Incorrect watcherinfo package attribute

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Incorrect watcherinfo package attribute

Franz Edler-3
Hi,

I think I found a bug in presence module regarding the XML-body of the
NOTIFY request on watcher-information.

Sometimes (I guess at the update of watcher-information) the XML-body of the
NOTIFY request looks like:

<?xml version="1.0"?>
<watcherinfo xmlns="urn:ietf:params:xml:ns:watcherinfo" version="2"
state="partial">
  <watcher-list resource="sip:[hidden email]" package="presence.winfo">
    <watcher id="MzEzMmUwNDhlNDMxN2U2ZGJkZDcwNjMyZTExYjgxMmY."
event="subscribe" status="active">sip:[hidden email]</watcher>
  </watcher-list>
</watcherinfo>

The content of the package attribute of the watcher-list in above XML-body
is incorrect. According to RFC 3858 it has to be "presence" instead of
"presence.winfo". This issue leads to a "488 Not Acceptable Here" response
of the client.

I tested with OpenSIPS 1.5.1 (revision 5645).

Regards
Franz


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Re: Incorrect watcherinfo package attribute

Anca Vamanu-2
Hi Franz,

You are right. Thanks for analyzing and finding the problem. I will fix
that.

Regards,
Anca

Franz Edler wrote:

> Hi,
>
> I think I found a bug in presence module regarding the XML-body of the
> NOTIFY request on watcher-information.
>
> Sometimes (I guess at the update of watcher-information) the XML-body of the
> NOTIFY request looks like:
>
> <?xml version="1.0"?>
> <watcherinfo xmlns="urn:ietf:params:xml:ns:watcherinfo" version="2"
> state="partial">
>   <watcher-list resource="sip:[hidden email]" package="presence.winfo">
>     <watcher id="MzEzMmUwNDhlNDMxN2U2ZGJkZDcwNjMyZTExYjgxMmY."
> event="subscribe" status="active">sip:[hidden email]</watcher>
>   </watcher-list>
> </watcherinfo>
>
> The content of the package attribute of the watcher-list in above XML-body
> is incorrect. According to RFC 3858 it has to be "presence" instead of
> "presence.winfo". This issue leads to a "488 Not Acceptable Here" response
> of the client.
>
> I tested with OpenSIPS 1.5.1 (revision 5645).
>
> Regards
> Franz
>
>
> _______________________________________________
> Users mailing list
> [hidden email]
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>  


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Re: Incorrect watcherinfo package attribute

Anca Vamanu-2
In reply to this post by Franz Edler-3
Hi,

I have committed the fix, also in 1.5.x svn branch. The bug was only
when a change occured in an xcap document and the openxcap server
notified opensips presence server with refreshWatchers MI command.

Thanks,
Anca

Franz Edler wrote:

> Hi,
>
> I think I found a bug in presence module regarding the XML-body of the
> NOTIFY request on watcher-information.
>
> Sometimes (I guess at the update of watcher-information) the XML-body of the
> NOTIFY request looks like:
>
> <?xml version="1.0"?>
> <watcherinfo xmlns="urn:ietf:params:xml:ns:watcherinfo" version="2"
> state="partial">
>   <watcher-list resource="sip:[hidden email]" package="presence.winfo">
>     <watcher id="MzEzMmUwNDhlNDMxN2U2ZGJkZDcwNjMyZTExYjgxMmY."
> event="subscribe" status="active">sip:[hidden email]</watcher>
>   </watcher-list>
> </watcherinfo>
>
> The content of the package attribute of the watcher-list in above XML-body
> is incorrect. According to RFC 3858 it has to be "presence" instead of
> "presence.winfo". This issue leads to a "488 Not Acceptable Here" response
> of the client.
>
> I tested with OpenSIPS 1.5.1 (revision 5645).
>
> Regards
> Franz
>
>
> _______________________________________________
> Users mailing list
> [hidden email]
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>  


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Re: Incorrect watcherinfo package attribute

Franz Edler-3
In reply to this post by Franz Edler-3
Thanks for the fix.
I will test it soon and confirm.

regards
Franz

----- Original Message -----
From: "Anca Vamanu"
Date: 18.05.2009 11:13
To: "Franz Edler"
Hi,

I have committed the fix, also in 1.5.x svn branch. The bug was only
when a change occured in an xcap document and the openxcap server
notified opensips presence server with refreshWatchers MI command.

Thanks,
Anca

Franz Edler wrote:
> Hi,
>
> I think I found a bug in presence module regarding the XML-body of the
> NOTIFY request on watcher-information.
>
> Sometimes (I guess at the update of watcher-information) the XML-body of the
> NOTIFY request looks like:
>
>
>
> state="partial">
>
>
> event="subscribe" status="active">sip:[hidden email]

>

>

>
> The content of the package attribute of the watcher-list in above XML-body
> is incorrect. According to RFC 3858 it has to be "presence" instead of
> "presence.winfo". This issue leads to a "488 Not Acceptable Here" response
> of the client.
>
> I tested with OpenSIPS 1.5.1 (revision 5645).
>
> Regards
> Franz
>
>
> _______________________________________________
> Users mailing list
> [hidden email]
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>




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no ringback

Jinsong Hu
In reply to this post by Anca Vamanu-2
Hi, There:
  I am using opensips/kamailio in front of asterisk pool. my user register
on the opensips, and pstn call are routed out via asterisk.  what I find out
is that when the caller calls callee, some of the UA doesn't generate ring
back. for example, if I use xlite, the ring back works fine. but if I use
sipura 3000,
I don't hear anything until the callee picks up phone.
  I did a debug and found that after INVITE, I get 200 back, and then the UA
sends out ACK. the callee never sends 180 or 183 back to the caller UA. so
before the callee pick up phone, all the caller can hear is just silence.

  if my user registers directly on the asterisk, he can hear the ringback
because the Dial() command by default
will send ring back to the UA.

  How do I solve this problem in this case ? I searched all over internet
and don't see any body having any solution.

Jimmy


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opensips before asterisk, but there is no ring back when calling. how to solve this ?

Jinsong Hu
Hi, There:
  I am using opensips/kamailio in front of asterisk pool. my user register
on the opensips, and pstn call are routed out via asterisk.  what I find out
is that when the caller calls callee, some of the UA doesn't generate ring
back. for example, if I use xlite, the ring back works fine. but if I use
sipura 3000,
I don't hear anything until the callee picks up phone.
  I did a debug and found that after INVITE, I get 200 back, and then the UA
sends out ACK. the callee never sends 180 or 183 back to the caller UA. so
before the callee pick up phone, all the caller can hear is just silence.

  if my user registers directly on the asterisk, he can hear the ringback
because the Dial() command by default
will send ring back to the UA.

  How do I solve this problem in this case ? I searched all over internet
and don't see any body having any solution.

Jimmy





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Re: no ringback

Bogdan-Andrei Iancu
In reply to this post by Jinsong Hu
Hi Jimmy ,

There is a simple thing you can do:

 - just before relaying the INVITE the Asterisk, from OpenSIPS cfg, to a
sl_send_reply("180","ringing"); to fire a local 180 - of course this is
a bit bogus from logical perspective (as the end party does not actually
ring, so you force some information that you cannot check).

Regards,
Bogdan

Jinsong Hu wrote:

> Hi, There:
>   I am using opensips/kamailio in front of asterisk pool. my user register
> on the opensips, and pstn call are routed out via asterisk.  what I find out
> is that when the caller calls callee, some of the UA doesn't generate ring
> back. for example, if I use xlite, the ring back works fine. but if I use
> sipura 3000,
> I don't hear anything until the callee picks up phone.
>   I did a debug and found that after INVITE, I get 200 back, and then the UA
> sends out ACK. the callee never sends 180 or 183 back to the caller UA. so
> before the callee pick up phone, all the caller can hear is just silence.
>
>   if my user registers directly on the asterisk, he can hear the ringback
> because the Dial() command by default
> will send ring back to the UA.
>
>   How do I solve this problem in this case ? I searched all over internet
> and don't see any body having any solution.
>
> Jimmy
>
>
> _______________________________________________
> Users mailing list
> [hidden email]
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>  


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Re: no ringback

Jinsong Hu
Hi, Bogdan:
  Yes, I tried that . The phone rings once , and then keeps silent. I tried
to put more
180 there, but the phone still only rings once.
  If there is a solution that sends 180 periodically at 2 to 3 seconds
interval until the
callee answers, probably then it will work, but is there anyway to get this
done ?

Jinsong


----- Original Message -----
From: "Bogdan-Andrei Iancu" <[hidden email]>
To: "Jinsong Hu" <[hidden email]>
Cc: <[hidden email]>
Sent: Thursday, August 13, 2009 2:31 AM
Subject: Re: [OpenSIPS-Users] no ringback


> Hi Jimmy ,
>
> There is a simple thing you can do:
>
> - just before relaying the INVITE the Asterisk, from OpenSIPS cfg, to a
> sl_send_reply("180","ringing"); to fire a local 180 - of course this is a
> bit bogus from logical perspective (as the end party does not actually
> ring, so you force some information that you cannot check).
>
> Regards,
> Bogdan
>
> Jinsong Hu wrote:
>> Hi, There:
>>   I am using opensips/kamailio in front of asterisk pool. my user
>> register on the opensips, and pstn call are routed out via asterisk.
>> what I find out is that when the caller calls callee, some of the UA
>> doesn't generate ring back. for example, if I use xlite, the ring back
>> works fine. but if I use sipura 3000,
>> I don't hear anything until the callee picks up phone.
>>   I did a debug and found that after INVITE, I get 200 back, and then the
>> UA sends out ACK. the callee never sends 180 or 183 back to the caller
>> UA. so before the callee pick up phone, all the caller can hear is just
>> silence.
>>
>>   if my user registers directly on the asterisk, he can hear the ringback
>> because the Dial() command by default
>> will send ring back to the UA.
>>
>>   How do I solve this problem in this case ? I searched all over internet
>> and don't see any body having any solution.
>>
>> Jimmy
>>
>> _______________________________________________
>> Users mailing list
>> [hidden email]
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
>


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Re: no ringback

Bogdan-Andrei Iancu
Hi Jinsong,

As per RFC3261, a single 180 Ringing is enough to switch the caller
device to ringing state and keep it so until other higher reply code is
received.

Can you check at signalling level if something else (other reply) is
sent after the 180 ? Maybe there is something else in there.

Regards,
Bogdan

Jinsong Hu wrote:

> Hi, Bogdan:
>  Yes, I tried that . The phone rings once , and then keeps silent. I
> tried to put more
> 180 there, but the phone still only rings once.
>  If there is a solution that sends 180 periodically at 2 to 3 seconds
> interval until the
> callee answers, probably then it will work, but is there anyway to get
> this done ?
>
> Jinsong
>
>
> ----- Original Message ----- From: "Bogdan-Andrei Iancu"
> <[hidden email]>
> To: "Jinsong Hu" <[hidden email]>
> Cc: <[hidden email]>
> Sent: Thursday, August 13, 2009 2:31 AM
> Subject: Re: [OpenSIPS-Users] no ringback
>
>
>> Hi Jimmy ,
>>
>> There is a simple thing you can do:
>>
>> - just before relaying the INVITE the Asterisk, from OpenSIPS cfg, to
>> a sl_send_reply("180","ringing"); to fire a local 180 - of course
>> this is a bit bogus from logical perspective (as the end party does
>> not actually ring, so you force some information that you cannot check).
>>
>> Regards,
>> Bogdan
>>
>> Jinsong Hu wrote:
>>> Hi, There:
>>>   I am using opensips/kamailio in front of asterisk pool. my user
>>> register on the opensips, and pstn call are routed out via asterisk.
>>> what I find out is that when the caller calls callee, some of the UA
>>> doesn't generate ring back. for example, if I use xlite, the ring
>>> back works fine. but if I use sipura 3000,
>>> I don't hear anything until the callee picks up phone.
>>>   I did a debug and found that after INVITE, I get 200 back, and
>>> then the UA sends out ACK. the callee never sends 180 or 183 back to
>>> the caller UA. so before the callee pick up phone, all the caller
>>> can hear is just silence.
>>>
>>>   if my user registers directly on the asterisk, he can hear the
>>> ringback because the Dial() command by default
>>> will send ring back to the UA.
>>>
>>>   How do I solve this problem in this case ? I searched all over
>>> internet and don't see any body having any solution.
>>>
>>> Jimmy
>>>
>>> _______________________________________________
>>> Users mailing list
>>> [hidden email]
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>
>>
>
>


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