Callerid=sipusername while using a tunk to vega 50 as fxo gateway for my default incoming/outgoing route

[sugar]

Hi all, I just installed the vega 50 and configured as fxo gateway for asterisk 20.

It can make/receive calls with this trunk, but the caller ID of the incoming calls from this trunk is set to “sipuser” (“sipuser” is the username that I did setup for the trunk authentication), instead of the actual caller id number. Where can I look at this or what can cause this? I doublechecked the trunk config, incoming route config, outgoing route config, and still cant find how to correct this.

Note the incoming route sends the calls to my test extension.

thanks!

[david551]

The INVITE received from the gateway.

If you have told it to use that as user name for authentication, it will use the From user, which would, otherwise, be the caller ID. You would need to enable one of Remote-Party-ID, or P-Asserted-Identity, at both ends.

Why are you authenticating inbound? Normally you want the gateway to do any authentication, as it is the one that can make toll calls.

[sugar]

I did set it up that way because I don’t know the correct way to do it, in fact this is my first time installing/touching asterisk and sangoma vega as well.

I am still learning, so if I disable authentication between the vega gateway and asterisk, it will use the real caller id for incoming calls?

Also, where do I find the “Remote-Party-ID” or “P-Asserted-Identity” in the vega gateway, or in some section of asterisk config? I don’t remember seeing these settings last night that I installed this setup.

thanks!

[david551]

I don’t know about Vega. For inbound handling of the headers, see

https://docs.asterisk.org/Latest_API/API_Documentation/Module_Configuration/res_pjsip/#trust_id_inbound

Also, see page 284 of https://ftp.sangoma.com/vega/custom/TTSL/Vega_Admin_Guide_R11_v1.4.pdf

[sugar]

I will read these documents

thanks!

[sugar]

by the way, is there another forum for commercial products like this vega? or should I just use the classic channel of registering it and open a ticket?

Edit, never-mind last question, I just found it

thanks again

[asterisknooob]

Can you share some screenshots of the settings if it’s GUI based or your trunk configuration if CLI based. With Blur or put xx on sensitive information like username password.

[sugar]

In the Trunk → PJSIP → Advanced, I have tried all four settings of “Send RPID/PAI” and since I disabled authentication for this trunk now incoming caller ID is always “Anonymous” and not showing the actual caller id:

As far as the trunk settings, does the otuput of “pjsip show endpoint MY-TRUNK” work?

Connected to Asterisk 20.5.2 currently running on freepbx (pid = 8563)
freepbx*CLI> pjsip show endpoint sangoma-vega

** Endpoint: <Endpoint/CID…> <State…> <Channels.>**
** I/OAuth: <AuthId/UserName…>**
** Aor: <Aor…> **
** Contact: <Aor/ContactUri…> <Hash…> <RTT(ms)…>**
** Transport: <TransportId…> <BindAddress…>**
** Identify: <Identify/Endpoint…>**
** Match: <criteria…>**
** Channel: <ChannelId…> <State…> <Time…>**
** Exten: <DialedExten…> CLCID: <ConnectedLineCID…>**
==========================================================================================

** Endpoint: sangoma-vega Not in use 0 of inf**
** Aor: sangoma-vega 0**
** Contact: sangoma-vega/sip:192.168.5.3:5060 27bfc6dead Avail 6.062**
** Transport: 0.0.0.0-udp udp 3 96 0.0.0.0:5060**
** Identify: sangoma-vega/sangoma-vega**
** Match: 192.168.5.3/32**

** ParameterName : ParameterValue**
** ===================================================================================================**
** 100rel : yes**
** accept_multiple_sdp_answers : false**
** accountcode : **
** acl : **
** aggregate_mwi : true**
** allow : (ulaw|alaw|gsm|g726|g722|h264|mpeg4)**
** allow_overlap : true**
** allow_subscribe : true**
** allow_transfer : true**
** allow_unauthenticated_options : false**
** aors : sangoma-vega**
** asymmetric_rtp_codec : false**
** auth : **
** 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 : from-pstn**
** 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 : auto**
** fax_detect : true**
** 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 : es_419**
** 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 : true**
** 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 : true**
** send_rpid : true**
** set_var : **
** srtp_tag_32 : false**
** stir_shaken : off**
** 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 : 0.0.0.0-udp**
** 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**

freepbx*CLI>

[sugar]

can something be wrong with my firmware?

I just noticed this setting under sip advanced settings, but it is overriding all the incoming calls to “Anonymous” not only the private ones. I cant find where else to look for.

Here a copy of my current config txt… If somebody knows about this platform or had similar issue I would really appreciate any help.

config(1).txt (86.8 KB)

Thanks!

[david551]

That’s a FreePBX GUI. That is not supported on this forum. You were asked for the Vega’s.

[sugar]

I apologize, my bad, I just uploaded the full vega config txt file on my previous reply.

[asterisknooob]

Send RPID/PAI Should be No.
If you mention from_user = XXXXXXX Remove this one no need because you will use extension’s caller ID. In case of no callerid is defined for user extension then this will be the default callerid.

[sugar]

@asterisknooob

RPID/PA

Now is setup like that but I still receive the calls as “Anonymous”

[asterisknooob]

Show me the extension settings you are using for the outbound.

[sugar]

You mean “Connectivity” → “Outbound Routes” or my “Applications” → “Extensions” settings?

Aldo

[david551]

OP’s problem is for inbound. In fact, analogue FXO lines cannot sent caller ID outbound.

Also, enabling RPID/PAI rarely breaks anything. In particular, if they are not received, inbound, they from user name is used instead, which is the same as it they weren’t enabled.

One reason for falling back to anonymous is that there is either no caller ID being received, or the caller ID is not being sent in the correct way for the Vega.

In terms of getting the right Asterisk configuration, the most useful thing to do is to use “pjsip set logger on” to capture the INVITE. If that includes the caller ID, one can work out how to get Asterisk to capture it. If it doesn’t, there is no way you can fix this from the Asterisk end and you need to work on the Vega until it starts sending the caller ID.

Also, the green themed screen shots relate to FreePBX, with which most people on this forum are not familiar, and does not give the user full control of endpoint configuraton.

[asterisknooob]

Both will be more good and easy to understand what is happening.

[david551]

This thread has moved to From: "Anonymous" <sip:Anonymous@x.x.x.x>.... why? - freepbx - FreePBX Community Forums

[sugar]

@david551

This thread has moved to From:

Hi, what means the message " This thread has moved to …" means that we should continue this discussion on the referenced post or Vice versa ?

Aldo

[david551]

That’s what I was hinting.