2 Asterisk Sip Account to the same phone

I have got a lab to do with 3 phones (Aastra 55i) and Asterisk.
This is easly, but my issue is one of this 3 phone should ring on line 1 if someone called 8540 and line 2 if some one called 8000.

My First thinking was to configure Aastra Line 1 with 8540 SIP Account and Line 2 with SIP Account 8000.
All of sip peers is register.
8000/8000 x.y.z.33 D A 5065 OK(32ms) CacheRT
8542/8542 x.y.z.33 D A 5065 OK(32ms) CacheRT

When called 8540 is ok, but when I try to call 8000 I have got hangup. Below SIP trace from asterisk:


set_destination: Parsing sip:8540@192.168.203.33:5065 for address/port to send to
set_destination: set destination to 192.168.203.33:5065
Reliably Transmitting (no NAT) to 192.168.203.33:5065:
NOTIFY sip:8540@192.168.203.33:5065 SIP/2.0
Via: SIP/2.0/UDP 192.168.203.211:5060;branch=z9hG4bK43e52c1e;rport
Max-Forwards: 70
From: sip:8542@srvvoip.ipkol.lan:5060;tag=as2262af67
To: “8540” sip:8540@srvvoip.ipkol.lan:5060;tag=660556b0b9
Contact: sip:8542@192.168.203.211:5060
Call-ID: 27712e72818804aa
CSeq: 118 NOTIFY
User-Agent: VOIPProd
Subscription-State: active
Event: dialog
Content-Type: application/dialog-info+xml
Content-Length: 212

<?xml version="1.0"?>confirmed

== Extension Changed 8542[blf-Standart] new state InUse for Notify User 8540
– Executing [8000@full:1] NoOp(“SIP/8542-0000000b”, “Appel interne depuis “” <497288542> vers 8000”) in new stack
– Executing [8000@full:2] Set(“SIP/8542-0000000b”, “_ALERT_INFO=info=alert-autoanswer”) in new stack
– Executing [8000@full:3] Dial(“SIP/8542-0000000b”, “SIP/8000,30,Tt”) in new stack
== Using SIP RTP CoS mark 5
– Couldn’t call 8000
Scheduling destruction of SIP dialog ‘62e69c9967b09a3333a56b9705b3bc5c@192.168.203.211:5060’ in 6400 ms (Method: INVITE)
== Everyone is busy/congested at this time (0:0/0/0)
– Auto fallthrough, channel ‘SIP/8542-0000000b’ status is ‘CHANUNAVAIL’
== Extension Changed 8542[blf-Standart] new state Idle for Notify User 8540 (queued)

<— SIP read from UDP:192.168.203.33:5065 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.203.211:5060;branch=z9hG4bK43e52c1e;rport=5060;received=192.168.203.211
From: sip:8542@srvvoip.ipkol.lan:5060;tag=as2262af67
To: “8540” sip:8540@srvvoip.ipkol.lan:5060;tag=660556b0b9
Call-ID: 27712e72818804aa
CSeq: 118 NOTIFY
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference, LocalModeStatus
Server: Aastra 55i/3.2.2.56
Supported: path
Content-Length: 0

<------------->
— (11 headers 0 lines) —
SIP Response message for INCOMING dialog NOTIFY arrived
set_destination: Parsing sip:8540@192.168.203.33:5065 for address/port to send to
set_destination: set destination to 192.168.203.33:5065
Reliably Transmitting (no NAT) to 192.168.203.33:5065:
NOTIFY sip:8540@192.168.203.33:5065 SIP/2.0
Via: SIP/2.0/UDP 192.168.203.211:5060;branch=z9hG4bK0762a355;rport
Max-Forwards: 70
From: sip:8542@srvvoip.ipkol.lan:5060;tag=as2262af67
To: “8540” sip:8540@srvvoip.ipkol.lan:5060;tag=660556b0b9
Contact: sip:8542@192.168.203.211:5060
Call-ID: 27712e72818804aa
CSeq: 119 NOTIFY
User-Agent: VOIPProd
Subscription-State: active
Event: dialog
Content-Type: application/dialog-info+xml
Content-Length: 213


I do not know how to solve this problem. Some could help me ?

Thks in advance.

Gad.

The trace shows no call attempt (INVITE transaction). It shows a NOTIFY transaction, used for things like message waiting indications.

He David,

First thinks for your answer. Here there more trace, where we can see “Couldn’t call 8000” :
— == Extension Changed 8542[blf-Standart] new state InUse for Notify User 8540
– Executing [8000@full:1] NoOp(“SIP/8542-0000000d”, “Appel interne depuis “” <497288542> vers 8000”) in new stack
– Executing [8000@full:2] Set(“SIP/8542-0000000d”, “_ALERT_INFO=info=alert-autoanswer”) in new stack
– Executing [8000@full:3] Dial(“SIP/8542-0000000d”, “SIP/8000,30,Tt”) in new stack
== Using SIP RTP CoS mark 5
– Couldn’t call 8000
Scheduling destruction of SIP dialog ‘73ed9638698483b75c1c10b859d4b66d@192.168.203.211:5060’ in 6400 ms (Method: INVITE)
== Everyone is busy/congested at this time (0:0/0/0)
– Auto fallthrough, channel ‘SIP/8542-0000000d’ status is ‘CHANUNAVAIL’
== Extension Changed 8542[blf-Standart] new state Idle for Notify User 8540 (queued)

I really do not understand why ? here is asterisk sip registration :
8000/8000 x.y.z.33 D A 5065 OK (30 ms) Cached RT
8540/8540 x.y.z.33 D A 5065 OK (30 ms) Cached RT
8541/8541 x.y.z.251 D A 16398 OK (6 ms) Cached RT

Dialplan info
’8000’ => 1. NoOp(Appel interne depuis ${CALLERID(all)} vers ${EXTEN}) [pbx_config]
2. Dial(SIP/${EXTEN},30,Tt) [pbx_config]

I do not understand why the 8000 extension could not contact ?

Thks in advance for your help.

Gad

It could be a problem with the AAstra itself.
What happens if You try to call 8000 from 8541 ? If this call gets through we may assume, that there’s a problem with the Aastra-configuration preventing You from handling two calls at the station in parallel (as 8000 and 8540 resides on the same Aastra handset).

It is the 8000 device that cannot be contacted. Does it exist in sip.conf and what does sip show users say?