[asterisk-dev] asterisk-dev Digest, Vol 47, Issue 17
bilal ghayyad
bilmar_gh at yahoo.com
Fri Apr 3 17:58:49 CDT 2009
Hi Raj;
Where is the below parameters can be found in the sip.conf? I did not find them and my asterisk is 1.4.19.2:
session-timers=originate
session-expires=600
session-minse=90
session-refresher=uas
Also, I think in my case, maybe these parameters does not help me? Because maybe I need something related to rtp, as we are talking about: how to disconnect the call if internet was disconnected, so the hangup has not been detected?
Regards
Bilal
> On Wed, Jun 4, 2008 at 12:12 PM, bilal ghayyad
> <bilmar_gh at yahoo.com> wrote:
> > And in case of SIP, I heared it is existed but I have
> > to check the session timer in the trunk version of the
> > sip_chan, but really I do not know how to check this
> > and wether there is any need to be done on the source
> > to be modified and compiled to obtain it successfully
> > working.
>
> There is no need to modify/compile the source code for
> this.
> Session-timers are configured in sip.conf. There are four
> flags that
> control the operation of this feature:
>
> session-timers=originate
> session-expires=600
> session-minse=90
> session-refresher=uas
>
> You can either set them globally or at a per user/peer
> level. The
> value of "originate" for session-timers flag
> means that Asterisk will
> actively request session-timers support from the other
> end-point. If
> the other end-point supports session-timers then the two
> will
> negotiate the frequency at which session will be refreshed
> and which
> side will initiate the refresh requests. In the
> "originate" mode even
> if the other end-point doesn't support this feature,
> Asterisk will
> refresh the session periodically anyway.
>
> The session-expires and session-mine are high and low
> watermarks for
> how tolerable Asterisk will be to the frequency of session
> refreshes.
> Feel free to tune these to what's more suitable in your
> environment.
>
> --
> Raj Jain
>
>
>
> ------------------------------
>
> Message: 2
> Date: Wed, 04 Jun 2008 12:54:23 -0500
> From: Russell Bryant <russell at digium.com>
> Subject: Re: [asterisk-dev] [policy] Discussion on IRC -
> how to make
> -dev more useful
> To: Asterisk Developers Mailing List
> <asterisk-dev at lists.digium.com>
> Message-ID: <4846D6CF.7040501 at digium.com>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> Tilghman Lesher wrote:
> > On Wednesday 04 June 2008 09:53:36 Matthew Fredrickson
> wrote:
> >> Not to be Mr. Negativity here, but I concur as
> well. Asterisk-dev is
> >> not a high traffic list at all (LKML is a great
> example of a high
> >> traffic list), and any solution that puts the load
> on the posters is
> >> going to have major trouble gaining traction.
> >
> > People who don't want to call extra attention to
> their proposals are free
> > to ignore it. Those of us who _want_ to draw
> attention and discussion from
> > the list will use the tags. Quite simple, really. I
> think I've already used
> > the [policy] tag once before, and it's nice to
> have a formal proposal, so that
> > we all know what the various tags mean.
>
> I agree with Tilghman.
>
> I don't think these tags should be a requirement, but I
> think they are a
> welcome addition to common practice for this list. If a
> poster doesn't
> want to use them, fine, it doesn't really matter.
> However, having more
> clearly defined subjects certainly isn't going to hurt
> anything.
>
> --
> Russell Bryant
> Senior Software Engineer
> Open Source Team Lead
> Digium, Inc.
>
>
>
> ------------------------------
>
> Message: 3
> Date: Wed, 04 Jun 2008 13:50:28 -0500
> From: Matthew Fredrickson <creslin at digium.com>
> Subject: Re: [asterisk-dev] [policy] Discussion on IRC -
> how to make
> -dev more useful
> To: Asterisk Developers Mailing List
> <asterisk-dev at lists.digium.com>
> Message-ID: <4846E3F4.8050605 at digium.com>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> Russell Bryant wrote:
> > Tilghman Lesher wrote:
> >> On Wednesday 04 June 2008 09:53:36 Matthew
> Fredrickson wrote:
> >>> Not to be Mr. Negativity here, but I concur as
> well. Asterisk-dev is
> >>> not a high traffic list at all (LKML is a
> great example of a high
> >>> traffic list), and any solution that puts the
> load on the posters is
> >>> going to have major trouble gaining traction.
> >> People who don't want to call extra attention
> to their proposals are free
> >> to ignore it. Those of us who _want_ to draw
> attention and discussion from
> >> the list will use the tags. Quite simple, really.
> I think I've already used
> >> the [policy] tag once before, and it's nice to
> have a formal proposal, so that
> >> we all know what the various tags mean.
> >
> > I agree with Tilghman.
> >
> > I don't think these tags should be a requirement,
> but I think they are a
> > welcome addition to common practice for this list. If
> a poster doesn't
> > want to use them, fine, it doesn't really matter.
> However, having more
> > clearly defined subjects certainly isn't going to
> hurt anything.
>
> As long as we don't set a policy that makes
> communication more
> difficult, I have no problems with it. So as long as it is
> not a
> requirement, that sounds pretty reasonable.
>
> --
> Matthew Fredrickson
> Software/Firmware Engineer
> Digium, Inc.
>
>
>
> ------------------------------
>
> Message: 4
> Date: Wed, 4 Jun 2008 14:37:59 -0500
> From: Tilghman Lesher
> <tilghman at mail.jeffandtilghman.com>
> Subject: Re: [asterisk-dev] [policy] Discussion on IRC -
> how to make
> -dev more useful
> To: Asterisk Developers Mailing List
> <asterisk-dev at lists.digium.com>
> Message-ID:
> <200806041437.59924.tilghman at mail.jeffandtilghman.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> On Wednesday 04 June 2008 10:45:27 Jay R. Ashworth wrote:
> > On Wed, Jun 04, 2008 at 10:26:48AM -0500, Tilghman
> Lesher wrote:
> > > People who don't want to call extra attention
> to their proposals are free
> > > to ignore it. Those of us who _want_ to draw
> attention and discussion
> > > from the list will use the tags. Quite simple,
> really. I think I've
> > > already used the [policy] tag once before, and
> it's nice to have a formal
> > > proposal, so that we all know what the various
> tags mean.
> >
> > And people who want to join in your
> >
> > 1452 N * 06/04 Tilghman Lesher ( 51) Re:
> [asterisk-dev] [policy]
> > Discussion on
> >
> > oh, wait. Discussion on *what*? Bummer. :-)
>
> Without the tag, the remainder of the subject would have
> been
> "Discussion on IRC - how to", which still
> doesn't give you any more
> information. The real key is probably that you need to
> widen your
> terminal, which I do believe Mutt will take advantage of to
> display
> more characters in the subject.
>
> --
> Tilghman
>
>
>
> ------------------------------
>
> Message: 5
> Date: Wed, 4 Jun 2008 20:51:14 +0100
> From: Tim Panton <thp at westhawk.co.uk>
> Subject: Re: [asterisk-dev] Media TimeOut for SIP and IAX
> Trunk
> To: Asterisk Developers Mailing List
> <asterisk-dev at lists.digium.com>
> Message-ID:
> <6DDBCB55-DDA2-454C-9167-6EE53155E34A at westhawk.co.uk>
> Content-Type: text/plain; charset=US-ASCII; format=flowed
>
>
> On 4 Jun 2008, at 17:12, bilal ghayyad wrote:
>
> > Hi List;
> >
> > Any one can advise me if IAX trunk support media time
> > out to disconnec the call automatically after certain
> > vaule of timeout in case no more media running between
> > end points and hangup signal was not available (which
> > happens in alot of cases)?
> >
> > And in case of SIP, I heared it is existed but I have
> > to check the session timer in the trunk version of the
> > sip_chan, but really I do not know how to check this
> > and wether there is any need to be done on the source
> > to be modified and compiled to obtain it successfully
> > working.
> >
> > Any help?
> >
>
> Isn't this is the default behavior in IAX2? Normally
> IAX
> doesn't separate the media from the control, so
> generally
> IAX will drop a call a few seconds after the last received
> packet.
>
>
>
>
>
>
>
> ------------------------------
>
> Message: 6
> Date: Wed, 4 Jun 2008 16:00:05 -0400
> From: "Jay R. Ashworth" <jra at baylink.com>
> Subject: Re: [asterisk-dev] [policy] Discussion on IRC -
> how to make
> -dev more useful
> To: asterisk-dev at lists.digium.com
> Message-ID: <20080604200005.GB674 at cgi.jachomes.com>
> Content-Type: text/plain; charset=us-ascii
>
> On Wed, Jun 04, 2008 at 02:37:59PM -0500, Tilghman Lesher
> wrote:
> > On Wednesday 04 June 2008 10:45:27 Jay R. Ashworth
> wrote:
> > > On Wed, Jun 04, 2008 at 10:26:48AM -0500,
> Tilghman Lesher wrote:
> > > > People who don't want to call extra
> attention to their proposals are free
> > > > to ignore it. Those of us who _want_ to
> draw attention and discussion
> > > > from the list will use the tags. Quite
> simple, really. I think I've
> > > > already used the [policy] tag once before,
> and it's nice to have a formal
> > > > proposal, so that we all know what the
> various tags mean.
> > >
> > > And people who want to join in your
> > >
> > > 1452 N * 06/04 Tilghman Lesher ( 51) Re:
> [asterisk-dev] [policy]
> > > Discussion on
> > >
> > > oh, wait. Discussion on *what*? Bummer. :-)
> >
> > Without the tag, the remainder of the subject would
> have been
> > "Discussion on IRC - how to", which still
> doesn't give you any more
> > information.
>
> Yes, but *that* isn't the fault of the tagging policy.
> It's the fault
> of the original poster.
>
> > The real key is probably that you need
> to widen your
> > terminal, which I do believe Mutt will take advantage
> of to display
> > more characters in the subject.
>
> I'm old. :-) I have to keep the xterm windows
> comfortably readable
> because I stare at them all day. And, ironically, I have
> another
> problem, which perhaps someone knows the answer to: If you
> *do* run
> mutt in an xterm wider than 80 characters, I have not yet
> found a
> reliable way to get *vi* (vim, actually) to autowrap based
> on the
> *left* margin rather than the right one, as is its default.
> This
> leaves you needing to reset your wm every time you resize
> the window,
> which is similarly unpleasant.
>
> Suggestions? Did I miss, say, the ability to
>
> set wm=-72
>
> ?
>
> Cheers,
> -- jra
> --
> Jay R. Ashworth Baylink
> jra at baylink.com
> Designer The Things I Think
> RFC 2100
> Ashworth & Associates http://baylink.pitas.com
> '87 e24
> St Petersburg FL USA http://photo.imageinc.us
> +1 727 647 1274
>
> Those who cast the vote decide nothing.
> Those who count the vote decide everything.
> -- (Joseph Stalin)
>
>
>
> ------------------------------
>
> Message: 7
> Date: Wed, 04 Jun 2008 15:09:13 -0500
> From: Mark Michelson <mmichelson at digium.com>
> Subject: Re: [asterisk-dev] [other] Reliability of TRANSFER
> event in
> queue_log
> To: Asterisk Developers Mailing List
> <asterisk-dev at lists.digium.com>
> Message-ID: <4846F669.8000800 at digium.com>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> Sa?l Ibarra wrote:
> > Hi all:
> >
> > I'm about to precess some queue events, and by
> reading the
> > documentation I found the event TRANSFER isn't
> reliable with native
> > sip transfers. Is this still true? Is there any
> workaround without
> > using asterisk transfers? Thanks in advance.
> >
>
> When you say "queue events" I assume you mean
> lines in the queue_log file. The
> TRANSFER event will only show up in the queue_log on a
> blind transfer. This can
> be done either with a native SIP transfer or with
> Asterisk's internal blind
> transfer mechanism. Attended transfers, no matter how they
> are initiated, will
> not show up in the queue_log.
>
> Mark Michelson
>
>
>
> ------------------------------
>
> Message: 8
> Date: Wed, 4 Jun 2008 16:10:22 -0400
> From: "Jay R. Ashworth" <jra at baylink.com>
> Subject: Re: [asterisk-dev] [policy] Discussion on IRC -
> how to make
> -dev more useful
> To: asterisk-dev at lists.digium.com
> Message-ID: <20080604201022.GC674 at cgi.jachomes.com>
> Content-Type: text/plain; charset=us-ascii
>
> On Wed, Jun 04, 2008 at 04:00:05PM -0400, Jay R. Ashworth
> wrote:
> > I'm old. :-) I have to keep the xterm windows
> comfortably readable
> > because I stare at them all day. And, ironically, I
> have another
> > problem, which perhaps someone knows the answer to:
> If you *do* run
> > mutt in an xterm wider than 80 characters, I have not
> yet found a
> > reliable way to get *vi* (vim, actually) to autowrap
> based on the
> > *left* margin rather than the right one, as is its
> default. This
> > leaves you needing to reset your wm every time you
> resize the window,
> > which is similarly unpleasant.
> >
> > Suggestions? Did I miss, say, the ability to
> >
> > set wm=-72
> >
> > ?
>
> I did: it's called "textwidth", and it
> figures from the left. Thanks,
> Tilghman, for prompting me to go look it up.
>
> Cheers,
> -- jra
> --
> Jay R. Ashworth Baylink
> jra at baylink.com
> Designer The Things I Think
> RFC 2100
> Ashworth & Associates http://baylink.pitas.com
> '87 e24
> St Petersburg FL USA http://photo.imageinc.us
> +1 727 647 1274
>
> Those who cast the vote decide nothing.
> Those who count the vote decide everything.
> -- (Joseph Stalin)
>
>
>
> ------------------------------
>
> Message: 9
> Date: Thu, 05 Jun 2008 09:10:56 +1200
> From: Nic Bellamy <nicb-lists at vadacom.co.nz>
> Subject: Re: [asterisk-dev] Another IAX2 problem with the
> latest
> security fix ...
> To: Asterisk Developers Mailing List
> <asterisk-dev at lists.digium.com>
> Message-ID: <484704E0.2010100 at vadacom.co.nz>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
> Russell Bryant wrote:
> > Tim Panton wrote:
> >
> >> It makes no sense to have a LAGRQ packet without a
> call set up .
> >> Arguably it makes no sense to have a PING without
> a call.
> >>
> >> For what it is worth, I think it would be better
> to
> >> implement the initial 'hack' i.e.
> don't send LAGRQ or PING
> >> untill the call is set up.
> >> Then add an additional hack where these two
> don't have their
> >> call numbers checked for backwards compatibility.
> >>
> >
> > Agreed. So, we'll go with my original hack, plus
> your proposed hack #2 which
> > will maintain backwards compatibility, without
> introducing any unsafe behavior.
> >
>
> Hi Russell,
> just a bit of feedback on this fix, which ended up in
> 1.2.29 -
> firstly, I've been running 1.2.29 for about 24 hours
> now, and haven't
> had any VNAK/INVAL floods, so I think we can consider that
> solved.
>
> The only oddity I've noticed is that a large proportion
> of the peers
> with qualify=yes go LAGGED for a short period after an
> "iax2 reload",
> with lag figures of 2000ms + nominal latency. Not exactly
> critical (at
> least not to me), but perhaps related to the PING/LAGRQ
> changes.
>
> Cheers,
> Nic.
>
> --
> Nic Bellamy,
> Head Of Engineering, Vadacom Ltd -
> http://www.vadacom.co.nz/
>
>
>
>
> ------------------------------
>
> Message: 10
> Date: Wed, 4 Jun 2008 21:44:35 +0000 (UTC)
> From: tony at softins.clara.co.uk (Tony Mountifield)
> Subject: Re: [asterisk-dev] mmichelson: branch 1.2 r620 -
> in
> /branches/1.2/asterisk-ooh323c: ooh323c/src/...
> To: asterisk-dev at lists.digium.com
> Message-ID: <g272c3$ff6$1 at softins.clara.co.uk>
>
> In article <E1K3wHH-0005CF-Lm at lists.digium.com>,
> SVN commits to the Digium repositories
> <svn-commits at lists.digium.com> wrote:
> > Author: mmichelson
> > Date: Wed Jun 4 11:55:58 2008
> > New Revision: 620
> >
> > URL:
> http://svn.digium.com/view/asterisk-addons?view=rev&rev=620
> > Log:
> > A few changes:
> >
> > [...]
> > 2. (1.2 only) There was a char * being used before
> being set. This was causing an almost
> > immediate crash when ooh323 would load. This commit
> fixes that issue.
>
> Please see below...
>
> > [...]
> > Modified:
> branches/1.2/asterisk-ooh323c/src/chan_h323.c
> > URL:
> >
> http://svn.digium.com/view/asterisk-addons/branches/1.2/asterisk-ooh323c/src/chan_h323.c?view=diff&rev=620&r1=619&r2=620
> >
> ==============================================================================
> > --- branches/1.2/asterisk-ooh323c/src/chan_h323.c
> (original)
> > +++ branches/1.2/asterisk-ooh323c/src/chan_h323.c Wed
> Jun 4 11:55:58 2008
> > @@ -1887,8 +1887,12 @@
> > ooconfig.mTCPPortStart = 12030;
> > ooconfig.mTCPPortEnd = 12230;
> >
> > + ast_log(LOG_NOTICE, "Check 1\n");
> > +
> > v = ast_variable_browse(cfg, "general");
> > while(v) {
> > +
> > + ast_log(LOG_NOTICE, "v is %s\n",
> v->name);
> >
> > if (!strcasecmp(v->name, "port"))
> {
> > gPort = (int)strtol(v->value, NULL, 10);
> > @@ -2046,6 +2050,7 @@
> > ast_parse_allow_disallow(&gPrefs,
> &gCapability, tcodecs, 1);
> > }
> > else if (!strcasecmp(v->name,
> "dtmfmode")) {
> > + ast_log(LOG_NOTICE, "v's value is
> %s\n", v->value);
> > if (!strcasecmp(v->value,
> "inband"))
> > gDTMFMode=H323_DTMF_INBAND;
> > else if (!strcasecmp(v->value,
> "rfc2833"))
>
> Are the above changes left-over debugging statements?
>
> > @@ -2070,8 +2075,9 @@
> > {
> > if(strcasecmp(cat, "general"))
> > {
> > - int friend_type = strcasecmp(utype,
> "friend");
> > + int friend_type;
> > utype = ast_variable_retrieve(cfg, cat,
> "type");
> > + friend_type = strcasecmp(utype,
> "friend");
> > if(utype)
> > {
> > if(!strcmp(utype, "user") || 0
> == friend_type)
>
> It's good to see this bug fixed in 1.2, as I also found
> it myself recently,
> but the strcasecmp with utype should really go inside the
> if(utype){ },
> else a missing type= line would still crash Asterisk.
>
> 1.4 and later are already correct in this respect.
>
> Cheers
> Tony
> --
> Tony Mountifield
> Work: tony at softins.co.uk - http://www.softins.co.uk
> Play: tony at mountifield.org - http://tony.mountifield.org
>
>
>
> ------------------------------
>
> _______________________________________________
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> End of asterisk-dev Digest, Vol 47, Issue 17
> ********************************************
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