How can I configure a sip phone with ASL 3 ?
thanks I will check it out
First thanks for the tip, and my fault not check entery documentation, so pjsip connected and receive audio right, but not audio outgoing at all, what I miss ?
Thx
Was audio working before you added SIP?
There are a lot of partsā¦so more info will be needed. What does the asterisk log say? Verified with different SIP clients? to make sure itās not the problem.
Hey good morning
I donāt know if my reply sent correctly, but I write again,. yes the audio worked right before, in fact I have two more nodes (hamvoip and asl2) running with sip accounts without issues using the same IP phone Grandstream.
ASL1/ASL2/HamVOIPās configurations are severely behind - 15+ years - current Asterisk configuration syntax. ASL3 is based on the latest Asterisk. Weāll need more information - logs, configuration files, etc. - to help you troubleshoot your setup.
Those version with 15+ years old are working stable, this latest version is runing in Debian 12, I had to stop using Debian 12 due many issues (for others project), so if you want, I can paste here, all pjsip verbose logs, honestly debian 12 itās a bad choice for that, and I hope you donāt take my opinion the wrong way, my only wish is that this works stable and it really isnāt.
There are other people using SIP endpoints with ASL3 just fine. My point about the configuration is that itās unlikely youāre going to be able to copy-paste a config from Asterisk 1.4 into Asterisk 20 and have it ājust workā. That has nothing to do with Debian 12. There are others in this community that likely can provide experienced advice in this area, but you havenāt provided any config samples, logs, or debugging information that gives anyone a place to start.
Im not copy and paste anything from previous ASL2, I follow the instructions step by step, using pjsip, but anyway, thanks for your reply, Im back to use ASL2, and Debian 12 having issues with many things you like it or not, 73, by the way here you are my config extension
Endpoint: <Endpoint/CIDā¦> <Stateā¦> <Channels.>
I/OAuth: <AuthId/UserNameā¦>
Aor: <Aorā¦>
Contact: <Aor/ContactUriā¦> <Hashā¦> <RTT(ms)ā¦>
Transport: <TransportIdā¦> <BindAddressā¦>
Identify: <Identify/Endpointā¦>
Match: <criteriaā¦>
Channel: <ChannelIdā¦> <Stateā¦> <Timeā¦>
Exten: <DialedExtenā¦> CLCID: <ConnectedLineCIDā¦>
Endpoint: 1001 In use 1 of inf
InAuth: 1001/1001
Aor: 1001 1
Contact: 1001/sip:1001@192.168.100.59:27676 3bc92c4959 NonQual nan
Transport: transport-udp udp 0 0 0.0.0.0:5060
Channel: PJSIP/1001-00000000/Rpt Up 00:03:10
Exten: 596481 CLCID: āā <>
ParameterName : ParameterValue
100rel : yes
accept_multiple_sdp_answers : false
accountcode :
acl :
aggregate_mwi : true
allow : (ulaw|gsm)
allow_overlap : true
allow_subscribe : true
allow_transfer : true
allow_unauthenticated_options : false
aors : 1001
asymmetric_rtp_codec : false
auth : 1001
bind_rtp_to_media_address : false
bundle : false
call_group :
callerid :
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 : sip-phones
cos_audio : 0
cos_video : 0
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 :
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 :
outbound_proxy :
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 : false
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 : false
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 : yes
timers_min_se : 90
timers_sess_expires : 1800
tone_zone :
tos_audio : 0
tos_video : 0
transport : transport-udp
trust_connected_line : yes
trust_id_inbound : false
trust_id_outbound : false
use_avpf : false
use_ptime : false
user_eq_phone : false
voicemail_extension :
webrtc : no
If Debian 12 is such a bad choice, someone might want to tell Asterisk as their FreePBX is now released on it (no more CentOS). I have a FreePBX server on Debian 12 running several PJSIP extensions and all are trunked to my ASL3 server. Audio quality is amazing.
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