2 Asterisk Sip Account to the same phone


#1

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.


#2

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


#3

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


#4

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).


#5

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