Re: [asterisk-users] SIP Simple support on Asterisk 11
Eloi, My responses are inline. Thanks a lot for this detailed answer : You're welcome. Thank you for responding. A lot of people forget to do so and future list readers are left wondering whether or not the proposed solution worked. - I managed to have it working disabling auth message request : auth_message_requests = no in sip.conf - pedantic=no does not resolve the issue - reenabling auth_message_requests = yes and removing pedantic option, your patch in chan_sip resolves the issues ! As it looks like pidgin has an issue, I guess that we can use it as a workaround. I'm glad my patch worked but keep in mind that it really is a workaround, because it will cause actual retransmits of MESSAGE requests to be treated as new requests. This may not cause you any issues, but the root problem should still be addressed by submitting a report to the Pidgin bug tracker [1]. It's a straightforward problem so if you provide a link to your original post [2] the developers should be able to resolve it quickly. I would like know to enable presence notification between each users. To fulfill it, I am using http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/DeviceStates_id265377.html Am I doing it in a good way ? Yes, but there is a 4th edition of Asterisk: The Definitive Guide available in the Open Feedback Publishing System [3] that is focused on documenting Asterisk 11. [1] https://developer.pidgin.im/wiki/TipsForBugReports [2] http://lists.digium.com/pipermail/asterisk-users/2013-June/279569.html [3] http://ofps.oreilly.com/titles/9781449332426/asterisk-DeviceStates.html Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Simple support on Asterisk 11
Eloi Bail wrote:
I am trying to enable SIP SIMPLE communication in my test environment.
I have the following env :
- one server (192.168.50.126) with Asterisk 11
- 2 clients using pidgin : demo-bob and demo-alice on my 192.168.50.143
I successfully had a phone call between clients.
I used the following link to enable SIMPLE messaging between my clients :
http://highsecurity.blogspot.ca/2012/03/asterisk-10-110-sms-messaging-or-sip.html
Both users managed to register.
Adding verbose on the server, I have the following traces when I send the
message MESSAGE FROM ALICE TO BOB from demo-alice to demo-bob
http://paste.fedoraproject.org/19489/37158861/
As you can see I succeed to have the message sent from alice to Asterisk.
When the server is trying to transmitting, I see a 401 error message.
According to this post ( http://forums.digium.com/viewtopic.php?f=1t=72814 )
the first 401 should be normal as authentication is requested.
Afterwards the server emit 202 message.
But demo-bob never receives a message.
I ran wireshark on server and client. It confirms that no message is sent from
Asterisk to demo-bob.
Could you please give me advice ?
Here are my extensions.conf and sip.conf according to the link I mentioned.
http://paste.fedoraproject.org/19626/16493741/
http://paste.fedoraproject.org/19627/49423137/
Eloi,
The trace shows that the initial MESSAGE from Alice does not include an
Authorization header so Asterisk responds with a 401 Unauthorized. Alice then
replies with a MESSAGE with an Authorization header, but reuses the same CSeq
header (CSeq: 6 MESSAGE) which causes Asterisk to ignore it as a retransmit:
[Jun 18 16:49:35] DEBUG[16266] chan_sip.c: Ignoring SIP message because of
retransmit (MESSAGE Seqno 6, ours 6)
I believe this is a bug in Pidgin because RFC 3261 [1] states:
CSeq or Command Sequence contains an integer and a method name. The
CSeq number is incremented for each new request within a dialog and
is a traditional sequence number.
...
Requests within a dialog MUST contain strictly monotonically
increasing and contiguous CSeq sequence numbers (increasing-by-one)
in each direction (excepting ACK and CANCEL of course, whose numbers
equal the requests being acknowledged or cancelled).
However, there is also a similar issue [2] that can be worked around by setting
pedantic=no in sip.conf. If that doesn't work, you can give the following
(untested) patch to chan_sip.c a try:
--- chan_sip.c.orig 2013-06-19 11:44:38.0 -0400
+++ chan_sip.c 2013-06-19 11:47:22.0 -0400
@@ -28078,6 +28078,7 @@
} else if (p-icseq
p-icseq == seqno
req-method != SIP_ACK
+ p-method != SIP_MESSAGE
(p-method != SIP_CANCEL || p-alreadygone)) {
/* ignore means don't do anything with it but still have to
respond appropriately. We do this if we receive a repeat of
Good luck and please let the list know how this works out.
[1] http://www.ietf.org/rfc/rfc3261.txt
[2] https://issues.asterisk.org/jira/browse/ASTERISK-19139
Regards,
Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer
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Re: [asterisk-users] SIP Simple support on Asterisk 11
On Wednesday, June 19, 2013 11:11:17 AM Matthew J. Roth wrote: Eloi Bail wrote: I am trying to enable SIP SIMPLE communication in my test environment. I use the following which semi-enables message broadcasting to multiple devices so a user who receives a message can reply from any of the devices. http://messinet.com/trac/wiki/Asterisk/Message -A -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Simple support on Asterisk 11
Hi,
Thanks a lot for this detailed answer :
- I managed to have it working disabling auth message request
: auth_message_requests = no in sip.conf
- pedantic=no does not resolve the issue
- reenabling auth_message_requests = yes and removing pedantic option,
your patch in chan_sip resolves the issues !
As it looks like pidgin has an issue, I guess that we can use it as a
workaround.
I would like know to enable presence notification between each users. To
fulfill it, I am using
http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/DeviceStates_id265377.html
Am I doing it in a good way ?
Thanks !
Eloi
On Wed, Jun 19, 2013 at 12:11 PM, Matthew J. Roth mr...@imminc.com wrote:
Eloi Bail wrote:
I am trying to enable SIP SIMPLE communication in my test environment.
I have the following env :
- one server (192.168.50.126) with Asterisk 11
- 2 clients using pidgin : demo-bob and demo-alice on my 192.168.50.143
I successfully had a phone call between clients.
I used the following link to enable SIMPLE messaging between my clients :
http://highsecurity.blogspot.ca/2012/03/asterisk-10-110-sms-messaging-or-sip.html
Both users managed to register.
Adding verbose on the server, I have the following traces when I send the
message MESSAGE FROM ALICE TO BOB from demo-alice to demo-bob
http://paste.fedoraproject.org/19489/37158861/
As you can see I succeed to have the message sent from alice to Asterisk.
When the server is trying to transmitting, I see a 401 error message.
According to this post (
http://forums.digium.com/viewtopic.php?f=1t=72814 )
the first 401 should be normal as authentication is requested.
Afterwards the server emit 202 message.
But demo-bob never receives a message.
I ran wireshark on server and client. It confirms that no message is
sent from
Asterisk to demo-bob.
Could you please give me advice ?
Here are my extensions.conf and sip.conf according to the link I
mentioned.
http://paste.fedoraproject.org/19626/16493741/
http://paste.fedoraproject.org/19627/49423137/
Eloi,
The trace shows that the initial MESSAGE from Alice does not include an
Authorization header so Asterisk responds with a 401 Unauthorized. Alice
then
replies with a MESSAGE with an Authorization header, but reuses the same
CSeq
header (CSeq: 6 MESSAGE) which causes Asterisk to ignore it as a
retransmit:
[Jun 18 16:49:35] DEBUG[16266] chan_sip.c: Ignoring SIP message because
of
retransmit (MESSAGE Seqno 6, ours 6)
I believe this is a bug in Pidgin because RFC 3261 [1] states:
CSeq or Command Sequence contains an integer and a method name. The
CSeq number is incremented for each new request within a dialog and
is a traditional sequence number.
...
Requests within a dialog MUST contain strictly monotonically
increasing and contiguous CSeq sequence numbers (increasing-by-one)
in each direction (excepting ACK and CANCEL of course, whose numbers
equal the requests being acknowledged or cancelled).
However, there is also a similar issue [2] that can be worked around by
setting
pedantic=no in sip.conf. If that doesn't work, you can give the
following
(untested) patch to chan_sip.c a try:
--- chan_sip.c.orig 2013-06-19 11:44:38.0 -0400
+++ chan_sip.c 2013-06-19 11:47:22.0 -0400
@@ -28078,6 +28078,7 @@
} else if (p-icseq
p-icseq == seqno
req-method != SIP_ACK
+ p-method != SIP_MESSAGE
(p-method != SIP_CANCEL || p-alreadygone)) {
/* ignore means don't do anything with it but still have
to
respond appropriately. We do this if we receive a
repeat of
Good luck and please let the list know how this works out.
[1] http://www.ietf.org/rfc/rfc3261.txt
[2] https://issues.asterisk.org/jira/browse/ASTERISK-19139
Regards,
Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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