9hG4bK-eE6hF;rport=;received=192.168.100.50 From: <sip:2001@192.168.100.32> To: <sip:2000@192.168.100.32>;tag=as3887b859 Call-ID: eE6h [!at] 192.168.100.32 (replace the [!at] with a @) CSeq: 3 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4e5f93f7" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'eE6h@192.168.100.32' in 32000 ms (Method: INVITE) Found user '2001' <--- SIP read from 192.168.100.50:5070 ---> INVITE sip:2000@192.168.100.32 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.50:5070;branch=z9hG4bK-eE6hF;rport= From: <sip:2001@192.168.100.32> To: <sip:2000@192.168.100.32>;tag=as3887b859 Call-ID: eE6h [!at] 192.168.100.32 (replace the [!at] with a @) CSeq: 3 INVITE Contact: <sip:2001@192.168.100.50> User-Agent: TwinSIP Communicator Content-Type: application/sdp Content-Length: 99 Proxy-Authorization: Digest username="2001", realm="asterisk", nonce="4e5f93f7", uri="192.168.100.32", response="39abd8d49d07c1cbb489b0fc467d2856", algorithm=MD5 <-------------> --- (11 headers 0 lines) --- Ignoring this INVITE request Retransmitting #1 (no NAT) to 192.168.100.50:5070: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.100.50:5070;branch=z9hG4bK-eE6hF;rport=;received=192.168.100.50 From: <sip:2001@192.168.100.32> To: <sip:2000@192.168.100.32>;tag=as3887b859 Call-ID: eE6h [!at] 192.168.100.32 (replace the [!at] with a @) CSeq: 3 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4e5f93f7" Content-Length: 0 --- <--- SIP read from 192.168.100.50:5070 ---> INVITE sip:2000@192.168.100.32 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.50:5070;branch=z9hG4bK-eE6hF;rport= From: <sip:2001@192.168.100.32> To: <sip:2000@192.168.100.32>;tag=as3887b859 Call-ID: eE6h [!at] 192.168.100.32 (replace the [!at] with a @) CSeq: 3 INVITE Contact: <sip:2001@192.168.100.50> User-Agent: TwinSIP Communicator Content-Type: application/sdp Content-Length: 99 Proxy-Authorization: Digest username="2001", realm="asterisk", nonce="4e5f93f7", uri="192.168.100.32", response="39abd8d49d07c1cbb489b0fc467d2856", algorithm=MD5 <-------------> --- (11 headers 0 lines) --- Ignoring this INVITE request Retransmitting #2 (no NAT) to 192.168.100.50:5070: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.100.50:5070;branch=z9hG4bK-eE6hF;rport=;received=192.168.100.50 From: <sip:2001@192.168.100.32> To: <sip:2000@192.168.100.32>;tag=as3887b859 Call-ID: eE6h [!at] 192.168.100.32 (replace the [!at] with a @) CSeq: 3 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4e5f93f7" Content-Length: 0 --- <--- SIP read from 192.168.100.50:5070 ---> INVITE sip:2000@192.168.100.32 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.50:5070;branch=z9hG4bK-eE6hF;rport= From: <sip:2001@192.168.100.32> To: <sip:2000@192.168.100.32>;tag=as3887b859 Call-ID: eE6h [!at] 192.168.100.32 (replace the [!at] with a @) CSeq: 3 INVITE Contact: <sip:2001@192.168.100.50> User-Agent: TwinSIP Communicator Content-Type: application/sdp Content-Length: 99 Proxy-Authorization: Digest username="2001", realm="asterisk", nonce="4e5f93f7", uri="192.168.100.32", response="39abd8d49d07c1cbb489b0fc467d2856", algorithm=MD5 <-------------> --- (11 headers 0 lines) --- Ignoring this INVITE request Retransmitting #3 (no NAT) to 192.168.100.50:5070: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.100.50:5070;branch=z9hG4bK-eE6hF;rport=;received=192.168.100.50 From: <sip:2001@192.168.100.32> To: <sip:2000@192.168.100.32>;tag=as3887b859 Call-ID: eE6h [!at] 192.168.100.32 (replace the [!at] with a @) CSeq: 3 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4e5f93f7" Content-Length: 0 --- <--- SIP read from 192.168.100.50:5070 ---> INVITE sip:2000@192.168.100.32 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.50:5070;branch=z9hG4bK-eE6hF;rport= From: <sip:2001@192.168.100.32> To: <sip:2000@192.168.100.32>;tag=as3887b859 Call-ID: eE6h [!at] 192.168.100.32 (replace the [!at] with a @) CSeq: 3 INVITE Contact: <sip:2001@192.168.100.50> User-Agent: TwinSIP Communicator Content-Type: application/sdp Content-Length: 99 Proxy-Authorization: Digest username="2001", realm="asterisk", nonce="4e5f93f7", uri="192.168.100.32", response="39abd8d49d07c1cbb489b0fc467d2856", algorithm=MD5 <-------------> --- (11 headers 0 lines) --- Ignoring this INVITE request Retransmitting #4 (no NAT) to 192.168.100.50:5070: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.100.50:5070;branch=z9hG4bK-eE6hF;rport=;received=192.168.100.50 From: <sip:2001@192.168.100.32> To: <sip:2000@192.168.100.32>;tag=as3887b859 Call-ID: eE6h [!at] 192.168.100.32 (replace the [!at] with a @) CSeq: 3 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4e5f93f7" Content-Length: 0 --- <--- SIP read from 192.168.100.50:5070 ---> INVITE sip:2000@192.168.100.32 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.50:5070;branch=z9hG4bK-eE6hF;rport= From: <sip:2001@192.168.100.32> To: <sip:2000@192.168.100.32>;tag=as3887b859 Call-ID: eE6h [!at] 192.168.100.32 (replace the [!at] with a @) CSeq: 3 INVITE Contact: <sip:2001@192.168.100.50> User-Agent: TwinSIP Communicator Content-Type: application/sdp Content-Length: 99 Proxy-Authorization: Digest username="2001", realm="asterisk", nonce="4e5f93f7", uri="192.168.100.32", response="39abd8d49d07c1cbb489b0fc467d2856", algorithm=MD5 <-------------> --- (11 headers 0 lines) --- Ignoring this INVITE request Retransmitting #5 (no NAT) to 192.168.100.50:5070: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.100.50:5070;branch=z9hG4bK-eE6hF;rport=;received=192.168.100.50 From: <sip:2001@192.168.100.32> To: <sip:2000@192.168.100.32>;tag=as3887b859 Call-ID: eE6h [!at] 192.168.100.32 (replace the [!at] with a @) CSeq: 3 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4e5f93f7" Content-Length: 0 --- <--- SIP read from 192.168.100.50:5070 ---> INVITE sip:2000@192.168.100.32 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.50:5070;branch=z9hG4bK-eE6hF;rport= From: <sip:2001@192.168.100.32> To: <sip:2000@192.168.100.32>;tag=as3887b859 Call-ID: eE6h [!at] 192.168.100.32 (replace the [!at] with a @) CSeq: 3 INVITE Contact: <sip:2001@192.168.100.50> User-Agent: TwinSIP Communicator Content-Type: application/sdp Content-Length: 99 Proxy-Authorization: Digest username="2001", realm="asterisk", nonce="4e5f93f7", uri="192.168.100.32", response="39abd8d49d07c1cbb489b0fc467d2856", algorithm=MD5 <-------------> --- (11 headers 0 lines) --- Ignoring this INVITE request Retransmitting #6 (no NAT) to 192.168.100.50:5070: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.100.50:5070;branch=z9hG4bK-eE6hF;rport=;received=192.168.100.50 From: <sip:2001@192.168.100.32> To: <sip:2000@192.168.100.32>;tag=as3887b859 Call-ID: eE6h [!at] 192.168.100.32 (replace the [!at] with a @) CSeq: 3 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4e5f93f7" Content-Length: 0 --- <--- SIP read from 192.168.100.50:5070 ---> INVITE sip:2000@192.168.100.32 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.50:5070;branch=z9hG4bK-eE6hF;rport= From: <sip:2001@192.168.100.32> To: <sip:2000@192.168.100.32>;tag=as3887b859 Call-ID: eE6h [!at] 192.168.100.32 (replace the [!at] with a @) CSeq: 3 INVITE Contact: <sip:2001@192.168.100.50> User-Agent: TwinSIP Communicator Content-Type: application/sdp Content-Length: 99 Proxy-Authorization: Digest username="2001", realm="asterisk", nonce="4e5f93f7", uri="192.168.100.32", response="39abd8d49d07c1cbb489b0fc467d2856", algorithm=MD5 <-------------> --- (11 headers 0 lines) --- Ignoring this INVITE request [Apr 24 17:31:57] WARNING[5391]: chan_sip.c:1939 retrans_pkt: Maximum retries exceeded on transmission eE6h [!at] 192.168.100.32 (replace the [!at] with a @) for seqno 3 (Non-critical Response) Really destroying SIP dialog 'eE6h@192.168.100.32' Method: INVITE

Is anything wrong with the SIP-Messages? Or what can we do to solve this problem?

Thanks in advance,
Alex

Its not ignoring the message.. its responding with a security challenge.

It looks like the user 2000 is registering successfully (from the REGISTER trace you posted), but the call attempts are being made by another user; 2001, whom is trying to contact the registered user 2000, but cannot correctly authenticate with the asterisk server.

Check the sip.conf profile for the user 2001, and make sure the softphone or whatever is issuing those INVITE's is configured with the correct username and password as is shown in the sip.conf for user 2001.

@ x-console :
Thanks for your answer. But the problem is still there.

Quote:

Its not ignoring the message.. its responding with a security challenge.

Yes, thats true.
And when my application recieves the security challenge it sends back the invite including the proxy-authorization header. But i get the same challenge again and again.
I think, that if the response was wrong, asterisk should send an error- or not authorized-message instead of sending the challange again.
So I wonder why in the sip-debug stands, that the Invite request has been ignored.

Quote:

It looks like the user 2000 is registering successfully (from the REGISTER trace you posted), but the call attempts are being made by another user; 2001, whom is trying to contact the registered user 2000, but cannot correctly authenticate with the asterisk server.

Both users are registered successfully.

Quote:

Check the sip.conf profile for the user 2001, and make sure the softphone or whatever is issuing those INVITE's is configured with the correct username and password as is shown in the sip.conf for user 2001.

The data is correct.

Quote:

I think, that if the response was wrong, asterisk should send an error- or not authorized-message instead of sending the challange again.

thats incorrect. It should resend the challenge for each INVITE that does not contain the correct credentials. There is something not right with the authorization header on the INVITE which is why asterisk is not accepting it.

Quote:

Quote: I think, that if the response was wrong, asterisk should send an error- or not authorized-message instead of sending the challange again. thats incorrect. It should resend the challenge for each INVITE that does not contain the correct credentials. There is something not right with the authorization header on the INVITE which is why asterisk is not accepting it.

Ok. I am increasing the CSeq-Number now, every time a message is sent. But now I recieve a 491 Request Pending-Message and the challenge again.
So it's still not working.

erm, a 491 is sent as a response when there is already an open dialog that the latest request wishes to modify. Typical example is re-INVITE's corssing on the wire. From your comment re: cseq, i'm guessing you are building your own sip client rather than using an existing one known to work? Check your generation/inclusion of to and from tags, and the call-id, all of which are used to uniquely identify a dialog.

incidentally, the behvaiour the client _should_ exhibit when receiving a valid 491 is to delay the re-transmission of the message by some random period (read the rfc for more info), and then re-transmit.. assuming a message regarding the conflicted transaction has not already been received.

Hi

What does your sip.conf look like as I see that you are using port 5070 not 5060 have you reflected this in the sip.conf?

Also do normal softphone calls work to asterisk ?

Ian

I've solved the problem.
I forgot to increase the CSeq-Numbers of the INVITE-Methods.
Now it is working.

Thank you for your help.