anyone gotten this to register? I simply can’t get incoming calls. I can make outbound calls but incoming goes to teh babytel voicemail like it was not associated with any device. 
they don’t really support asterisk, i followed their documentation for setting it up but to no avail…
I did a sip debug, when I call in it still just drops me to the voice mail or says"this number is not valid, please try again later" this is the whole sip debug and at the end it hangs me up…
HELP if anyone can see whats up?
asterisk1*CLI> sip debug ip sip.babytel.ca
SIP Debugging Enabled for IP: 216.18.125.7
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 216.18.125.7:5065:
REGISTER sip:sip.babytel.ca SIP/2.0
Via: SIP/2.0/UDP 192.168.1.136:5060;branch=z9hG4bK0a6dbfaf;rport
From: sip:1xxxxxxxxxx@sip.babytel.ca;tag=as01875097
To: sip:1xxxxxxxxxx@sip.babytel.ca
Call-ID: 7b771e232f12938c72e8eed915707ea1@127.0.0.1
CSeq: 109 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username=“1xxxxxxxxxx”, realm=“sip.babytel.ca”, algorithm=MD5, uri=“sip:sip.babytel.ca”, nonce=“4596b97cde9570e3b725a1ce082edd2568d7a3ba”, response=“2d84d30409563592cdb7ae1180788a4f”, opaque=""
Expires: 120
Contact: sip:200@192.168.1.136
Event: registration
Content-Length: 0
asterisk1*CLI>
<-- SIP read from 216.18.125.7:5065:
SIP/2.0 200 OK
To: sip:1xxxxxxxxxx@sip.babytel.ca;tag=8456388d4087aadf9b60d2ce631e1e1a.2f3e
From: sip:1xxxxxxxxxx@sip.babytel.ca;tag=as01875097
Via: SIP/2.0/UDP 192.168.1.136:5060;branch=z9hG4bK0a6dbfaf
Call-ID: 7b771e232f12938c72e8eed915707ea1@127.0.0.1
CSeq: 109 REGISTER
Contact: sip:200@192.168.1.136;expires=51
Date: Sat, 30 Dec 2006 19:06:30 GMT
Content-Length: 0
— (9 headers 0 lines) —
Scheduling destruction of call ‘7b771e232f12938c72e8eed915707ea1@127.0.0.1’ in 32000 ms
asterisk1*CLI>
<-- SIP read from 216.18.125.7:5065:
INVITE sip:200@192.168.1.136 SIP/2.0
To: sip:1xxxxxxxxxx@216.18.125.3;user=phone
From: "NICOLAS DE CRIS"sip:6134444444@sip.babytel.ca;tag=d177c741
Via: SIP/2.0/UDP 216.18.125.7:5065;branch=z9hG4bK-d87543-aaba810e1b86374225fd-1-cHBmNmM3YmU2YjM3YWE3MDM2NDA4NA…-d87543-
Call-ID: 0b69709d405e09d8a4e17ab6f209f97f
CSeq: 483011905 INVITE
Contact: sip:A21G9nPtOoPN7pp1Kxtl_eVBVi_g2Iy@216.18.125.7:5065
Max-Forwards: 68
Allow: REGISTER, OPTIONS, INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, INFO, SUBSCRIBE
Content-Type: application/sdp
Content-Length: 352
v=0
o=AudiocodesGW 1629571664 1513447177 IN IP4 216.18.125.7
s=Phone-Call
c=IN IP4 216.18.125.7
t=0 0
m=audio 11766 RTP/AVP 18 0 4 2 101
a=fmtp:18 annexb=no
a=fmtp:4 annexa=no
a=fmtp:101 0-15
a=ptime:20
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
— (11 headers 16 lines) —
Using INVITE request as basis request - 0b69709d405e09d8a4e17ab6f209f97f
Sending to 216.18.125.7 : 5065 (non-NAT)
Found peer 'babytel-out’
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 4
Found RTP audio format 2
Found RTP audio format 101
Peer audio RTP is at port 216.18.125.7:11766
Found description format G729
Found description format PCMU
Found description format G723
Found description format G726-32
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x115 (g723|ulaw|g726|g729)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 200 in from-sip-external (domain 192.168.1.136)
list_route: hop: sip:A21G9nPtOoPN7pp1Kxtl_eVBVi_g2Iy@216.18.125.7:5065
Transmitting (NAT) to 216.18.125.7:5065:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 216.18.125.7:5065;branch=z9hG4bK-d87543-aaba810e1b86374225fd-1-cHBmNmM3YmU2YjM3YWE3MDM2NDA4NA…-d87543-;received=216.18.125.7
From: "NICOLAS DE CRIS"sip:6134444444@sip.babytel.ca;tag=d177c741
To: sip:1xxxxxxxxxx@216.18.125.3;user=phone
Call-ID: 0b69709d405e09d8a4e17ab6f209f97f
CSeq: 483011905 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:200@192.168.1.136
Content-Length: 0
Transmitting (NAT) to 216.18.125.7:5065:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 216.18.125.7:5065;branch=z9hG4bK-d87543-aaba810e1b86374225fd-1-cHBmNmM3YmU2YjM3YWE3MDM2NDA4NA…-d87543-;received=216.18.125.7
From: "NICOLAS DE CRIS"sip:6134444444@sip.babytel.ca;tag=d177c741
To: sip:1xxxxxxxxxx@216.18.125.3;user=phone;tag=as62f6c046
Call-ID: 0b69709d405e09d8a4e17ab6f209f97f
CSeq: 483011905 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:200@192.168.1.136
Content-Length: 0
We’re at 192.168.1.136 port 13986
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 216.18.125.7:5065:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.18.125.7:5065;branch=z9hG4bK-d87543-aaba810e1b86374225fd-1-cHBmNmM3YmU2YjM3YWE3MDM2NDA4NA…-d87543-;received=216.18.125.7
From: "NICOLAS DE CRIS"sip:6134444444@sip.babytel.ca;tag=d177c741
To: sip:1xxxxxxxxxx@216.18.125.3;user=phone;tag=as62f6c046
Call-ID: 0b69709d405e09d8a4e17ab6f209f97f
CSeq: 483011905 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:200@192.168.1.136
Content-Type: application/sdp
Content-Length: 218
v=0
o=root 29914 29914 IN IP4 192.168.1.136
s=session
c=IN IP4 192.168.1.136
t=0 0
m=audio 13986 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
asterisk1*CLI>
<-- SIP read from 216.18.125.7:5065:
ACK sip:200@192.168.1.136 SIP/2.0
To: sip:1xxxxxxxxxx@216.18.125.3;user=phone;tag=as62f6c046
From: "NICOLAS DE CRIS"sip:6134444444@sip.babytel.ca;tag=d177c741
Via: SIP/2.0/UDP 216.18.125.7:5065;branch=z9hG4bK-d87543-a259e15201f50966-1-cHBmYWQwOTc2ZWM4NTcxYzE3YzM5MQ…-d87543-
Call-ID: 0b69709d405e09d8a4e17ab6f209f97f
CSeq: 483011905 ACK
Contact: sip:A21G9nPtOoPN7pp1Kxtl_eVBVi_g2Iy@216.18.125.7:5065
Max-Forwards: 68
Allow: REGISTER, OPTIONS, INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, INFO, SUBSCRIBE
Supported: em, timer, replaces, path
User-Agent: Audiocodes-Sip-Gateway-TrunkPack 1610/v.4.80A.034.004
Content-Length: 0
— (12 headers 0 lines) —
Destroying call '7b771e232f12938c72e8eed915707ea1@127.0.0.1’
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 216.18.125.7:5065:
REGISTER sip:sip.babytel.ca SIP/2.0
Via: SIP/2.0/UDP 192.168.1.136:5060;branch=z9hG4bK7ebe46cf;rport
From: sip:1xxxxxxxxxx@sip.babytel.ca;tag=as6dbd9923
To: sip:1xxxxxxxxxx@sip.babytel.ca
Call-ID: 7b771e232f12938c72e8eed915707ea1@127.0.0.1
CSeq: 110 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username=“1xxxxxxxxxx”, realm=“sip.babytel.ca”, algorithm=MD5, uri=“sip:sip.babytel.ca”, nonce=“4596b97cde9570e3b725a1ce082edd2568d7a3ba”, response=“2d84d30409563592cdb7ae1180788a4f”, opaque=""
Expires: 120
Contact: sip:200@192.168.1.136
Event: registration
Content-Length: 0
asterisk1*CLI>
<-- SIP read from 216.18.125.7:5065:
SIP/2.0 200 OK
To: sip:1xxxxxxxxxx@sip.babytel.ca;tag=a3ff6d35
From: sip:1xxxxxxxxxx@sip.babytel.ca;tag=as6dbd9923
Via: SIP/2.0/UDP 192.168.1.136:5060;branch=z9hG4bK7ebe46cf
Call-ID: 7b771e232f12938c72e8eed915707ea1@127.0.0.1
CSeq: 110 REGISTER
Contact: sip:200@192.168.1.136;expires=52
Content-Length: 0
— (8 headers 0 lines) —
Scheduling destruction of call ‘7b771e232f12938c72e8eed915707ea1@127.0.0.1’ in 32000 ms
Scheduling destruction of call ‘0b69709d405e09d8a4e17ab6f209f97f’ in 32000 ms
set_destination: Parsing sip:A21G9nPtOoPN7pp1Kxtl_eVBVi_g2Iy@216.18.125.7:5065 for address/port to send to
set_destination: set destination to 216.18.125.7, port 5065
Reliably Transmitting (NAT) to 216.18.125.7:5065:
BYE sip:A21G9nPtOoPN7pp1Kxtl_eVBVi_g2Iy@216.18.125.7:5065 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.136:5060;branch=z9hG4bK10d7ccdd;rport
From: sip:1xxxxxxxxxx@216.18.125.3;user=phone;tag=as62f6c046
To: "NICOLAS DE CRIS"sip:6134444444@sip.babytel.ca;tag=d177c741
Contact: sip:200@192.168.1.136
Call-ID: 0b69709d405e09d8a4e17ab6f209f97f
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
asterisk1*CLI>
<-- SIP read from 216.18.125.7:5065:
SIP/2.0 200 OK
To: "NICOLAS DE CRIS"sip:6134444444@sip.babytel.ca;tag=d177c741
From: sip:1xxxxxxxxxx@216.18.125.3;user=phone;tag=as62f6c046
Via: SIP/2.0/UDP 192.168.1.136:5060;branch=z9hG4bK10d7ccdd
Call-ID: 0b69709d405e09d8a4e17ab6f209f97f
CSeq: 102 BYE
Contact: sip:A21G9nPtOoPN7pp1Kxtl_eVBVi_g2Iy@216.18.125.7:5065
Allow: REGISTER, OPTIONS, INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, INFO, SUBSCRIBE
Server: Audiocodes-Sip-Gateway-TrunkPack 1610/v.4.80A.034.004
Supported: em, timer, replaces, path
Content-Length: 0
— (11 headers 0 lines) —
Destroying call ‘0b69709d405e09d8a4e17ab6f209f97f’