Loop Back

[Scott1234]

Attempting to get loop back working so if you dial another on-net so to speak DID that's on the PBX it will connect internally. 

I notice on loopback it wants to UPDATE the SIP Contact to say, 200@125.125.125.125  i.e extension@rawip from the extension@domain original contact. 

I suspect my loop back calls are not working when testing because I have , Ignore packets that do not match a domain on the system turned on, is there a way to do it preserving the contact domain?

I can see otherwise its matching to the other domain no problem 

 

 

 

 

[Vodia PBX]

The host in the contact does not interfere with the domain check. This should be no problem. Do you see the call in the tenant call log?

[Scott1234]

4 hours ago, Vodia PBX said:

The host in the contact does not interfere with the domain check. This should be no problem. Do you see the call in the tenant call log?

I just get static and the call gets held up there for ever until ended on Domain A never shows up in Domain B logs

Trunk debug 9 on Domain A shows its matched the DID on Domain B 

Trunk debug 9 on Domain B shows nothing

 

[9] 19:36:50.282 TRUN: Dialplan: Simple match begin of 0877777777 to *

[9] 19:36:50.282 TRUN: Last message repeated 2 times

[9] 19:36:50.282 TRUN: Dialplan: Evaluating * against 0877777777@domainA.com

[5] 19:36:50.282 TRUN: Dialplan "loopback-test": Match 0877777777@domainA to sip:+61877777777@domainB;user=phone on Try Loopback trunk

[7] 19:36:50.283 MEDI: Port 345: Set codec preference count 2

[7] 19:36:50.283 MEDI: Port 346: Set codec preference count 2

[8] 19:36:50.283 MEDI: Port 346: state code from 0 to 100

[9] 19:36:50.283 MEDI: Port 346: Adding codec PCMA/8000 to available list

[9] 19:36:50.283 MEDI: Port 346: Update codecs preference size 2, available codecs size 2

[7] 19:36:50.283 MEDI: Port 346: Allocated ports 53646 and 53647

[8] 19:36:50.284 MEDI: Port 345: state code from 0 to 183

[8] 19:36:50.284 MEDI: Port 345: Ignore double SDP

[9] 19:36:50.284 MEDI: Port 345: Adding codec PCMA/8000 to available list

[9] 19:36:50.284 MEDI: Port 345: Update codecs preference size 2, available codecs size 2

[6] 19:36:50.284 MEDI: Port 345: Choose codec PCMA/8000

[6] 19:36:50.520 MEDI: Port 345: Sending RTP to my.wan.ip:49940, codec PCMA/8000

[7] 19:36:50.601 MEDI: Port 345: Set DTLS SRTP key for client

[9] 19:36:50.623 MEDI: Port 345: Received first RTP packet

Static so hangup,

[8] 19:36:58.967 MEDI: Port 345: Clearing port with SIP Call-ID kwegJ14w

[8] 19:36:58.969 MEDI: Port 346: state code from 100 to 487

[8] 19:36:58.969 MEDI: Port 346: Send hangup with reason bye

[8] 19:37:30.969    MEDI:    Port 346: Clearing port with SIP Call-ID be3cbe2b@pbx

[Scott1234]

I think it might be because I am using non standard sip ports and its not factoring that in when doing the loopback? 

I forgot to look before but the PCAP does not define any SIP ports so it would be defaulting to 5060 ?

The loop back call user agent shows up as Vodia-WEBRTC, maybe as my attempts where vodia app.

[Scott1234]

After further testing the only way I could get loop back to work when the base system was on non-standard SIP ports was to also add the original ports back in to the "SIP Settings" page. Even with specifying the 127.0.0.1:customport on the loopback trunk to use the nonstandard port it would not work.

¯\_(ツ)_/¯

Edit - Got it working on the standard port. I will post some doco as I was not able to find much to get this working.

[Scott1234]

Inter domain dialling multi-tenant.

As I was not able to find good doco when attempting this, I have made my own notes to share. Obviously special considerations are needed for overlapping extension domains, which the standard doco talks about.

Ideally, I would like a way to only allow the alias/DID to be matched on the loop back not extension, mainly so when I dial the on-net tenant DID it goes internally, I don’t necessarily care about extension number to extension number but an option to pick would be nice. But I might be able to do that with, Use a list of expressions option for the inbound side matching. 

Note,

• Using global tenant dial plans
• This does not leverage a "try loop back" dial plan at all, not sure the danger but seems to be fine. 
• No need to disable Loopback detection 

What I did,

1. Create localhost domain
2. Create trunk in localhost domain called on-net, sip gateway no registration, Outbound Proxy = sip:127.0.0.1:5060 (or pbx custom sip port) with the following settings.

3. aadr: 127.0.0.1
   analog: false
   ani_emergency: 
   ani_regular: 
   area_code: 
   bcp: 
   behind_nat: false
   cid_update: 
   cobusy: 500 Line Unavailable
   code: 
   codec_count: 
   codec_lock: false
   codecs: 
   codest: 
   contact_hdr: 
   country_code: 
   dial_extension: 
   dialplan: 
   dir: 
   dis: false
   dtmf: false
   dtmf_mode: 
   earlymedia: true
   expires: 3600
   failover: except_busy
   fraction: 128
   from_source: ppi
   from_user: 
   glob: 
   global: true
   hcv: 
   hd: 
   hf: {from}
   hpai: 
   hppi: 
   hpr: 
   hrpi: 
   hru: {request-uri}
   ht: <{request-uri}>
   ice: false
   icid: 
   identity: 
   ignore_18x_sdp: true
   info_agent: false
   interoffice: false
   minimum: 10
   name: on-net
   other_headers: 
   outbound_proxy: sip:127.0.0.1:5060
   pidflo: false
   prack: false
   prefix: 
   redirect: false
   reg_account: 
   reg_display: 
   reg_keep: 
   reg_registrar: 
   reg_user: 
   remote_party: 
   request_timeout: 
   require: 
   reregister_dns: false
   rfcrtp: false
   ring180: false
   rtcpxr: false
   rtp_begin: 
   rtp_end: 
   sdpreq: 
   send_email: 
   sip_port: 
   spam: false
   stir: 
   t38_enabled: false
   teams: false
   tel: true
   trusted: true
   type: gateway
   use_epid: false
   use_history: false
   use_sdes: 
   use_uuid: false
   user_defined_hdr: 
   wrtc_dest_name: 
   wrtc_dest_number: 

    

4. Edit your dial plans and place below emergency but above your normal region dial plans. 

   15;on-net;;*;*;;false

As trunk on-net is set to fail-over except busy if no internal match is found it will flow as per the rest of the dial plan.