Sponsor: VoiceMeUp - Corporate & Wholesale VoIP Services

VoIP Mailing List Archives
Mailing list archives for the VoIP community
 SearchSearch 

[asterisk-users] MWI problem with Siemens Gigaset S675 IP


 
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> Asterisk Users
View previous topic :: View next topic  
Author Message
jwinius at umrk.to
Guest





PostPosted: Wed Feb 13, 2008 1:45 pm    Post subject: [asterisk-users] MWI problem with Siemens Gigaset S675 IP Reply with quote

Hi list,

Before purchasing a number of Siemens DECT SIP phones, the Gigaset
S675 IP, I read that the problems with MWI had been fixed with the
latest firmware version (see
http://www.voip-info.org/wiki/view/Siemens+Gigaset+S675IP). Now I'm
not so sure that's the case.

After setting up a network mailbox for one of these phones, as well as
an Asterisk voicemail account (ext. 1000) in voicemail.conf's default
context, I added the following line to my phone's context in sip.conf:

mailbox=1000

However, soon after executing a 'sip reload' on the console, the
following error message will appear every three minutes:

[Feb 13 19:18:22] WARNING[14171]: chan_sip.c:12621 handle_response:
Remote host can't match request NOTIFY to call
'1742fb0c598dc6ec444a5ea64a2103ae at 192.168.10.10'. Giving up.

The IP address belongs to my server. At the same time, I used tcpdump
to see what else might be going on. I found the following:

19:18:22.540113 IP bitis.umrk.to.sip > gigaset.umrk.to.sip: SIP, length: 545
E..=.... at ......
.........)..NOTIFY sip:1000 at 192.168.10.5:5060 SIP/2.0
Via: SIP/2.0
19:18:22.571452 IP gigaset.umrk.to.sip > bitis.umrk.to.sip: SIP, length: 308
E..P.......f.......
.....<a_SIP/2.0 481 Call Leg/Transaction Does Not Exist
Via:

The latest comment on the voip-info.org page above outlines the same
problem. Can anyone here confirm that this is indeed a Siemens
problem, or might it be an Asterisk problem after all?

I'm running Asterisk v1.4.14 on a Debian etch server.

Thanks,

Jaap
Back to top
hdevito at mchsi.com
Guest





PostPosted: Wed Feb 13, 2008 3:58 pm    Post subject: [asterisk-users] MWI problem with Siemens Gigaset S675 IP Reply with quote

Try adding mailbox=1000 at default (or what ever your voicemail contexxt is)

I've had to add the voicemail context to get MWI to work correctly in the
past.
----- Original Message -----
From: "Jaap Winius" <jwinius at umrk.to>
To: <asterisk-users at lists.digium.com>
Sent: Wednesday, February 13, 2008 12:45 PM
Subject: [asterisk-users] MWI problem with Siemens Gigaset S675 IP
Quote:
Hi list,

Before purchasing a number of Siemens DECT SIP phones, the Gigaset
S675 IP, I read that the problems with MWI had been fixed with the
latest firmware version (see
http://www.voip-info.org/wiki/view/Siemens+Gigaset+S675IP). Now I'm
not so sure that's the case.

After setting up a network mailbox for one of these phones, as well as
an Asterisk voicemail account (ext. 1000) in voicemail.conf's default
context, I added the following line to my phone's context in sip.conf:

mailbox=1000

However, soon after executing a 'sip reload' on the console, the
following error message will appear every three minutes:

[Feb 13 19:18:22] WARNING[14171]: chan_sip.c:12621 handle_response:
Remote host can't match request NOTIFY to call
'1742fb0c598dc6ec444a5ea64a2103ae at 192.168.10.10'. Giving up.

The IP address belongs to my server. At the same time, I used tcpdump
to see what else might be going on. I found the following:

19:18:22.540113 IP bitis.umrk.to.sip > gigaset.umrk.to.sip: SIP, length:
545
E..=.... at ......
.........)..NOTIFY sip:1000 at 192.168.10.5:5060 SIP/2.0
Via: SIP/2.0
19:18:22.571452 IP gigaset.umrk.to.sip > bitis.umrk.to.sip: SIP, length:
308
E..P.......f.......
.....<a_SIP/2.0 481 Call Leg/Transaction Does Not Exist
Via:

The latest comment on the voip-info.org page above outlines the same
problem. Can anyone here confirm that this is indeed a Siemens
problem, or might it be an Asterisk problem after all?

I'm running Asterisk v1.4.14 on a Debian etch server.

Thanks,

Jaap

_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Back to top
jwinius at umrk.to
Guest





PostPosted: Wed Feb 13, 2008 4:52 pm    Post subject: [asterisk-users] MWI problem with Siemens Gigaset S675 IP Reply with quote

Quoting Henry Devito <hdevito at mchsi.com>:

Quote:
Try adding mailbox=1000 at default (or what ever your voicemail
contexxt is) I've had to add the voicemail context to get MWI
to work correctly in the past.

According to the documentation, you shouldn't have to add @<context>
if the context is 'default'. But, I went ahead and tried it out
anyway. I even tried using some other context names, but it makes no
difference: the error remains the same.

Thanks anyway,

Jaap
Back to top
steve.langstaff at cit...
Guest





PostPosted: Thu Feb 14, 2008 6:06 am    Post subject: [asterisk-users] MWI problem with Siemens Gigaset S675 IP Reply with quote

The "481 Call Leg/Transaction Does Not Exist" response to the
NOTIFY makes me think that you might need to configure the
phone to SUBSCRIBE to MWI - do you see any SUBSCRIBE messages
from the phone when it is booted?
Quote:
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of
Jaap Winius
Sent: 13 February 2008 18:46

Hi list,

Before purchasing a number of Siemens DECT SIP phones, the Gigaset
S675 IP, I read that the problems with MWI had been fixed
with the latest firmware version (see
http://www.voip-info.org/wiki/view/Siemens+Gigaset+S675IP).
Now I'm not so sure that's the case.

After setting up a network mailbox for one of these phones,
as well as an Asterisk voicemail account (ext. 1000) in
voicemail.conf's default context, I added the following line
to my phone's context in sip.conf:

mailbox=1000

However, soon after executing a 'sip reload' on the console,
the following error message will appear every three minutes:

[Feb 13 19:18:22] WARNING[14171]: chan_sip.c:12621 handle_response:
Remote host can't match request NOTIFY to call
'1742fb0c598dc6ec444a5ea64a2103ae at 192.168.10.10'. Giving up.

The IP address belongs to my server. At the same time, I used
tcpdump to see what else might be going on. I found the following:

19:18:22.540113 IP bitis.umrk.to.sip >
gigaset.umrk.to.sip: SIP, length: 545
E..=.... at ......
.........)..NOTIFY sip:1000 at 192.168.10.5:5060 SIP/2.0
Via: SIP/2.0
19:18:22.571452 IP gigaset.umrk.to.sip >
bitis.umrk.to.sip: SIP, length: 308
E..P.......f.......
.....<a_SIP/2.0 481 Call Leg/Transaction Does Not Exist
Via:

The latest comment on the voip-info.org page above outlines
the same problem. Can anyone here confirm that this is indeed
a Siemens problem, or might it be an Asterisk problem after all?

I'm running Asterisk v1.4.14 on a Debian etch server.
Back to top
jwinius at umrk.to
Guest





PostPosted: Thu Feb 14, 2008 11:10 am    Post subject: [asterisk-users] MWI problem with Siemens Gigaset S675 IP Reply with quote

Quoting Steve Langstaff <steve.langstaff at citel.com>:

Quote:
The "481 Call Leg/Transaction Does Not Exist" response to the
NOTIFY makes me think that you might need to configure the
phone to SUBSCRIBE to MWI - do you see any SUBSCRIBE messages
from the phone when it is booted?

Yeah, sure. And there are some error messages mixed in too:

==============================

14:01:23.425955 IP gigaset.umrk.to.sip > bitis.umrk.to.sip: SIP, length: 473
...........
........SUBSCRIBE sip:1000 at umrk.to SIP/2.0
Via: SIP/2.0/UDP 1
14:01:23.426075 IP bitis.umrk.to.sip > gigaset.umrk.to.sip: SIP, length: 509
E...-... at ..W...
...........vSIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.10.5
14:01:23.480238 IP gigaset.umrk.to.sip > bitis.umrk.to.sip: SIP, length: 634
E..........k.......
......F.SUBSCRIBE sip:1000 at umrk.to SIP/2.0
Via: SIP/2.0/UDP 1
14:01:23.480375 IP bitis.umrk.to.sip > gigaset.umrk.to.sip: SIP, length: 432
E...-... at ......
...........)SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.10.5:50
14:01:23.918830 arp who-has gigaset.umrk.to tell bitis.umrk.to
........../.E....
..........
14:01:23.921726 arp reply gigaset.umrk.to is-at 00:01:e3:77:f8:67 (oui
Unknown)
...........w.g....../.E....
......................
14:01:24.539636 IP gigaset.umrk.to.sip > bitis.umrk.to.sip: SIP, length: 476
E..................
......2gSUBSCRIBE sip:1000 at umrk.to SIP/2.0
Via: SIP/2.0/UDP 1
14:01:24.539816 IP bitis.umrk.to.sip > gigaset.umrk.to.sip: SIP, length: 512
E...-... at ..R...
...........ySIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.10.5
14:01:24.594442 IP gigaset.umrk.to.sip > bitis.umrk.to.sip: SIP, length: 634
E..........i.......
........SUBSCRIBE sip:1000 at umrk.to SIP/2.0
Via: SIP/2.0/UDP 1
14:01:24.594557 IP bitis.umrk.to.sip > gigaset.umrk.to.sip: SIP, length: 432
E...- .. at ......
...........)SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.10.5:50

==============================

Before this was a series of REGISTER messages, and afterwards a series
of OPTIONS messages. However, no errors there.

Also, this is without having set 'mailbox=1000' or 'mailbox=1000 at default' in
/etc/asterisk/sip.conf. And, now that I look at it again, the network
mailbox settings for the Siemens phone won't have anything to do with
these errors either, since it simply makes it possible to associate a
button on each handset with an extension used to access a voicemail
account.

Thanks,

Jaap
Back to top
Display posts from previous:   
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> Asterisk Users All times are GMT - 5 Hours
Page 1 of 1

 
Jump to:  
You cannot post new topics in this forum
You cannot reply to topics in this forum
You cannot edit your posts in this forum
You cannot delete your posts in this forum
You cannot vote in polls in this forum


Powered by phpBB © 2001, 2005 phpBB Group

VoiceMeUp - Corporate & Wholesale VoIP Services