Hello,
I use voiptalk.org in the UK for my UK numbers. It’s worked well for years, but voiptalk have announced that they are ending support for iax, so I have to switch to sip. All of there sip configuration documents for Asterisk are ancient, based on version 1.4 and chan_sip. So I am attempting to do it with pjsip. Incoming calls work fine, but I can’t make any outgoing calls. Here’s the configuration:
Transport: <TransportId…> <BindAddress…>
Transport: transport-udp-ipv4 udp 3 96 0.0.0.0:5060
ParameterName : ParameterValue
allow_reload : false
allow_wildcard_certs : No
async_operations : 1
bind : 0.0.0.0:5060
ca_list_file :
ca_list_path :
cert_file :
cipher :
cos : 3
domain :
external_media_address : my.ipv4.add
external_signaling_address : my.ipv4.add
external_signaling_port : 0
local_net : 10.47.229.0/255.255.255.0
method : unspecified
password :
priv_key_file :
protocol : udp
require_client_cert : No
symmetric_transport : false
tcp_keepalive_enable : true
tcp_keepalive_idle_time : 30
tcp_keepalive_interval_time : 1
tcp_keepalive_probe_count : 5
tos : 96
verify_client : No
verify_server : No
websocket_write_timeout : 100Auth: voiptalk-ilj/myusernameParameterName : ParameterValue
auth_type : userpass
md5_cred :
nonce_lifetime : 32
oauth_clientid :
oauth_secret :
password : mypassword
realm :
refresh_token :
username : myusernameParameterName : ParameterValue
authenticate_qualify : false
contact : sip:myusername@voiptalk.org:5060
default_expiration : 3600
mailboxes :
max_contacts : 0
maximum_expiration : 7200
minimum_expiration : 60
outbound_proxy : sip:nat.voiptalk.org:5065;lr
qualify_frequency : 41
qualify_timeout : 3.000000
remove_existing : false
remove_unavailable : false
support_path : false
voicemail_extension :Endpoint: 020xxxxxxxx/+4420xxxxxxxx Not in use 0 of inf
OutAuth: voiptalk-ilj/myusername
Aor: voiptalk-ilj 0
Contact: voiptalk-ilj/sip:myusername@voiptalk.org:50 a3b4d82b8b Avail 2.576
Transport: transport-udp-ipv4 udp 3 96 0.0.0.0:5060
Identify: voiptalk-ilj/020xxxxxxxx
Header: To: sip:myusername@voiptalk.orgParameterName : ParameterValue
100rel : yes
accept_multiple_sdp_answers : false
accountcode : ionos-in-voiptalk-1
acl :
aggregate_mwi : true
allow : (g722|ulaw|g729)
allow_overlap : true
allow_subscribe : true
allow_transfer : true
allow_unauthenticated_options : false
aors : voiptalk-ilj
asymmetric_rtp_codec : false
auth :
bind_rtp_to_media_address : false
bundle : false
call_group :
callerid : “Ian Jones” <+4420xxxxxxxx>
callerid_privacy : allowed_not_screened
callerid_tag :
codec_prefs_incoming_answer : prefer:pending, operation:intersect, keep:all, transcode:allow
codec_prefs_incoming_offer : prefer:pending, operation:intersect, keep:all, transcode:allow
codec_prefs_outgoing_answer : prefer:pending, operation:intersect, keep:all, transcode:allow
codec_prefs_outgoing_offer : prefer:pending, operation:union, keep:all, transcode:allow
connected_line_method : invite
contact_acl :
context : incoming-on-voiptalk
cos_audio : 5
cos_video : 4
device_state_busy_at : 0
direct_media : false
direct_media_glare_mitigation : none
direct_media_method : invite
disable_direct_media_on_nat : false
dtls_auto_generate_cert : No
dtls_ca_file :
dtls_ca_path :
dtls_cert_file :
dtls_cipher :
dtls_fingerprint : SHA-256
dtls_private_key :
dtls_rekey : 0
dtls_setup : active
dtls_verify : No
dtmf_mode : rfc4733
fax_detect : false
fax_detect_timeout : 0
follow_early_media_fork : true
force_avp : false
force_rport : true
from_domain :
from_user : myusername
g726_non_standard : false
geoloc_incoming_call_profile :
geoloc_outgoing_call_profile :
ice_support : false
identify_by : username,ip
ignore_183_without_sdp : false
inband_progress : false
incoming_call_offer_pref : local
incoming_mwi_mailbox :
language :
mailboxes :
max_audio_streams : 1
max_video_streams : 1
media_address :
media_encryption : no
media_encryption_optimistic : false
media_use_received_transport : false
message_context :
moh_passthrough : false
moh_suggest : default
mwi_from_user :
mwi_subscribe_replaces_unsolicited : no
named_call_group :
named_pickup_group :
notify_early_inuse_ringing : false
one_touch_recording : false
outbound_auth : voiptalk-ilj
outbound_proxy : sip:nat.voiptalk.org:5065;lr
outgoing_call_offer_pref : remote_merge
overlap_context :
pickup_group :
preferred_codec_only : false
record_off_feature : automixmon
record_on_feature : automixmon
refer_blind_progress : true
rewrite_contact : true
rpid_immediate : false
rtcp_mux : false
rtp_engine : asterisk
rtp_ipv6 : false
rtp_keepalive : 0
rtp_symmetric : false
rtp_timeout : 0
rtp_timeout_hold : 0
sdp_owner : -
sdp_session : Asterisk
security_mechanisms :
security_negotiation : no
send_aoc : false
send_connected_line : yes
send_diversion : true
send_history_info : false
send_pai : false
send_rpid : true
set_var :
srtp_tag_32 : false
stir_shaken : no
stir_shaken_profile :
sub_min_expiry : 0
subscribe_context :
suppress_q850_reason_headers : false
t38_bind_udptl_to_media_address : false
t38_udptl : false
t38_udptl_ec : none
t38_udptl_ipv6 : false
t38_udptl_maxdatagram : 0
t38_udptl_nat : false
timers : no
timers_min_se : 90
timers_sess_expires : 1800
tone_zone :
tos_audio : 184
tos_video : 136
transport : transport-udp-ipv4
trust_connected_line : yes
trust_id_inbound : true
trust_id_outbound : true
use_avpf : false
use_ptime : false
user_eq_phone : false
voicemail_extension :
webrtc : no
Here is what happens:
09:34:08.396352 IP (tos 0x60, ttl 64, id 55636, offset 0, flags [DF], proto UDP (17), length 1093)
lisette.mydom.net.sip > 77.240.48.201.5065: [bad udp cksum 0x7230 → 0x7945!] SIP, length: 1065
INVITE sip:0902@voiptalk.org:5060 SIP/2.0
Via: SIP/2.0/UDP my.ipv4.add:5060;rport;branch=z9hG4bKPj08780fee-f986-4a42-9d9d-6b7963085393
From: sip:myusername@10.47.229.5;tag=70f6f0b5-143c-4813-ae37-b95bce27ab47
To: sip:0902@voiptalk.org
Contact: sip:myusername@my.ipv4.add:5060
Call-ID: f33b64aa-5fe9-4e83-bc10-ae86f16bd07f
CSeq: 27152 INVITE
Route: sip:nat.voiptalk.org:5065;lr
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, INFO, MESSAGE
Supported: 100rel, replaces, norefersub, histinfo
Remote-Party-ID: “Lisette” sip:28@10.47.229.5;party=calling;privacy=off;screen=no
Max-Forwards: 70
User-Agent: Asterisk PBX 20.8.1
Content-Type: application/sdp
Content-Length: 310v=0 o=- 1338062286 1338062286 IN IP4 my.ipv4.add s=Asterisk c=IN IP4 my.ipv4.add t=0 0 m=audio 10042 RTP/AVP 0 9 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:9 G722/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:140 a=sendrecv09:34:08.398000 IP (tos 0x0, ttl 57, id 0, offset 0, flags [DF], proto UDP (17), length 441)
77.240.48.201.5065 > lisette.mydom.net.sip: [udp sum ok] SIP, length: 413
SIP/2.0 403 Forbidden - no OBP
Via: SIP/2.0/UDP my.ipv4.add:5060;rport=5060;branch=z9hG4bKPj08780fee-f986-4a42-9d9d-6b7963085393
From: sip:myusername@10.47.229.5;tag=70f6f0b5-143c-4813-ae37-b95bce27ab47
To: sip:0902@voiptalk.org;tag=3eeda7e8f0549c38df1ae575f0b8a0ef.8892
Call-ID: f33b64aa-5fe9-4e83-bc10-ae86f16bd07f
CSeq: 27152 INVITE
Server: OpenSIPS (1.6.2-notls (i386/linux))
Content-Length: 009:34:08.398642 IP (tos 0x60, ttl 64, id 55637, offset 0, flags [DF], proto UDP (17), length 487)
lisette.mydom.net.sip > 77.240.48.201.5065: [bad udp cksum 0x6fd2 → 0x1972!] SIP, length: 459
ACK sip:0902@voiptalk.org:5060 SIP/2.0
Via: SIP/2.0/UDP my.ipv4.add:5060;rport;branch=z9hG4bKPj08780fee-f986-4a42-9d9d-6b7963085393
From: sip:myusername@10.47.229.5;tag=70f6f0b5-143c-4813-ae37-b95bce27ab47
To: sip:0902@voiptalk.org;tag=3eeda7e8f0549c38df1ae575f0b8a0ef.8892
Call-ID: f33b64aa-5fe9-4e83-bc10-ae86f16bd07f
CSeq: 27152 ACK
Route: sip:nat.voiptalk.org:5065;lr
Max-Forwards: 70
User-Agent: Asterisk PBX 20.8.1
Content-Length: 009:34:08.937213 IP (tos 0x0, ttl 57, id 0, offset 0, flags [DF], proto UDP (17), length 300)
77.240.48.201.5065 > lisette.mydom.net.sip: [udp sum ok] SIP, length: 272
OPTIONS sip:myusername@my.ipv4.add:5060;line=evqboew SIP/2.0
Via: SIP/2.0/UDP 77.240.48.201:5065;branch=0
From: sip:vnt-10@voiptalk.org;tag=b2660808
To: sip:myusername@voiptalk.org
Call-ID: 72d4ac57-f489c988-80d52e@77.240.48.201
CSeq: 1 OPTIONS
Content-Length: 009:34:08.937915 IP (tos 0x60, ttl 64, id 55744, offset 0, flags [DF], proto UDP (17), length 825)
lisette.mydom.net.sip > 77.240.48.201.5065: [bad udp cksum 0x7124 → 0xec6a!] SIP, length: 797
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 77.240.48.201:5065;rport=5065;received=77.240.48.201;branch=0
Call-ID: 72d4ac57-f489c988-80d52e@77.240.48.201
From: sip:vnt-10@voiptalk.org;tag=b2660808
To: sip:myusername@voiptalk.org;tag=0
CSeq: 1 OPTIONS
Accept: application/sdp, application/simple-message-summary, application/pidf+xml, application/xpidf+xml, application/cpim-pidf+xml, application/dialog-info+xml, application/pidf+xml, application/dialog-info+xml, application/simple-message-summary, message/sipfrag;version=2.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, INFO, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Accept-Encoding: identity
Accept-Language: en
Server: Asterisk PBX 20.8.1
Content-Length: 009:34:11.745303 IP (tos 0x60, ttl 64, id 56141, offset 0, flags [DF], proto UDP (17), length 547)
lisette.mydom.net.sip > 77.240.48.201.5065: [bad udp cksum 0x700e → 0x13ff!] SIP, length: 519
OPTIONS sip:myusername@voiptalk.org:5060 SIP/2.0
Via: SIP/2.0/UDP my.ipv4.add:5060;rport;branch=z9hG4bKPjccabeb67-8d9c-4476-b151-baf599f99e5e
From: sip:myusername@voiptalk.org;tag=8e456f2a-f13e-4e59-b165-82c9ca979671
To: sip:myusername@voiptalk.org
Contact: sip:myusername@my.ipv4.add:5060
Call-ID: bac73275-0f53-41e6-8014-723dee56f55d
CSeq: 6269 OPTIONS
Route: sip:nat.voiptalk.org:5065;lr
Route: sip:nat.voiptalk.org:5065;lr
Max-Forwards: 70
User-Agent: Asterisk PBX 20.8.1
Content-Length: 009:34:11.746366 IP (tos 0x0, ttl 57, id 0, offset 0, flags [DF], proto UDP (17), length 651)
77.240.48.201.5065 > lisette.mydom.net.sip: [udp sum ok] SIP, length: 623
SIP/2.0 200 OK
Record-Route: sip:xuser@77.240.48.201;r2=on;lr=on;ftag=8e456f2a-f13e-4e59-b165-82c9ca979671
Record-Route: sip:xuser@77.240.48.201:5065;r2=on;lr=on;ftag=8e456f2a-f13e-4e59-b165-82c9ca979671
Via: SIP/2.0/UDP my.ipv4.add:5060;received=my.ipv4.add;rport=5060;branch=z9hG4bKPjccabeb67-8d9c-4476-b151-baf599f99e5e
From: sip:myusername@voiptalk.org;tag=8e456f2a-f13e-4e59-b165-82c9ca979671
To: sip:myusername@voiptalk.org;tag=c97b4d1cb1f3d0da549e06a8d482ef63.cc47
Call-ID: bac73275-0f53-41e6-8014-723dee56f55d
CSeq: 6269 OPTIONS
Server: OpenSIPS (1.5.3-notls (x86_64/linux))
Content-Length: 0
Any help will be appreciated!
Ian