Asterisk / SER / SIP interconnect

i am trying to connect to a voice provider using SIP. all i am given is a username, password and Ip address. i tried using Linksys ATA and is working fine, next i tried with X-ten and no problem there. however, when i try to dial out using asterisk. i got stuck there.

## Deliverables

[url removed, login to view] [general] bindport = 5060 ; Port to bind to (SIP is 5060) bindaddr = [url removed, login to view] ; Address to bind to (all addresses on machine) disallow=all allow=ulaw allow=alaw srvlookup=yes notifyringing=yes notifyhold=yes useragent=voxcom limitonpeers=yes callerid = Unknown tos=0x68 [tone] username=015xxxxxxx type=user secret=xxxxx qualify=yes port=5060 nat=no host=[url removed, login to view] dtmfmode=rfc2833 canreinvite=yes allow=all call-limit=50 register=015xxxxx:xxxx@[url removed, login to view] following is a sip debug voxcom*CLI> REGISTER 12 headers, 0 lines REGISTER attempt 1 to 015xxx@[url removed, login to view] Reliably Transmitting (no NAT) to [url removed, login to view]: REGISTER sip:[url removed, login to view] SIP/2.0 Via: SIP/2.0/UDP [url removed, login to view];branch=z9hG4bK4e5e407f;rport From: ;tag=as0ea27f72 To: Call-ID: 097da0cd14050caa1e4fea487f2fb3d3@[url removed, login to view] CSeq: 102 REGISTER User-Agent: voxcom Max-Forwards: 70 Expires: 120 Contact: Event: registration C voxcom*CLI> ontent-Length: 0 --- voxcom*CLI> <--- SIP read from [url removed, login to view] ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP [url removed, login to view];branch=z9hG4bK4e5e407f;rport=5060 From: ;tag=as0ea27f72 To: ;tag=[url removed, login to view] Call-ID: 097da0cd14050caa1e4fea487f2fb3d3@[url removed, login to view] CSeq: 102 REGISTER WWW-Authenticate: Digest realm="tone", nonce="47e7510631a4234d47c6d1ab1f5f8d579365cbeb" Server: Sip EXpress router (0.9.6 (i386/linux)) Content-Length: 0 voxcom*CLI> Warning: 392 [url removed, login to view] "Noisy feedback tells: pid=15241 req_src_ip=[url removed, login to view] req_src_port=5060 in_uri=sip:[url removed, login to view] out_uri=sip:[url removed, login to view] via_cnt==1" <-------------> --- (10 headers 0 lines) --- Responding to challenge, registration to domain/host name [url removed, login to view] REGISTER 13 headers, 0 lines REGISTER attempt 2 to 015xxxx@[url removed, login to view] Reliably Transmitting (no NAT) to [url removed, login to view]: REGISTER sip:[url removed, login to view] SIP/2.0 Via: SIP/2.0/UDP [url removed, login to view];branch=z9hG4bK1bf8571c;rport From: ;tag=as7ba15166 To: Call-ID: 097da0cd14050caa1e4fea487f2fb3d3@[url removed, login to view] CSeq: 103 REGISTER User-Agent: voxcom Max-Forwards: 70 Authorization: Digest username="015xxxx", realm="tone", algorithm=MD5, uri="sip:[url removed, login to view]", nonce="47e7510631a4234d47c6d1ab1f5f8d579365cbeb", response="f1f04a3a2f2f1b393878caec82916fc8", opaque="" Expires: 120 Contact: Event: registration Content-Length: 0 --- voxcom*CLI> <--- SIP read from [url removed, login to view] ---> SIP/2.0 200 OK Via: SIP/2.0/UDP [url removed, login to view];branch=z9hG4bK1bf8571c;rport=5060 From: ;tag=as7ba15166 To: ;tag=[url removed, login to view] Call-ID: 097da0cd14050caa1e4fea487f2fb3d3@[url removed, login to view] CSeq: 103 REGISTER Contact: ;expires=130 Server: Sip EXpress router (0.9.6 (i386/linux)) Content-Length: 0 Warning: 392 [url removed, login to view] "Noisy feedback tells: pid=15238 req_src_ip=[url removed, login to view] req_src_port=5060 in_uri=sip:[url removed, login to view] out_uri=sip:[url removed, login to view] via_cnt==1" <-------------> --- (10 headers 0 lines) --- Scheduling destruction of SIP dialog '097da0cd14050caa1e4fea487f2fb3d3@[url removed, login to view]' in 32000 ms (Method: REGISTER) <--- SIP read from [url removed, login to view] ---> SIP/2.0 200 OK Via: SIP/2.0/UDP [url removed, login to view];branch=z9hG4bK1bf8571c;rport=5060 From: ;tag=as7ba15166 To: ;tag=[url removed, login to view] Call-ID: 097da0cd14050caa1e4fea487f2fb3d3@[url removed, login to view] CSeq: 103 REGISTER Contact: ;expires=130 Contact: ;expires=130 Server: Sip EXpress router (0.9.6 (i386/linux)) Content-Length: 0 Warning: 392 [url removed, login to view] "Noisy feedback tells: pid=15238 req_src_ip=[url removed, login to view] req_src_port=5060 in_uri=sip:[url removed, login to view] out_uri=sip:[url removed, login to view] via_cnt==1" <-------------> --- (11 headers 0 lines) --- Really destroying SIP dialog '097da0cd14050caa1e4fea487f2fb3d3@[url removed, login to view]' Method: REGISTER i am no expert, but it looks like i needed registeration or some sort. i did a sip show registry Host Username Refresh State [url removed, login to view] [url removed, login to view] 015xx 115 Registered Mon, 24 it look like is registered and when i do a dial Audio is at [url removed, login to view] port 13780 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to [url removed, login to view]: INVITE sip:0xxx98688@tone SIP/2.0 Via: SIP/2.0/UDP [url removed, login to view];branch=z9hG4bK68ab52ab;rport From: "015xxxx" ;tag=as6635bdd8 To: Contact: Call-ID: 0a6478b949f860083beed39e6844f932@[url removed, login to view] CSeq: 102 INVITE User-Agent: voxcom Max-Forwards: 70 Date: Mon, 24 Mar 2008 06:37:51 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 262 v=0 o=root 2652 2652 IN IP4 [url removed, login to view] s=session c=IN IP4 [url removed, login to view] t=0 0 m=audio 13780 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Retransmitting #1 (no NAT) to [url removed, login to view]: INVITE sip:0xxx98688@tone SIP/2.0 Via: SIP/2.0/UDP [url removed, login to view];branch=z9hG4bK68ab52ab;rport From: "015xxxx" ;tag=as6635bdd8 To: Contact: Call-ID: 0a6478b949f860083beed39e6844f932@[url removed, login to view] CSeq: 102 INVITE User-Agent: voxcom Max-Forwards: 70 Date: Mon, 24 Mar 2008 06:37:51 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 262 v=0 o=root 2652 2652 IN IP4 [url removed, login to view] s=session c=IN IP4 [url removed, login to view] t=0 0 m=audio 13780 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Retransmitting #2 (no NAT) to [url removed, login to view]: INVITE sip:0xxxx8688@tone SIP/2.0 Via: SIP/2.0/UDP [url removed, login to view];branch=z9hG4bK68ab52ab;rport From: "015xxxx" ;tag=as6635bdd8 To: Contact: Call-ID: 0a6478b949f860083beed39e6844f932@[url removed, login to view] CSeq: 102 INVITE User-Agent: voxcom Max-Forwards: 70 Date: Mon, 24 Mar 2008 06:37:51 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 262 v=0 o=root 2652 2652 IN IP4 [url removed, login to view] s=session c=IN IP4 [url removed, login to view] t=0 0 m=audio 13780 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Scheduling destruction of SIP dialog '0a6478b949f860083beed39e6844f932@[url removed, login to view]' in 32000 ms (Method: INVITE) your job is consider done when i can successfully connected and make an outgoing call to the voice provider thanks thank you

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