# Failed to authenticate on INVITE - SIP/2.0 403 Forbidden **URL:** \ **Category:** Asterisk Support\ **Created:** [June 16, 2014, 12:04pm UTC](https://community.asterisk.org/t/failed-to-authenticate-on-invite-sip-2-0-403-forbidden/46416 "2014-06-16T12:04:10Z")\ **Posts on this page:** 1\ **Page:** 1
**Author:** ![jilirio](https://avatars.discourse-cdn.com/v4/letter/j/919ad9/32.png) [@jilirio](https://community.asterisk.org/u/jilirio)\ **Post date:** [June 16, 2014, 12:04pm UTC](https://community.asterisk.org/t/failed-to-authenticate-on-invite-sip-2-0-403-forbidden/46416/1 "2014-06-16T12:04:10Z")
I have two telephony servers running on the same subnet, TS1 (172.19.57.42/255.255.255.252.0) and TS2(172.19.57.44/255.255.255.252.0) Asterisk running and a SIP trunk (totelephony) between them. There are no problems when dialling SIP extensions belonging to the same TS but I cannot call SIP extensions from TS2 to TS1 or in the other way around. During this test I used: 4755 extension registered in TS1 and 4811 registered in TS2 and tried to make a call from 4811 to 4755, here is the outcome: On both sites, TS1 and TS2 in SIP.conf: [totelephony] type=peer host=172.19.57.44 -\> in TS1 host=172.19.57.42 -\> in TS2 context=dialler disallow=all allow=alaw canreinvite=no insecure=port,invite On TS1 in sip.conf [4755] type=friend secret=4811 port=5060 host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no disallow=all allow=alaw On TS2 in sip.conf: [4811] type=friend secret=4811 port=5060 host=dynamic dtmfmode=rfc2833 context=from-internal canreinvite=no disallow=all allow=alaw On both sites, TS1 and TS2, in extensions.conf [from-internal] exten =\> \_554XXX,1,Dial(SIP/totelephony/${EXTEN:2}) exten =\> \_554XXX,2,Hangup() exten =\> \_4XXX,1,Dial(SIP/${EXTEN}) exten =\> \_4XXX,2,Hangup() SIP logs: [Jun 16 11:53:11] VERBOSE[13585] logger.c: \<— SIP read from 172.19.41.195:5060 —\> INVITE [sip:554755@172.19.57.44](mailto:sip:554755@172.19.57.44);transport=UDP SIP/2.0 Via: SIP/2.0/UDP 172.19.41.195:5060;branch=z9hG4bK-d8754z-2463e75d7bf86691-1—d8754z-;rport Max-Forwards: 70 Contact: sip:4811@172.19.41.195:5060;transport=UDP To: sip:554755@172.19.57.44;transport=UDP From: sip:4811@172.19.57.44;transport=UDP;tag=1620a86d Call-ID: YjQ3OTI4MjIxNmQ5NDA3MmIzZmQ4YzjU. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO Content-Type: application/sdp User-Agent: Zoiper for Windows rev.2875 Content-Length: 208 [Jun 16 11:53:11] VERBOSE[13585] logger.c: \<— Reliably Transmitting (no NAT) to 172.19.41.195:5060 —\> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 172.19.41.195:5060;branch=z9hG4bK-d8754z-2463e75d7bf86691-1—d8754z-;received=172.19.41.195;rport=5060 From: sip:4811@172.19.57.44;transport=UDP;tag=1620a86d To: sip:554755@172.19.57.44;transport=UDP;tag=as65fb9e7d Call-ID: YjQ3OTI4MjIxNmQ5NDA3MmIzZUzZjU. CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="4e769d22" Content-Length: 0 [Jun 16 11:53:11] VERBOSE[13585] logger.c: \<— SIP read from 172.19.41.195:5060 —\> ACK [sip:554755@172.19.57.44](mailto:sip:554755@172.19.57.44);transport=UDP SIP/2.0 Via: SIP/2.0/UDP 172.19.41.195:5060;branch=z9hG4bK-d8754z-275d7bf86691-1—d8754z-;rport To: sip:554755@172.19.57.44;transport=UDP;tag=as65f9e7d From: sip:4811@172.19.57.44;transport=UDP;tag=162a86d Call-ID: YjQ3OTI4MjIxNmQ5NDA3MmIzZmQ4YzMxMjYzMGUzZjU. CSeq: 1 ACK Content-Length: 0 [Jun 16 11:53:11] VERBOSE[13585] logger.c: \<— SIP read from 172.19.41.195:5060 —\> INVITE [sip:554755@172.19.57.44](mailto:sip:554755@172.19.57.44);transport=UDP SIP/2.0 Via: SIP/2.0/UDP 172.19.41.195:5060;branch=z9hG4bK-d8754z-96682b7d500ec-1—d8754z-;rport Max-Forwards: 70 Contact: sip:4811@172.19.41.195:5060;transport=UDP To: sip:554755@172.19.57.44;transport=UDP From: sip:4811@172.19.57.44;transport=UDP;tag=1620a86d Call-ID: YjQ3OTI4MjIxNmQ5NDA3MmIzZmQ4YzMxMjYzMGUzZjU. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO Content-Type: application/sdp Proxy-Authorization: Digest [username=“4811”,realm=“asterisk”,nonce=“4e769d22”,uri="sip:554755@172.19.57.44](mailto:username=%224811%22,realm=%22asterisk%22,nonce=%224e769d22%22,uri=%22sip:554755@172.19.57.44);transport=UDP",response=“d203e3bdedc3a37f78ffddd6eb9a225e”,algorithm=MD5 User-Agent: Zoiper for Windows rev.2875 Content-Length: 208 [Jun 16 11:53:11] VERBOSE[13585] logger.c: Looking for 554755 in from-internal (domain 172.19.57.44) [Jun 16 11:53:11] VERBOSE[13585] logger.c: list\_route: hop: sip:4811@172.19.41.195:5060;transport=UDP [Jun 16 11:53:11] VERBOSE[13585] logger.c: \<— Transmitting (NAT) to 172.19.41.195:5060 —\> SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.19.41.195:5060;branch=z9hG4bK-d8754z-96682b32f7d500ec-1—d8754z-;received=172.19.41.195;rport=5060 From: sip:4811@172.19.57.44;transport=UDP;tag=1620a86d To: sip:554755@172.19.57.44;transport=UDP Call-ID: YjQ3OTI4MjIxNmQ5NDA3MmIzZmQ4YzMzMGUzZjU. CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: sip:554755@172.19.57.44 Content-Length: 0 [Jun 16 11:53:11] VERBOSE[19077] logger.c: – Executing [554755@from-internal:1] Dial(“SIP/4811-00000f40”, “SIP/totelephony/4755”) in new stack [Jun 16 11:53:11] VERBOSE[19077] logger.c: Audio is at 172.19.57.44 port 14942 [Jun 16 11:53:11] VERBOSE[19077] logger.c: Adding codec 0x8 (alaw) to SDP [Jun 16 11:53:11] VERBOSE[19077] logger.c: Adding non-codec 0x1 (telephone-event) to SDP [Jun 16 11:53:11] VERBOSE[19077] logger.c: Reliably Transmitting (no NAT) to 172.19.57.42:5060: INVITE [sip:4755@172.19.57.42](mailto:sip:4755@172.19.57.42) SIP/2.0 Via: SIP/2.0/UDP 172.19.57.44:5060;branch=z9hG4bK4fe655ba;rport From: “device” sip:4811@172.19.57.44;tag=as54fef7f4 To: sip:4755@172.19.57.42 Contact: sip:4811@172.19.57.44 Call-ID: [1609872a73662819562bafc20875be5a@172.19.57.44](mailto:1609872a73662819562bafc20875be5a@172.19.57.44) CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 16 Jun 2014 09:53:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Type: application/sdp Content-Length: 213 [Jun 16 11:53:11] VERBOSE[13585] logger.c: \<— SIP read from 172.19.57.42:5060 —\> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 172.19.57.44:5060;branch=z9hG4bK4fe655ba;received=172.19.57.44;rport=5060 From: “device” sip:4811@172.19.57.44;tag=as54fef7f4 To: sip:4755@172.19.57.42;tag=as1c71ac32 Call-ID: [1609872a73662819562bafc20875be5a@172.19.57.44](mailto:1609872a73662819562bafc20875be5a@172.19.57.44) CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="290a924f" Content-Length: 0 \<-------------\> [Jun 16 11:53:11] VERBOSE[13585] logger.c: — (11 headers 0 lines) — [Jun 16 11:53:11] VERBOSE[13585] logger.c: Transmitting (no NAT) to 172.19.57.42:5060: ACK [sip:4755@172.19.57.42](mailto:sip:4755@172.19.57.42) SIP/2.0 Via: SIP/2.0/UDP 172.19.57.44:5060;branch=z9hG4bK4fe655ba;rport From: “device” sip:4811@172.19.57.44;tag=as54fef7f4 To: sip:4755@172.19.57.42;tag=as1c71ac32 Contact: sip:4811@172.19.57.44 Call-ID: [1609872a73662819562bafc20875be5a@172.19.57.44](mailto:1609872a73662819562bafc20875be5a@172.19.57.44) CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 * * * [Jun 16 11:53:11] NOTICE[13585] chan\_sip.c: Failed to authenticate on INVITE to ‘“device” sip:4811@172.19.57.44;tag=as54fef7f4’ [Jun 16 11:53:11] VERBOSE[19077] logger.c: – SIP/totelephony-00000f41 is circuit-busy [Jun 16 11:53:11] VERBOSE[19077] logger.c: == Everyone is busy/congested at this time (1:0/1/0) [Jun 16 11:53:11] VERBOSE[19077] logger.c: – Executing [554755@from-internal:2] Hangup(“SIP/4811-00000f40”, “”) in new stack [Jun 16 11:53:11] VERBOSE[13585] logger.c: Really destroying SIP dialog ‘1609872a73662819562bafc20875be5a@172.19.57.44’ Method: INVITE [Jun 16 11:53:11] VERBOSE[19077] logger.c: == Spawn extension (from-internal, 554755, 2) exited non-zero on ‘SIP/4811-00000f40’ [Jun 16 11:53:11] VERBOSE[19077] logger.c: Scheduling destruction of SIP dialog ‘YjQ3OTI4MjIxNmQ5NDA3MmIzZmQ4YzYzMGUzZjU.’ in 32000 ms (Method: INVITE) [Jun 16 11:53:11] VERBOSE[19077] logger.c: \<— Reliably Transmitting (NAT) to 172.19.41.195:5060 —\> SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 172.19.41.195:5060;branch=z9hG4bK-d8754z-96682b32f7d500ec-1—d8754z-;received=172.19.41.195;rport=5060 From: sip:4811@172.19.57.44;transport=UDP;tag=1620a86d To: sip:554755@172.19.57.44;transport=UDP;tag=as7111b7f7 Call-ID: YjQ3OTI4MjIxNmQ5NDA3ZmQ4YzMxMjYzMGUzZjU. CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 I know I am missing something but I cannot find what and right now I am a bit loss. Any help would be greatly appreciated. Thanks.