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Summary:ASTERISK-06228: [patch] MWI Subsription not working on AVM Boxes
Reporter:dlu (dlu)Labels:
Date Opened:2006-02-01 09:29:21.000-0600Date Closed:2006-03-27 22:22:26.000-0600
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/Subscriptions
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:
Description:If a register an AVM box details in sip-debug the registration is fine and the subscription of the MWI is fine too. After a message arrive the NOTIFY will Fail with 481 Calleg down not Exist. I remember with older version of firmware it was working but since a few month i guess it still isnt working.

So MWI on asterisk side works with Snom Telephones as example well. Only AVM Boxes are involved.


****** ADDITIONAL INFORMATION ******

The register with subscribe

<-- SIP read from 192.168.0.215:5060:
REGISTER sip:domain.com SIP/2.0
Via: SIP/2.0/UDP 192.168.0.215:5060;branch=z9hG4bK94EE830FEF1FAC5B03F9F591140B4
From: <sip:123456@domain.com>;tag=233228002
To: <sip:123456@domain.com>
Call-ID: DCAAF7698619DB7766ED8FB42162D@192.168.0.215
CSeq: 1 REGISTER
Max-Forwards: 70
User-Agent: AVM FRITZ!Box Fon WLAN 08.03.99 (Jan 25 2006)
Supported: 100rel, replaces
Allow-Events: telephone-event, refer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, UPDATE, PRACK, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE
Accept: application/sdp, multipart/mixed
Accept-Encoding: identity
Content-Length: 0

--- (14 headers 0 lines)---
Using latest REGISTER request as basis request
Sending to 192.168.0.215 : 5060 (non-NAT)
Transmitting (no NAT) to 192.168.0.215:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.215:5060;branch=z9hG4bK94EE830FEF1FAC5B03F9F591140B4;received=192.168.0.215
From: <sip:123456@domain.com>;tag=233228002
To: <sip:123456@domain.com>
Call-ID: DCAAF7698619DB7766ED8FB42162D@192.168.0.215
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:123456@192.168.0.1>
Content-Length: 0


---
Transmitting (no NAT) to 192.168.0.215:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.215:5060;branch=z9hG4bK94EE830FEF1FAC5B03F9F591140B4;received=192.168.0.215
From: <sip:123456@domain.com>;tag=233228002
To: <sip:123456@domain.com>;tag=as251c7057
Call-ID: DCAAF7698619DB7766ED8FB42162D@192.168.0.215
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:123456@192.168.0.1>
WWW-Authenticate: Digest realm="domain.com", nonce="5c6f696c"
Content-Length: 0


---
Scheduling destruction of call 'DCAAF7698619DB7766ED8FB42162D@192.168.0.215' in 15000 ms
node1*CLI>
<-- SIP read from 192.168.0.215:5060:
REGISTER sip:domain.com SIP/2.0
Via: SIP/2.0/UDP 192.168.0.215:5060;branch=z9hG4bKC507A4333EBBEC7C18BFE987F24BA
From: <sip:123456@domain.com>;tag=233228002
To: <sip:123456@domain.com>
Call-ID: DCAAF7698619DB7766ED8FB42162D@192.168.0.215
CSeq: 2 REGISTER
Authorization: Digest username="123456", realm="domain.com", nonce="5c6f696c", uri="sip:domain.com", response="a9d9025b7e8b77bf065c6419c8eb0fe4"
Max-Forwards: 70
User-Agent: AVM FRITZ!Box Fon WLAN 08.03.99 (Jan 25 2006)
Supported: 100rel, replaces
Allow-Events: telephone-event, refer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, UPDATE, PRACK, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE
Accept: application/sdp, multipart/mixed
Accept-Encoding: identity
Content-Length: 0

--- (15 headers 0 lines)---
Using latest REGISTER request as basis request
Sending to 192.168.0.215 : 5060 (non-NAT)
Transmitting (no NAT) to 192.168.0.215:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.215:5060;branch=z9hG4bKC507A4333EBBEC7C18BFE987F24BA;received=192.168.0.215
From: <sip:123456@domain.com>;tag=233228002
To: <sip:123456@domain.com>
Call-ID: DCAAF7698619DB7766ED8FB42162D@192.168.0.215
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:123456@192.168.0.1>
Content-Length: 0


---
Transmitting (no NAT) to 192.168.0.215:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.215:5060;branch=z9hG4bKC507A4333EBBEC7C18BFE987F24BA;received=192.168.0.215
From: <sip:123456@domain.com>;tag=233228002
To: <sip:123456@domain.com>;tag=as251c7057
Call-ID: DCAAF7698619DB7766ED8FB42162D@192.168.0.215
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Expires: 0
Date: Wed, 01 Feb 2006 15:56:01 GMT
Content-Length: 0


---
Scheduling destruction of call 'DCAAF7698619DB7766ED8FB42162D@192.168.0.215' in 15000 ms
node1*CLI>
<-- SIP read from 192.168.0.215:5060:
REGISTER sip:domain.com SIP/2.0
Via: SIP/2.0/udp 192.168.0.215:5060;branch=z9hG4bKFBCD1ADEE77CF0764B4AC3F5A8840
From: <sip:123456@domain.com>;tag=233228002
To: <sip:123456@domain.com>
Call-ID: DCAAF7698619DB7766ED8FB42162D@192.168.0.215
CSeq: 3 REGISTER
Contact: <sip:123456@192.168.0.215;uniq=F4D44036E98EF2AC112C835DC4495>
Authorization: Digest username="123456", realm="domain.com", nonce="5c6f696c", uri="sip:domain.com", response="a9d9025b7e8b77bf065c6419c8eb0fe4"
Expires: 1800
Max-Forwards: 70
User-Agent: AVM FRITZ!Box Fon WLAN 08.03.99 (Jan 25 2006)
Supported: 100rel, replaces
Allow-Events: telephone-event, refer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, UPDATE, PRACK, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE
Accept: application/sdp, multipart/mixed
Accept-Encoding: identity
Content-Length: 0

--- (17 headers 0 lines)---
Using latest REGISTER request as basis request
Sending to 192.168.0.215 : 5060 (non-NAT)
Transmitting (no NAT) to 192.168.0.215:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/udp 192.168.0.215:5060;branch=z9hG4bKFBCD1ADEE77CF0764B4AC3F5A8840;received=192.168.0.215
From: <sip:123456@domain.com>;tag=233228002
To: <sip:123456@domain.com>
Call-ID: DCAAF7698619DB7766ED8FB42162D@192.168.0.215
CSeq: 3 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:123456@192.168.0.1>
Content-Length: 0


---
   -- Registered SIP '123456' at 192.168.0.215 port 5060 expires 1800
Transmitting (no NAT) to 192.168.0.215:5060:
SIP/2.0 200 OK
Via: SIP/2.0/udp 192.168.0.215:5060;branch=z9hG4bKFBCD1ADEE77CF0764B4AC3F5A8840;received=192.168.0.215
From: <sip:123456@domain.com>;tag=233228002
To: <sip:123456@domain.com>;tag=as251c7057
Call-ID: DCAAF7698619DB7766ED8FB42162D@192.168.0.215
CSeq: 3 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Expires: 1800
Contact: <sip:123456@192.168.0.215;uniq=F4D44036E98EF2AC112C835DC4495>;expires=1800
Date: Wed, 01 Feb 2006 15:56:01 GMT
Content-Length: 0


---
Scheduling destruction of call 'DCAAF7698619DB7766ED8FB42162D@192.168.0.215' in 15000 ms
node1*CLI>
<-- SIP read from 192.168.0.215:5060:

--- (0 headers 0 lines) Nat keepalive ---
node1*CLI>
<-- SIP read from 192.168.0.215:5060:
SUBSCRIBE sip:123456@domain.com SIP/2.0
Via: SIP/2.0/udp 192.168.0.215:5060;branch=z9hG4bKDA2573175771BCE402FBD7F580598
From: <sip:123456@domain.com>;tag=2517038591
To: <sip:123456@domain.com>
Call-ID: 72AC97C6B0772E9DB34BD80FF4E7C@192.168.0.215
CSeq: 4 SUBSCRIBE
Contact: <sip:123456@192.168.0.215;uniq=F4D44036E98EF2AC112C835DC4495>
Event: message-summary
Expires: 3600
Max-Forwards: 70
User-Agent: AVM FRITZ!Box Fon WLAN 08.03.99 (Jan 25 2006)
Allow: NOTIFY
Accept: application/simple-message-summary
Content-Length: 0

--- (14 headers 0 lines)---
Using latest SUBSCRIBE request as basis request
Sending to 192.168.0.215 : 5060 (non-NAT)
Found peer '123456'
Transmitting (no NAT) to 192.168.0.215:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/udp 192.168.0.215:5060;branch=z9hG4bKDA2573175771BCE402FBD7F580598;received=192.168.0.215
From: <sip:123456@domain.com>;tag=2517038591
To: <sip:123456@domain.com>;tag=as1933b5cb
Call-ID: 72AC97C6B0772E9DB34BD80FF4E7C@192.168.0.215
CSeq: 4 SUBSCRIBE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:123456@192.168.0.1>
WWW-Authenticate: Digest realm="domain.com", nonce="3b7692d4"
Content-Length: 0


---
Scheduling destruction of call '72AC97C6B0772E9DB34BD80FF4E7C@192.168.0.215' in 15000 ms
node1*CLI>
<-- SIP read from 192.168.0.215:5060:
SUBSCRIBE sip:123456@domain.com SIP/2.0
Via: SIP/2.0/udp 192.168.0.215:5060;branch=z9hG4bK1910B9BC4B2E9B30584DCD1626EC7
From: <sip:123456@domain.com>;tag=1949280590
To: <sip:123456@domain.com>
Call-ID: 72AC97C6B0772E9DB34BD80FF4E7C@192.168.0.215
CSeq: 5 SUBSCRIBE
Contact: <sip:123456@192.168.0.215;uniq=F4D44036E98EF2AC112C835DC4495>
Authorization: Digest username="123456", realm="domain.com", nonce="3b7692d4", uri="sip:123456@domain.com", response="306c2ce4d2a75ae1df5e7b2da1cc6520"
Event: message-summary
Expires: 3600
Max-Forwards: 70
User-Agent: AVM FRITZ!Box Fon WLAN 08.03.99 (Jan 25 2006)
Allow: NOTIFY
Accept: application/simple-message-summary
Content-Length: 0

--- (15 headers 0 lines)---
Using latest SUBSCRIBE request as basis request
Sending to 192.168.0.215 : 5060 (non-NAT)
Found peer '123456'
Transmitting (no NAT) to 192.168.0.215:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/udp 192.168.0.215:5060;branch=z9hG4bK1910B9BC4B2E9B30584DCD1626EC7;received=192.168.0.215
From: <sip:123456@domain.com>;tag=1949280590
To: <sip:123456@domain.com>;tag=as02422de1
Call-ID: 72AC97C6B0772E9DB34BD80FF4E7C@192.168.0.215
CSeq: 5 SUBSCRIBE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:123456@192.168.0.1>
WWW-Authenticate: Digest realm="domain.com", nonce="72a2b92b"
Content-Length: 0


---
Scheduling destruction of call '72AC97C6B0772E9DB34BD80FF4E7C@192.168.0.215' in 15000 ms
12 headers, 3 lines
Reliably Transmitting (no NAT) to 192.168.0.215:5060:
NOTIFY sip:123456@192.168.0.215;uniq=F4D44036E98EF2AC112C835DC4495 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK32062169;rport
From: "asterisk" <sip:asterisk@dus.net>;tag=as76c40e3f
To: <sip:123456@192.168.0.215;uniq=F4D44036E98EF2AC112C835DC4495>
Contact: <sip:asterisk@192.168.0.1>
Call-ID: 7d1536ec7361484433d05c025ebef35f@dus.net
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 84

Messages-Waiting: no
Message-Account: sip:vmail@dus.net
Voice-Message: 0/0 (0/0)

---
Scheduling destruction of call '7d1536ec7361484433d05c025ebef35f@dus.net' in 15000 ms
node1*CLI>
<-- SIP read from 192.168.0.215:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK32062169;rport=5060
From: "asterisk" <sip:asterisk@dus.net>;tag=as76c40e3f
To: <sip:123456@192.168.0.215;uniq=F4D44036E98EF2AC112C835DC4495>;tag=A04427F1E48798547417465D9B1EA
Call-ID: 7d1536ec7361484433d05c025ebef35f@dus.net
CSeq: 102 NOTIFY
User-Agent: AVM FRITZ!Box Fon WLAN 08.03.99 (Jan 25 2006)
Content-Length: 0                                                                                                                                                                                              


after a call produce a message the following happen



Reliably Transmitting (no NAT) to 192.168.0.215:5060:
NOTIFY sip:123456@192.168.0.215;uniq=F4D44036E98EF2AC112C835DC4495 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK5278eef4;rport
From: "asterisk" <sip:asterisk@dus.net>;tag=as609a0dbd
To: <sip:123456@192.168.0.215;uniq=F4D44036E98EF2AC112C835DC4495>
Contact: <sip:asterisk@192.168.0.1>
Call-ID: 462e5a0d04537a991f640e7e04ebb342@dus.net
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 85

Messages-Waiting: yes
Message-Account: sip:vmail@dus.net
Voice-Message: 1/0 (0/0)

---
Scheduling destruction of call '462e5a0d04537a991f640e7e04ebb342@dus.net' in 15000 ms
node1*CLI>
<-- SIP read from 192.168.0.215:5060:
SIP/2.0 481 Call Leg/Transaction Does Not Exist
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK5278eef4;rport=5060
From: "asterisk" <sip:asterisk@dus.net>;tag=as609a0dbd
To: <sip:123456@192.168.0.215;uniq=F4D44036E98EF2AC112C835DC4495>
Call-ID: 462e5a0d04537a991f640e7e04ebb342@dus.net
CSeq: 102 NOTIFY
User-Agent: AVM FRITZ!Box Fon WLAN 08.03.99 (Jan 25 2006)
Content-Length: 0




Comments:By: Serge Vecher (serge-v) 2006-02-01 10:05:45.000-0600

dlu: please always attach logs as _attachments_!
By: Tilghman Lesher (tilghman) 2006-02-01 10:19:23.000-0600

So why do you believe this is an issue with Asterisk and not an issue with AVM?
By: dlu (dlu) 2006-02-01 10:40:43.000-0600

@Corydon

I am not the real sip guru. But the box works well with other systems then asterisk. But i need to bring the avm box to work with it.

Any differences are still the because it isnt working.
By: Olle Johansson (oej) 2006-02-01 11:49:59.000-0600

The AVM box is totally correct. Asterisk does not support subscriptions of MWI status and when we notify, we use a random call iD, which leads to the missing call leg message.

This is the way the Asterisk implementation currently works, and it is supported by most devices even though it is not standard compliant.

If you can't configure the AVM box to *not* subscribe for MWI and accept Asterisks MWI messages anyway, I suggest you set up a bounty or contract someone to fix this. It's not a one-line fix.

I will categorize this as a feature request.
By: dlu (dlu) 2006-02-02 00:56:08.000-0600

Hi oje,

i dont agree with you that its a feature request. If asterisk support MWI it should be correct. I understand its perhaps quit complicated to implement but that are many things in asterisk. So i will be happy if you categorize it as todo and not as feature request.

Asterisk must be in the line of rfc to be compatible with many hardware it comes in the future. So its one of the main goals to fit in rfc´s
By: Olle Johansson (oej) 2006-02-02 01:04:45.000-0600

I do understand you. But since subscriptions for MWI is a feature we never claimed to support, it is not a bug, mut a missing implementation that surely is on the to-do list for reaching RFC compatibility, alongside with a lot of other missing features.

Since the list is long, the usual open source principle still applies: If you need a new feature badly, code it yourself or find someone that can do it for you.

Very few phones request this and accept the non-RFC way we do MWI notification today, which is a reason why this problem hasn't been a priority to fix.

I am *not* saying you are wrong, just trying to explain what will happen. This is a known issue since a long time, and no one has prioritized to pay a developer to fix it or fixed it themselves... yet.

If you search, I think there are propably one or two bug reports in this topic in the bug tracker archives and all was solved by changing the configuration of the phone.
By: dlu (dlu) 2006-02-02 02:13:00.000-0600

Hello oej,

yes. I understand the way asterisk went. We can discuss a long time about the need  and sense of features in asterisk. I love asterisk because its a great work and is established as a stable mediagateway and pbx. But imho its time to force the reaching rfc compatibillity prior adding many features they are not.

I thank you for your answers and i hope to fix the part of eventhandling with a couple of developer.
By: Olle Johansson (oej) 2006-02-02 02:28:39.000-0600

I fully agree that we need to be more RFC compatible, it has been my goal for a long time and is the reason why we are starting the work on a new SIp channel. It's just a matter of priorities on where to start.
By: Olle Johansson (oej) 2006-02-02 02:29:30.000-0600

Please feel free to contact me on IRC or over e-mail if you need advice on how this can be implemented.
By: Max (tintin) 2006-02-08 12:39:43.000-0600

dlu,

did you ever contact AVM about this? It should be in their best interest to support the "Asterisk way", as quite a few VoIP providers use Asterisk and their customers will be happy to receive MWI via AVM boxes. I am going to send this to them as a suggestion.

I'm very interested in finding a solution for this problem as well.
By: Olle Johansson (oej) 2006-02-21 13:36:10.000-0600

Working on this in the "subscribemwi" branch
By: Olle Johansson (oej) 2006-02-28 09:41:35.000-0600

Fixed in the "subscribemwi" branch.
By: dlu (dlu) 2006-02-28 09:43:10.000-0600

The subscribemwi branch from oje works perfect. It also works now on Grandstream Phones and ATAs.

Great work dude.

We are now a little closer to RCF3261 now.
By: Olle Johansson (oej) 2006-03-27 22:21:41.000-0600

Committed to svn trunk. Thanks!
By: Olle Johansson (oej) 2006-03-27 22:22:25.000-0600

Removing branches.

Revision 15476