[asterisk-dev] asterisk-dev Digest, Vol 92, Issue 131
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Fri Mar 23 07:38:14 CDT 2012
Plz stop emails
Sent from my BlackBerry® smartphone from Warid.
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Subject: asterisk-dev Digest, Vol 92, Issue 131
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Today's Topics:
1. Re: [Code Review]: DIALOG_INFO_XML timeout notification stops
BLF's from working. (Alec Davis)
2. adding codec+format (Be?at Urteaga)
----------------------------------------------------------------------
Message: 1
Date: Fri, 23 Mar 2012 11:23:44 -0000
From: "Alec Davis" <reviewboard at asterisk.org>
Subject: Re: [asterisk-dev] [Code Review]: DIALOG_INFO_XML timeout
notification stops BLF's from working.
To: "Kevin Fleming" <kpfleming at digium.com>, "Alec Davis"
<sivad.a at paradise.net.nz>, "Asterisk Developers"
<asterisk-dev at lists.digium.com>, "Mark Michelson"
<mmichelson at digium.com>, "Alec Davis" <reviewboard at asterisk.org>
Message-ID: <20120323112344.7756.42610 at hotblack.digium.com>
Content-Type: text/plain; charset="utf-8"
> On March 22, 2012, 6:14 p.m., Mark Michelson wrote:
> > No.
> >
> > From RFC 4235 Section 4.1: "Versions start at 0, and increment by one for each new document sent to a subscriber. Versions are scoped within a subscription."
> >
> > When the phone reSUBSCRIBEs with a new call-id, that is a new subscription, and so the version starts back at 0 again. The fact that the phone you are using is trying to compare the version of a new subscription against the version of a previous subscription is just completely wrong.
> >
> > Now, in Asterisk, we try to work around some vendor's improper behavior, and it can probably be done here as well, just not in the way you have done it. What you have done is to make dialog-info subscriptions never expire. This means that you are purposely adding a sip_pvt reference leak into the code in the case that a subscription times out. If I understand things correctly, if you perform multiple reboots on your phones, then the number of active subscriptions in Asterisk will stack more and more. I'm willing to bet that if you run 'sip show subscriptions' in your setup after a few reboots of those phones, you'd see a LOT more subscriptions than there should be. In addition, if you look at the SIP traffic after a few reboots, there are probably way more NOTIFYs than necessary going to the phones on every state change.
> >
> > I honestly have no idea how the code change you've presented causes things to work. When your phone reboots, I can see that a new subscription is created based on the fact that the XML has a new call-id. With your change, the old subscription that normally would have timed out does not now. What this means is that Asterisk is maintaining two subscriptions for the phone. Asterisk, on every dialog state change, will be sending two NOTIFYs to the phone. One will be for the old subscription, with a version the phone likes. The other NOTIFY will be for the new subscription, with a version starting at 0 and incrementing on each new NOTIFY. Now, here's where things get weird. The phone is presumably honoring the NOTIFYs for the old subscription, but not the NOTIFYs for the new subscription. Why? Does the phone still have knowledge of the old subscription despite the reboot? Does the phone just accept any unsolicited dialog-info NOTIFY? If the phone knows about the old subscription, and it also knows that it has created a new subscription, then why doesn't the phone unsubscribe from the old one? Why is it trying to apply a per-dialog version to multiple dialogs? Seriously, WTF?
> >
> > I have a possible solution for this that is safer, but I don't know the specific mechanics behind the brokenness of these phones. Currently, the version reported in dialog-info is obtained from the sip_pvt, meaning each subscription maintains its own independent version counter. We could create an option so that for phones like yours, we store the version in the sip_peer instead. This way, if the peer resubscribes, then the version reported in the new subscription will continue from the same point it was in the old subscription. This would allow for the old subscription to still time out and have its resources released properly. If implementing this also causes BLF to be reported properly to your phones, then I'd be willing to accept such a change into Asterisk. Just expect some strongly worded vitriol in the sample config file about how such an option should never have to exist in the first place :)
Working with the senario that a phone has restarted due to loss of power for some reason.
Trusting a phone to unsubscribe to a previous subscription is just wrong, it won't know what it's previous call-id was.
The cleaner removed the power. Then plugged it back in... so many senarios :)
So the phone has just finished booting.
-- Registered SIP 'GXP0001' at 192.168.5.141:5066
[Mar 23 23:54:15] NOTICE[23637]: chan_sip.c:13138 state_notify_build_xml: ALEC <dialog id="8601GXP0001" call-id="e607e6a8e79400dd at 192.168.5.141"> version=0 terminated
astrid-test*CLI> sip show subscriptions
Peer User Call ID Extension Last state Type Mailbox Expiry
192.168.5.141 GXP0001 e607e6a8e79400d 8601GXP0001 at tru Idle dialog-info+xml <none> 000300
192.168.5.141 GXP0001 5a1f3df6ce2e4fc 8601GXP0001 at tru Idle dialog-info+xml <none> 000300
192.168.5.141 GXP0001 052f0371d160f58 -- <none> mwi 8612 000300
3 active SIP subscriptions
[Mar 23 23:56:45] NOTICE[23637]: chan_sip.c:13138 state_notify_build_xml: ALEC <dialog id="8601GXP0001" call-id="e607e6a8e79400dd at 192.168.5.141"> version=1 terminated
-- Registered SIP 'GXP0001' at 192.168.5.141:5064
[Mar 23 23:58:25] NOTICE[23637]: chan_sip.c:13133 state_notify_build_xml: ALEC <dialog id="8601GXP0001" call-id="5a1f3df6ce2e4fcf at 192.168.5.141" direction="recipient"> version=2 terminated
[Mar 23 23:58:25] NOTICE[23637]: chan_sip.c:3868 __sip_autodestruct: ALEC TIMEOUT of SIP subscription 5a1f3df6ce2e4fcf at 192.168.5.141
astrid-test*CLI> sip show subscriptions
Peer User Call ID Extension Last state Type Mailbox Expiry
192.168.5.141 GXP0001 e607e6a8e79400d 8601GXP0001 at tru Idle dialog-info+xml <none> 000300
192.168.5.141 GXP0001 052f0371d160f58 -- <none> mwi 8612 000300
2 active SIP subscriptions
>From this, I take it as call-id of '5a1f3df6ce2e4fc' has been killed off, and with my change the phone wasn't sent the timeout to the device.
I'm wondering why we didn't finish up with 2 MWI subscriptions, there may be a clue there, did the code kill off the previous mwi subscription when another for the same device arrived, or did the phone unsubscribe.
- Alec
-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/1813/#review5867
-----------------------------------------------------------
On March 13, 2012, 5:35 p.m., Alec Davis wrote:
>
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/1813/
> -----------------------------------------------------------
>
> (Updated March 13, 2012, 5:35 p.m.)
>
>
> Review request for Asterisk Developers.
>
>
> Summary
> -------
>
> Some phones (Grandstream GXP20XX series 1.2.5.3 firmware) if rebooted, loose the BLF lights due to the new subscription's "version" being less than the previous subscription's "version".
> This happens the instant that asterisk timeouts the previous subscription - which has the much larger "version" sequence number.
>
> The only way for the lights to start working again, is for the current subscription's "version" number to increment past the previous old "version" number.
> That period of time could be huge, if the phone isn't rebooted for days.
>
> The workaround has been to reboot the phone twice, within a few minutes of each reboot.
> 1st time wait for the BLF's to fail (then reboot again) - which is the timeout period - a few minutes.
> 2nd time the lights will work, then fail, then after a few minutes start working again.
>
> This fix prevents the old subscription timeout from updating the phones "version" number.
>
> This issue doesn't affter the GXP21XX series.
>
> Attempt to show issue byway of example below with some debug output showing call-id and version number.
>
> astrid-test*CLI> core show hints
>
> -= Registered Asterisk Dial Plan Hints =-
> 8601GXP0001 at trusted : Custom:q8601_a8612 State:InUse Watchers 1
> Debug output:
>
> ===Set the BLF state to InUse
> - Executing [s at macro-custom-agent-inout:13] Set("SIP/GXP0001-00000005", "DEVICE_STATE(Custom:q8601_a8612)=INUSE") in new stack
> ...
> [Mar 14 11:01:39] NOTICE[10819]: chan_sip.c:13136 state_notify_build_xml: ALEC <dialog id="8601GXP0001" call-id="3c65aed02ae1d5df at 192.168.5.141"> version=5
>
> =============
> Reboot the phone, it gets a new BLF subscription.
> =============
> [Mar 14 11:02:30] NOTICE[10826]: chan_sip.c:13136 state_notify_build_xml: ALEC <dialog id="8601GXP0001" call-id="79956c0168125400 at 192.168.5.141"> version=0
> ===The BLF has correct state of InUse.
>
> ...
> [Mar 14 11:05:00] NOTICE[10826]: chan_sip.c:13136 state_notify_build_xml: ALEC <dialog id="8601GXP0001" call-id="79956c0168125400 at 192.168.5.141"> version=1
> ===The BLF still has correct state of InUse.
>
> ...
> The timeout 'last notification'
> [Mar 14 11:05:09] NOTICE[10826]: chan_sip.c:13131 state_notify_build_xml: ALEC <dialog id="8601GXP0001" call-id="3c65aed02ae1d5df at 192.168.5.141" direction="recipient"> version=6
> [Mar 14 11:05:09] NOTICE[10826]: chan_sip.c:3868 __sip_autodestruct: ALEC TIMEOUT of SIP subscription 3c65aed02ae1d5df at 192.168.5.141
> ===The BLF has incorrectly gone Idle.
>
> ...
> [Mar 14 11:15:01] NOTICE[10826]: chan_sip.c:13136 state_notify_build_xml: ALEC <dialog id="8601GXP0001" call-id="79956c0168125400 at 192.168.5.141"> version=5
> ===The BLF is still incorrect at Idle.
>
> -- Registered SIP 'GXP0001' at 192.168.5.141:5066
> [Mar 14 11:17:31] NOTICE[10826]: chan_sip.c:13136 state_notify_build_xml: ALEC <dialog id="8601GXP0001" call-id="79956c0168125400 at 192.168.5.141"> version=6
> ===The BLF now has correct state of InUse.
>
>
> Diffs
> -----
>
> trunk/channels/chan_sip.c 358608
>
> Diff: https://reviewboard.asterisk.org/r/1813/diff
>
>
> Testing
> -------
>
> In place on a 1.8 deployment for months, our production box with ~80 GXP phones, half of which are the older GXP20XX series.
>
>
> Thanks,
>
> Alec
>
>
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Message: 2
Date: Fri, 23 Mar 2012 13:23:30 +0100
From: Be?at Urteaga <burteaga at traintic.com>
Subject: [asterisk-dev] adding codec+format
To: "asterisk-dev at lists.digium.com" <asterisk-dev at lists.digium.com>
Message-ID:
<BFF5B4E4912E58409575205E00B8CE0E4D0FB88527 at exchsvr.traintic.com>
Content-Type: text/plain; charset="iso-8859-1"
Hi,
I'm using asterisk 1.8.8.4 and I'd like to add an audio codec and format in order to be able to send audio to my end SIP devices at a 8 bit 24 kHz. The devices of course support it. How could I get this? I guess I should add codec_whatever.c and format_whatever.c files to the source code directories, but: must I add anything else? For example, may be a SDP negotiation may need to be made, so I should add something else apart from the codec and format files?
I think format_pcm.c and codec_ulaw.c could be a good starting point (example), but do you recommend something different?
Any help would be much appreciated!!
Thank you!
Ben.
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