Re: [asterisk-users] Queues and penalties

[Steve Davies]

I'm fairly sure the patch to App Queue that was added to Asterisk 13+
should do the job... It causes agent priorities to "float up" over time so
that new agents are included without excluding old agents.

I can't find it right now but there can't be that many app_queue patches to
ast 13 in the last 18 months

Steve


On Fri, 30 Nov 2018 at 09:18, Paddy Grice  wrote:

> Thanks Leon
>
> I will implement and test but I knew there would be a fix for what I
> believe is a short coming in app_queue. How do I suggest this as a option
> to the base code?
>
> Paddy
>
> --
> *From:* Leon Wright [mailto:lwri...@corpcloud.com.au]
> *Sent:* 30 November 2018 02:17
> *To:* pa...@wizaner.com; asterisk-users@lists.digium.com
> *Cc:* johnkinis...@gmail.com
> *Subject:* Re: [asterisk-users] Queues and penalties
>
> Paddy,
>
> This appears to be how the queue app works. I ended up patching the queue
> app:
>
> diff --git a/apps/app_queue.c b/apps/app_queue.c
> index e3a4e22..72072d0 100644
> --- a/apps/app_queue.c
> +++ b/apps/app_queue.c
> @@ -4571,7 +4571,7 @@ static int ring_one(struct queue_ent *qe, struct
> callattempt *outgoing, int *bus
> struct callattempt *cur;
> /* Ring everyone who shares this best metric (for
> ringall) */
> for (cur = outgoing; cur; cur = cur->q_next) {
> -   if (cur->stillgoing && !cur->chan &&
> cur->metric <= best->metric) {
> +   if (cur->stillgoing && !cur->chan &&
> cur->metric >= qe->min_penalty * 100 && cur->metric <= qe->max_penalty
> * 100) {
> ast_debug(1, "(Parallel) Trying
> '%s' with metric %d\n", cur->interface, cur->metric);
> ret |= ring_entry(qe, cur, busies);
> }
>
> So the penalties get calculated during the 'ringall' strategy and allowing
> the queue app to exit, looping and raising the max penalty and calling the
> queue app again.
>
> Leon
>
> On Thu, 29 Nov 2018 at 18:24, Paddy Grice  wrote:
>
>> Hi John
>>
>> This works fine providing extensions 1001,1002 and 1003 are "Incall" or
>> "Paused" - the problem appears to be that is a handset say 1002 is
>> "ringing" then the 2xxx then the penalty is not honoured.
>>
>> This is well described in the History section of the following link
>> https://wiki.freepbx.org/display/PPS/lazymembers+patch+to+app_queue
>>
>> As I say this seems to be a real shortcoming in app_queue.
>>
>> Any ideas, suggestions, anyone want to work with me to sort this ?
>>
>> Paddy
>>
>>
>> --
>> *From:* John Kiniston [mailto:johnkinis...@gmail.com]
>> *Sent:* 28 November 2018 21:17
>> *To:* pa...@wizaner.com; Asterisk Users Mailing List - Non-Commercial
>> Discussion
>> *Subject:* Re: [asterisk-users] Queues and penalties
>>
>> This should work, How are you defining your timeouts in the queues.conf ?
>>
>> And to verify, in your extensions.conf you are calling Queue with the
>> queue name and the ruleset to apply from queuerules.conf?
>>
>> On Wed, Nov 28, 2018 at 12:45 PM Paddy Grice  wrote:
>>
>>> Hi All
>>>
>>> I have been looking at this problem for a few days/weeks now and after
>>> some advice please.
>>>
>>> I currently have a customer on 11.25.3 and I am in the process of
>>> upgrading versions and OS (Debian) and all things that involves mysql ->
>>> PDO etc
>>>
>>> The problem I have is the customer want a simple call distribution like
>>> this
>>>
>>> Extn 1001, 1002, 1003 to be called on an incoming call - if they don't
>>> answer after 20 seconds then 2001, 2002, 2003 to be added to the ringing
>>> extensions and if no one answers after another 20 seconds the add in 3001,
>>> 3002, 3003.
>>>
>>> Seems a simple queue application to me
>>>
>>> 1001, 1002 and 1003 in the queue with a penalty of 1 strategy ringall
>>> 2001, 2002 and 2003 in the queue with a penalty of 2 strategy ringall
>>> 3001, 3002 and 3003 in the queue with a penalty of 3 strategy ringall
>>>
>>> and rules
>>>
>>> increasing the maxpenalty 1->2 after 20 seconds
>>> and increasing maxpenalty 2->3 after another 20 seconds.
>>>
>>> But this doesn't work if users don't answer!!
>>>
>>> if user 1002 or (2001 etc)  just lets his phone ring - he forgot to
>>> logoff or DND then the penalty is ignored.
>>>
>>> There seems to have been a patch for FreePBX on V13 - LazyMembers - but
>>> that is all I can find and later versions have no mention of this
>>>
>>> I guess I can use autopause and some AMI / Script but this stops phones
>>> ringing because of the timeout so the user has a ringing phone and then it
>>> stops and then it starts again whereas the penalty just adds handsets into
>>> the ringing group.
>>>
>>> This seems to be a real shortcoming in app_queue.
>>>
>>> Any ideas, suggestions, anyone want to work with me to sort this ?
>>>
>>> Paddy Grice
>>>
>>>
>>>
>>>
>>>
>>>
>>>

Re: [asterisk-users] RTP Timestamp rewind

[Steve Davies]

Mark,

You have cropped the image you inserted above and removed a very important
part of the line you highlighted. I think is says ",Mark" after the time
value - You can even see the un-cropped comma in your picture.

RTP timestamps can be reset mid-stream if needed - It is part of the spec,
and most commonly happens when initially (eg Asterisk) generated audio is
replaced with audio from an external source once the call is bridged. The
early timestamp comes from Asterisk, and the subsequent timestamp is
retained from the new source of the RTP.

No packets should be dropped though in my experience some jitter buffers
can handle it poorly.

Hope that helps,
Steve

On Tue, 29 Aug 2017 at 19:39 Mark Wiater  wrote:

> Hi folks.
>
> I have a couple of questions regarding RTP.
>
> The background of my inquiry is that I have packet captures of SIP and RTP
> traffic on an Asterisk and Broadworks SIP trunk and the RTP many times has
> a time stamp that rewinds by 480 using g.711u. The Sequence number
> continues to increment appropriately, but the timestamp just rewinds.
>
>
>
> It doesn't happen on every call, but it's frequent enough to make me want
> to understand it better.
>
> My questions are:
>
> Is there ever a circumstance where it would be normal or logical to see
> the RTP timestamp go backwards during the RTP stream?  Consistently by 480,
> 3 voice frames?
>
> Will Asterisk just drop the packets that compromise the rewind?
>
> Thanks
>
> Mark
>
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Re: [asterisk-users] Asterisk sip_autodestruct messages - extensions locked

[Steve Davies]

Based on the line number of that error in chan_sip.c, it looks like you're
running Asterisk 1.8 or earlier.

AFAIK, The issue you are seeing was fixed years ago, but not THAT many
years ago!

If I'm right, you should upgrade to fix that issue.

Cheers,
Steve

On Fri, 30 Jun 2017 at 13:39 Stefan Viljoen 
wrote:

> Hi guys
>
>
>
> Does anybody have any opinion on what causes tens of thousands of these
> messages per hour to pop up in the CLI:
>
>
>
> [Jun 30 14:24:59] WARNING[2209]: chan_sip.c:4057 __sip_autodestruct:
> Autodestruct on dialog '7e9597ae6ce95fef23374f4b380a9b70@192.168.0.1:5060'
> with owner SIP/1148-0005bb2d in place (Method: BYE). Rescheduling
> destruction for 1 ms
>
> [Jun 30 14:25:01] WARNING[2209]: chan_sip.c:4057 __sip_autodestruct:
> Autodestruct on dialog '6faefcc24547f1e774864ca87e3ff335@192.168.0.1:5060'
> with owner SIP/1028-0005bb3f in place (Method: BYE). Rescheduling
> destruction for 1 ms
>
> [Jun 30 14:25:02] WARNING[2209]: chan_sip.c:4057 __sip_autodestruct:
> Autodestruct on dialog '0d64480f4052a9e9054153552f1af7ba@192.168.0.1:5060'
> with owner SIP/1412-0005bb5d in place (Method: BYE). Rescheduling
> destruction for 1 ms
>
>
>
> Symptoms are that an extension will dial, converse and hang up, but then
> be unable to dial for up to four minutes after the initial call that
> extension made - all the while thousands of the above messages scroll by in
> the CLI. Along with then hundreds of “too many calls” warnings for each
> extension which are limited to 1 call per extension at a time in sip.conf
>
>
>
> I’ve googled intensively, no AGI is being run, hangup literally calls
> Hangup(). I’ve already disabled and unloaded CDR-TDS, and ODBC CEL and CDR
> logging to MySQL (in case db issues are causing a lock or something when
> writing CDRs)
>
>
>
> I see on big G that many many people have had this issue, but nowhere is
> there any kind of concrete end result or answer to any of the questions.
> This has started to happen at our site this Monday... it comes and goes
> driven by no factor I can determine, except maybe workload. Server load
> average when this is going on is 8 in top for a quad-Core i7 machine with
> Centos 7 and 8GB of RAM, but the machine has hit 16 as load avg without
> this happening. It also sometimes happens with the load average at 4 or
> less...
>
>
>
> The particular server has been running undisturbed for about three years
> and has handled tens of millions of calls, and only this Monday started
> exhibiting this behaviour. The dialplan was last changed about three months
> (and about 3 million calls) ago.
>
>
>
> Any body got ANY advice or ideas where I can begin to diagnose this?
>
>
>
> Thanks,
>
>
>
> Stefan
>
>
>
>
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Re: [asterisk-users] OT: Explain where mailing list bouncing comes from ?

[Steve Davies]

I am also getting this, three or four times in the last month after years
of no problems.

I agree that Gmail is the likely common factor, but I would love to have
access to these bounce messages to know whether it is actually an
overly-paranoid list server!

Steve

On Mon, 12 Jun 2017 at 09:09 Andrew Furey  wrote:

> Ditto; a Gmail issue?
>
> Andrew
>
> On 12 June 2017 at 16:00, Marcelo Terres  wrote:
>
>> It is happening the same with me.
>>
>> Regards,
>> Marcelo H. Terres 
>> IM: mhter...@jabber.mundoopensource.com.br
>> https://www.mundoopensource.com.br
>> https://twitter.com/mhterres
>> https://linkedin.com/in/marceloterres
>>
>>
>> On 12 June 2017 at 08:07, Olivier  wrote:
>> > Hello,
>> >
>> > I'm a faithful reader of this mailing list, for several years now.
>> >
>> > Lately, I'm receiving emails asking me to re-enable my list
>> subscription due
>> > to "excessive bouncing".
>> >
>> > What does this exactly mean and why am I receiving this ?
>> > Beside re-enabling my subscription, what can I do to improve things ?
>> >
>> > Regards
>> >
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>
>
>
> --
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> reason that only children read books with only pictures in them.
> Language, be it English or something else, is the only tool flexible
> enough to accomplish a sufficiently broad range of tasks.
>   -- Bill Garrett
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Re: [asterisk-users] Using queue priorities to add agents

[Steve Davies]

Hi,

Thanks for this suggestion - I believe it does not quite fit the
requirement as follows:

- When you move up to priority 2, you will stop ringing 'Receptionist' as
they are out of scope penalty 1 < 2.

- Changing the penalty from
penaltychange => 20,2,2
to
penaltychange => 20,1,2
in order to include Receptionist does not work either, as it will still not
treat 'Receptionist', 'Kelli', 'Traci' equally as required.

In THEORY (I've not tried this disgusting hack yet), I could use:

[mrule]
penaltychange => 20,2,2
penaltychange => 40,3,3
penaltychange => 80,4,4
penaltychange => 120,5,5
penaltychange => 150,6,6
penaltychange => 180,1,1
penaltychange => 200,2,2
penaltychange => 220,3,3
penaltychange => 240,4,4
penaltychange => 260,5,5
penaltychange => 280,6,6
penaltychange => 300,1,1

[myqueue]
member => SIP/100.2,1,Receptionist
member => SIP/100.2,2,Receptionist
member => SIP/101.2,2,Kelli
member => SIP/102.2,2,Traci
member => SIP/100.2,3,Receptionist
member => SIP/101.2,3,Kelli
member => SIP/102.2,3,Traci
member => SIP/103.2,3,Debi
member => SIP/100.2,4,Receptionist
member => SIP/101.2,4,Kelli
member => SIP/102.2,4,Traci
member => SIP/103.2,4,Debi
member => SIP/104.2,4,Debbie
member => SIP/105.2,4,Luci
...and so on...

:) See my problem?

Cheers,
Steve

On Thu, 11 May 2017 at 16:44 John Kiniston <johnkinis...@gmail.com> wrote:

> I have a real ugly queue that has this in it's rules
>
> [mrule]
> penaltychange => 20,2,2
> penaltychange => 40,3,3
> penaltychange => 80,4,4
> penaltychange => 120,5,5
> penaltychange => 150,6,6
> penaltychange => 180,1,1
> penaltychange => 200,2,2
> penaltychange => 220,3,3
> penaltychange => 240,4,4
> penaltychange => 260,5,5
> penaltychange => 280,6,6
> penaltychange => 300,1,1
>
> [myqueue]
> member => SIP/100.2,1,Receptionist
> member => SIP/101.2,2,Kelli
> member => SIP/102.2,2,Traci
> member => SIP/103.2,3,Debi
> member => SIP/104.2,4,Debbie
> member => SIP/105.2,4,Luci
> member => SIP/106.2,5,Sheila
> member => SIP/107.2,6,Mike
>
> So every 20 seconds it jumps up to the next Penalty and every few minutes
> it resets the penalty back down to 1 and starts again.
>
>
> On Thu, May 11, 2017 at 4:17 AM, Steve Davies <davies...@gmail.com> wrote:
>
>> Hi,
>>
>> I have a scenario that I am failing to implement using the Queue app, but
>> which I had thought would be commonplace...
>>
>> 1) (this bit works fine) I want a queue caller to have access to the
>> basic set of agents initially, with an overflow to additional agents if
>> they are busy - This is done using penalty:
>>
>> queues.conf:
>> member => SIP/dev1,0,Agent1
>> member => SIP/dev2,0,Agent2
>> member => SIP/dev3,1,Agent3 is overflow
>>
>> 2) But, after 60 seconds, I want Agent 3 to be included whether the 1 and
>> 2 are busy or not. None of the queuerules options seem to achieve this
>> because regardless of which agents are included or not, the penalty used to
>> group them is also penalising them.
>>
>> Help? Is what I want possible?
>>
>> PS. I did consider hacking the meaning of QUEUE_MIN_PENALTY so that it
>> actually increases lower penalties to it's current value, thus putting them
>> on an even footing, instead of blocking out agents.
>>
>> Thanks,
>> Steve
>>
>>
>
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Re: [asterisk-users] Using queue priorities to add agents

[Steve Davies]

Thanks for the suggestion - That is what is currently in place, but it
allows queue-jumping as Asterisk does not know that one queue should be
serviced (drained) before the other. That can be improved upon by doing a
Waiting count on the 2nd queue etc etc, but there is always a q-jumping
scenario unless the whole thing is managed inside a single queue.

Cheers,
Steve

On Thu, 11 May 2017 at 16:36 Alexander Lopez <alex.lo...@opsys.com> wrote:

> If after 60 seconds you mean ’60 seconds of caller hold time’ then set up
> another queue as overflow,
>
>
>
> Set the first queue to timeout after 60 secs. Then send to the overflow
> queue with all agents/members as same priority.
>
>
>
>
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Steve Davies
> *Sent:* Thursday, May 11, 2017 7:18 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion <
> asterisk-users@lists.digium.com>
> *Subject:* [asterisk-users] Using queue priorities to add agents
>
>
>
> Hi,
>
>
>
> I have a scenario that I am failing to implement using the Queue app, but
> which I had thought would be commonplace...
>
>
>
> 1) (this bit works fine) I want a queue caller to have access to the basic
> set of agents initially, with an overflow to additional agents if they are
> busy - This is done using penalty:
>
>
>
> queues.conf:
>
> member => SIP/dev1,0,Agent1
>
> member => SIP/dev2,0,Agent2
>
> member => SIP/dev3,1,Agent3 is overflow
>
>
>
> 2) But, after 60 seconds, I want Agent 3 to be included whether the 1 and
> 2 are busy or not. None of the queuerules options seem to achieve this
> because regardless of which agents are included or not, the penalty used to
> group them is also penalising them.
>
>
>
> Help? Is what I want possible?
>
>
>
> PS. I did consider hacking the meaning of QUEUE_MIN_PENALTY so that it
> actually increases lower penalties to it's current value, thus putting them
> on an even footing, instead of blocking out agents.
>
>
>
> Thanks,
>
> Steve
>
>
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[asterisk-users] Using queue priorities to add agents

[Steve Davies]

Hi,

I have a scenario that I am failing to implement using the Queue app, but
which I had thought would be commonplace...

1) (this bit works fine) I want a queue caller to have access to the basic
set of agents initially, with an overflow to additional agents if they are
busy - This is done using penalty:

queues.conf:
member => SIP/dev1,0,Agent1
member => SIP/dev2,0,Agent2
member => SIP/dev3,1,Agent3 is overflow

2) But, after 60 seconds, I want Agent 3 to be included whether the 1 and 2
are busy or not. None of the queuerules options seem to achieve this
because regardless of which agents are included or not, the penalty used to
group them is also penalising them.

Help? Is what I want possible?

PS. I did consider hacking the meaning of QUEUE_MIN_PENALTY so that it
actually increases lower penalties to it's current value, thus putting them
on an even footing, instead of blocking out agents.

Thanks,
Steve
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Re: [asterisk-users] ** in extensions.conf

[Steve Davies]

On Wed, 26 Apr 2017 at 20:29 Jerry Geis  wrote:

> I just tried this in my extensions.conf
>
> exten => **,1,Noop(Testing)
> exten => **,n,Playback(demo-congrats)
>
> Did a reload... and the above does not happen.
> I created as 12 instead of the ** and that works fine.
>
> Is there anyway to get the ** to work?  I also am using a polycom phone if
> that affects things. I'm using asterisk 13.15.0
>
> Thanks
>
> Jerry
>
>
On a Polycom handset, dialling '**' will by default be translated into '+'
before it is dialled. You could:

1) dial *..pause..* which will overcome that AFAIK.
2) Configure call.InternationalDialing.enabled="0" on the handset to stop
it.
3) Put a pattern of _[+],1,... into your dialplan.

That would be by guess anyway :)
Steve
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Re: [asterisk-users] Cisco IP 8841 asterisk integration

[Steve Davies]

If you need to know what the provisioning XML should look like for a 3PCC
build of a Cisco 78xx or 88xx phone, then let it boot without provisioning,
and then log in to its web interface. Select admin mode and log-in if
necessary. Then edit the URL in the browser from:

http://ip-address/admin/
to
http://ip-address/admin/cfg.xml

To see the handset's current configuration.

Note that these phones are VERY fussy about being sent clean, compliant XML
for provisioning.

Hope that helps,
Steve

On Tue, 13 Dec 2016 at 12:37 Gopalakrishnan N 
wrote:

> Thank you for the information. Actually the phone came with sip firmware.
> I tried with TFTP with SEPMAC.CNF.XML and other relevant xml files. But the
> phone stuck with blank screen while bootup.
>
> And Cisco TAC support says, the phones part number is with enterprise
> firmware and it can't work with Asterisk.
>
> Regards.
>
>
>
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Re: [asterisk-users] Cisco IP 8841 asterisk integration

[Steve Davies]

I tried... repeatedly... I failed. I bought some 3PCC phones, and they just
worked.

If you have the relevant Cisco telephony server product you might be able
to trick it into doing what you want, as that has the proper upgrader for
that model of phone.

I previously had experience of upgrading the Cisco build to the SIP build
on Cisco 7641 handsets, which have 2 similar builds, but none of the
techniques seemed to apply this time around.

Cheers,
Steve


On Sun, 4 Dec 2016 at 16:03 Gopalakrishnan N <gopalakrishnan...@gmail.com>
wrote:

> Can't I upload the 3PCC firmware ? available from the Cisco website?
>
> Actually it came with sip88xx firmware.
>
> Regards .
>
>
> On Fri, 2 Dec 2016, 10:38 p.m. Steve Davies, <davies...@gmail.com> wrote:
>
> Hi,
>
> You have to buy the 3PCC version for this to work. Once you have this,
> they work very much like the Cisco SPA handsets.
>
> I also ended up with a non-3PCC handset and it is useless, and as far as I
> can tell they cannot be re-flashed.
>
> Cheers,
> Steve
>
>
>
> On Fri, 2 Dec 2016 at 16:16 Gopalakrishnan N <gopalakrishnan...@gmail.com>
> wrote:
>
> Anyone tried integrating Cisco IP 8841 phone with Asterisk 11.x. I have
> the phone with sip firmware came along with sip88xx-11.0.1SR xx. I tried to
> upload woth TFTP due to some reason it's getting failed. Do I need to load
> 3pcc firmware  or anyway to Configure from the phone itself or from the
> GUI?
>
> I have the SEPMAC.cnf.xml as well.
>
> Any suggestions would be appreciated.
>
> Regards .
>
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>
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>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
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>
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>
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Re: [asterisk-users] Cisco IP 8841 asterisk integration

[Steve Davies]

Hi,

You have to buy the 3PCC version for this to work. Once you have this, they
work very much like the Cisco SPA handsets.

I also ended up with a non-3PCC handset and it is useless, and as far as I
can tell they cannot be re-flashed.

Cheers,
Steve



On Fri, 2 Dec 2016 at 16:16 Gopalakrishnan N 
wrote:

> Anyone tried integrating Cisco IP 8841 phone with Asterisk 11.x. I have
> the phone with sip firmware came along with sip88xx-11.0.1SR xx. I tried to
> upload woth TFTP due to some reason it's getting failed. Do I need to load
> 3pcc firmware  or anyway to Configure from the phone itself or from the
> GUI?
>
> I have the SEPMAC.cnf.xml as well.
>
> Any suggestions would be appreciated.
>
> Regards .
> --
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> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
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Re: [asterisk-users] pedantic=yes in sip.conf

[Steve Davies]

Alan,

A little more context would be useful. Where are you putting the '#' and
why? ( If all else fails, print it out and mail it to them ;-) )

%23 is the correct encoding for a hash '#' symbol in many SIP contexts, and
should be decoded by a properly functioning far-end.

Regards,
Steve


On Wed, 30 Sep 2015 at 19:20 sysad...@reed-media.com <
sysad...@reed-media.com> wrote:

> Hi guys
> i'm using asterisk 11.18.0.
> I need to send the pound # sign to my SIP provider, but each time it's
> reencoded in %23.
> I try to put pedantic=yes in the sip.conf as advised here:
> http://www.voip-info.org/wiki/view/Asterisk+SIP+pedantic
>
> but nothing's changed.
>
> Have someone already met this issue please ?
>
> thanks a lot,
>
>
> regards,
>
> Alan
>
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Re: [asterisk-users] Retransmission Timeout results in dropped calls after 32 seconds

[Steve Davies]

Hi,

In my experience, all Yealink phones work just fine with Asterisk, we have
hundreds (perhaps even low-thousands) out there with customers on Asterisk
1.2, 1.6.2, 1.8 and 11.

If you are accurately representing the SIP trace on the phone and the SIP
trace on Asterisk, then I would strongly suggest a SIP ALG exists in the
network between the two devices and that SIP ALG does not understand SIP
properly. The two halves simply do not match, so something must surely be
interfering.

In my experience it is often an innocent looking Cisco router. Cisco's SIP
implementation is SIP By Cisco rather than RFC compliant SIP. If that
is the case Cisco call it a SIP fixup and you just need to disable it.

Hope that helps,
Steve


On Wed, 13 May 2015 at 16:59 Andrew Martin amar...@xes-inc.com wrote:



 - Original Message -
  From: Joshua Colp jc...@digium.com
  To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
  Sent: Wednesday, May 13, 2015 10:50:02 AM
  Subject: Re: [asterisk-users] Retransmission Timeout results in
 dropped calls   after 32 seconds
 
  Andrew Martin wrote:
   Since some packet loss is a possibility, I assume the protocol has
   mechanisms
   for dealing with it. What should be happening differently in the
   communication
   when packet loss occurs? Should the phone just be re-sending the OK,
   instead of
   printing 0  | ERROR | receive a request with same cseq?? to its
 log? Or
   should
   Asterisk be starting with a new cseq on each INVITE retry?
 
  The 200 OK should be retransmitted until an ACK is received. It honestly
  looks like the phone can't talk to Asterisk and it's just generally
  screwing up signaling.
 

 Thanks for the clarification and help debugging this problem. I will work
 with the phone vendor to see if they can resolve this from their end. If
 you
 have any other ideas about how to disable re-INVITEs on the asterisk side,
 beyond what I have done already, please let me know.

 Thanks,

 Andrew

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Re: [asterisk-users] hangup call gw FXO

[Steve Davies]

Looking at the pastebin, the Vega device sends a CANCEL with reason:

Reason: Q.850 ;cause=16.

Cause 16 is normal clearing and suggests that the original caller has
disconnected. I would take a look at the Vega's logs

Regards,
Steve


On Thu, 5 Mar 2015 at 11:41 ricky gutierrez xserverli...@gmail.com wrote:



 On Wednesday, March 4, 2015, ricky gutierrez xserverli...@gmail.com
 wrote:

 I'm having some problems with a vega sangoma, if a call comes into my
 ivr and hangs up, the call continues to ring and leaves hanging the
 channel, I have to restart Asterisk and everything works Ok

 my sangoma is a vega 50 , 4 FXO .

 I tried different tone of countries and does not work,

 this is the trace of which is for hanging up the channel:

 http://pastebin.com/y410Rhzt

 I was thinking that might help rpt timeout , I have put in 30s, but
 does not work

 any advice?

 regardss



  something strange, I have some extensions not connected to Asterisk and
 if I call, I get the message busy, the version I'm using is asterisk 11.15


 --
 rickygm

 http://gnuforever.homelinux.com
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Re: [asterisk-users] Connected Line presentation in 1.8.x upwards

[Steve Davies]

On 29 July 2013 16:55, Kevin Larsen kevin.lar...@pioneerballoon.com wrote:



 From:Steve Davies davies...@gmail.com
 To:Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com,
 Date:07/29/2013 10:53 AM
 Subject:[asterisk-users] Connected Line presentation in 1.8.x
 upwards
 Sent by:asterisk-users-boun...@lists.digium.com
 --



 Hi,

 I've searched the *asterisk.org* http://asterisk.org/ and voip-info
 wiki sites, but not found an answer that seems to match.

 Hopefully this is a simple question. COLP is working very well on our
 system - Unfortunately it is working a bit TOO well in some circumstances.
 We have some untrusted trunks. On these trunks, an initial CallerID can
 be used, but any redirected caller numbers, COLP updates etc are not safe
 to accept. Sadly I cannot find how to cause COLP updates to be ignored for
 a trunk.

 I need solutions for SIP, IAX and DAHDI, what options do I have? This
 applies to both in- and out-bound calls.

 Are there some variables that I can set just before dialling an outbound
 call, and immediately on receiving an inbound call to determine what the
 callerID values will be for the entire duration of the call? (ie. old-style
 pre-COLP behaviour for specific trunks)

 Thanks for any pointers.

 Regards,
 Steve



 I believe what you are looking for in Dial is the 'I' option.



Ah. Many thanks.

It appears that the normally reliable voip-info wiki is out of date and
does not include that option. I should probably have just used Asterisk's
built-in documentation anyway :)

I guess on an inbound call I will have to conditionally set 'I' on the Dial
command based on the originating channel?

I will also have to go and check what affect this has when a call is SIP
REFER'ed as that might result in an asymmetric requirement. The internal
SIP handset will want updating, but the external SIP trunk call will not.

Regards,
Steve
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[asterisk-users] Connected Line presentation in 1.8.x upwards

[Steve Davies]

Hi,

I've searched the asterisk.org and voip-info wiki sites, but not found an
answer that seems to match.

Hopefully this is a simple question. COLP is working very well on our
system - Unfortunately it is working a bit TOO well in some circumstances.
We have some untrusted trunks. On these trunks, an initial CallerID can
be used, but any redirected caller numbers, COLP updates etc are not safe
to accept. Sadly I cannot find how to cause COLP updates to be ignored for
a trunk.

I need solutions for SIP, IAX and DAHDI, what options do I have? This
applies to both in- and out-bound calls.

Are there some variables that I can set just before dialling an outbound
call, and immediately on receiving an inbound call to determine what the
callerID values will be for the entire duration of the call? (ie. old-style
pre-COLP behaviour for specific trunks)

Thanks for any pointers.

Regards,
Steve
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Re: [asterisk-users] ignore 183 session progress in parallel call scenarios

[Steve Davies]

I am sure I submitted the following alternative behaviour to the
bug-tracker in the past, but cannot find any reference to it. Here is the
patch I use to IMHO improve this behaviour.

In case it is not officially uploaded, I will state here that this code is
disclaimed and unencumbered as if uploaded to JIRA.

Regards,
Steve

On 15 July 2013 17:20, Hristo Trendev dist.li...@gmail.com wrote:

 I think I have found the answer to my questions in the source code of Dial:

 case AST_CONTROL_PROGRESS:
   ast_verb(3, %s is making progress passing it to %s\n, 
 ast_channel_name(c), ast_channel_name(in));
   /* Setup early media if appropriate */
   if (single  !caller_entertained
CAN_EARLY_BRIDGE(peerflags, in, c)) {
   ast_channel_early_bridge(in, c);
   }
   if (!ast_test_flag64(outgoing, OPT_RINGBACK)) {
   if (single || (!single  !pa-sentringing)) {
   ast_indicate(in, AST_CONTROL_PROGRESS);
   }
   }

 .
 .


 Asterisk will attempt to bridge the media only for the case of a single 
 outgoing channel, but at the same time it will happily forward progress 
 messages for parallel calls: (!single  !pa-sentringing) as long as no 180 
 Ringing message was sent out to the caller yet. The questions still remains 
 if this should be reported as bug or if there is indeed a use case when 
 sending 183 progress message, without actually bridging the media stream is 
 desired.



 On Mon, Jul 15, 2013 at 4:14 PM, Hristo Trendev dist.li...@gmail.comwrote:

 Hi,
 I am using asterisk 1.8.22 and have a problem when calling in parallel
 several SIP endpoints and I am not sure how to resolve it. In this case
 Asterisk will not bridge any audio to the caller before the 200 OK. Which
 means any progress announcements, including remotely generated ringback,
 are not passed back to the caller.

 This behavior is completely correct, because there is no way to know
 which early media audio stream to pass back to the caller in a parallel
 call scenario (as in this case several endpoint may indicate session
 progress all at the same time).

 The question is why is asterisk still sending 183 session progress back
 to the caller if no audio is to be bridged before the 200 OK anyway? If 183
 are not passed back to the caller, then at least a 180 Ringing that may
 come from another endpoint will cause the calling endpoint to generate
 local ringback. This won't happen if the caller has received a 183 already.

 So it's a bit of a race condition as well - if the first endpoint to
 reply sends a 183 session progress this means the caller will not hear
 any ringback even if some of the other endpoints are sending back 180
 Ringing.

 The question is can I somehow block 183 messages from being passed back
 to the calling endpoint when dialing several destinations in parallel? I
 don't see a point (please correct me if I'm wrong) to pass only the 183 SIP
 message back to the caller without the corresponding RTP stream, so it may
 be much better to actually ignore it when dealing with parallel call
 scenarios (bug?).

 BR,
 Hristo



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multiple_early_media
Description: Binary data
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Re: [asterisk-users] IPcortex: ERROR[23444] chan_zap.c: You cannot use cause 17 number when in state 6

[Steve Davies]

Hi Xavier,

The issue you are seeing is an old Asterisk/Bristuff bug that was fixed
years ago.

Basically ISDN is unable to understand a call going from RING state to BUSY
state, so Asterisk converts the BUSY into a HANGUP/Normal Clearing, and
warns that this is happening.

Sadly, in that old version there was a resource leak of the call object
when this happened.

I would suggest calling IPCortex directly to see what can be done about
this.

Regards,
Steve



On 11 July 2013 12:04, Mitul Limbani mi...@enterux.in wrote:

 Chan_zap has been deprecated more then 2-3 yrs back. You might have to
 ping ipcortex helpdesk to get fix.

 Mitul
 On Jul 11, 2013 4:32 PM, Xavier Singer - EcuTek xav...@ecutek.com
 wrote:

 We use an IPcortex PABX running Asterisk
 1.2.39-BRIstuffed-0.3.0-PRE-1y-y. We have recently implemented Call Queuing
 for our main incoming line with hold music. The call queue type is: Ring
 all - One call at a time (no position announcement).

 Since implementing this feature we've been receiving the below error
 daily in the mornings and lunchtime when the queue will jump to the next
 available phone, as the main reception phone is in Do Not Disturb mode:

 Jul 11 08:30:54 WARNING[23444] chan_zap.c: 1 Cause code 17 not
 allowed when disconnecting an active call. Changing to cause 16.
 Jul 11 08:30:54 ERROR[23444] chan_zap.c: You cannot use cause 17
 number when in state 6! Corrected.
 Jul 11 08:30:54 WARNING[7133] chan_zap.c: Call specified, but not
 found?
 Jul 11 08:30:54 NOTICE[7133] chan_zap.c: Hangup, did not find
 cref 1, tei 127
 Jul 11 08:30:54 WARNING[7133] chan_zap.c: Hangup on bad channel
 0/1 on span 1
 Jul 11 08:30:58 WARNING[7133] chan_zap.c: Call specified, but not
 found?
 Jul 11 08:30:58 NOTICE[7133] chan_zap.c: Hangup, did not find
 cref 1, tei 127
 Jul 11 08:30:58 WARNING[7133] chan_zap.c: Hangup on bad channel
 0/1 on span 1
 Jul 11 08:47:04 WARNING[7133] chan_zap.c: 1 received SETUP
 message for call that is not a new call (retransmission), peercallstate 19
 ourcallstate 0 cr 1,
 Jul 11 08:47:08 WARNING[7133] chan_zap.c: 1 received SETUP
 message for call that is not a new call (retransmission), peercallstate 19
 ourcallstate 0 cr 1,
 Jul 11 08:47:19 WARNING[7133] chan_zap.c: 1 received SETUP
 message for call that is not a new call (retransmission), peercallstate 19
 ourcallstate 0 cr 1,
 Jul 11 08:47:23 WARNING[7133] chan_zap.c: 1 received SETUP
 message for call that is not a new call (retransmission), peercallstate 19
 ourcallstate 0 cr 1,

 The ERROR happens when the call is ended. I can't replicate the error
 either...

 I suspect that the chan_zap driver has a bug and is possibly trying to
 hang up the call on the first phone in the queue, rather than the phone
 that answered the call.

 I have investigated the different state and causes listed in the above
 log file, and this is what I think they mean (please correct me if I got it
 wrong):
 ISDN State 6 = not initialised
 Cause 16 = normal call clearing
 Cause 17 = user busy
 TEI 127 = reserved as the broadcast TEI


 So my questions are:
 1. What could be causing the error and any suggestions on how to
 troubleshoot this issue?
 2. Can I upgrade the chan_zap driver for the ISDN card without breaking
 the IPcortex frontend (we have root access)?
 3. Should I supply any config files?


 Thanks!
 Xavier



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Re: [asterisk-users] FW: IPcortex: ERROR[23444] chan_zap.c: You cannot use cause 17 number when in state 6

[Steve Davies]

Xavier,

DoNotDisturb generates a Busy indication. Insert that into my earlier
response, and you have an explanation of why the call tries to go from RING
to BUSY, and confirms my theory.

No you cannot replace the Zaptel card driver on its own (and the problem
was bigger than that anyway), as Asterisk is compiled and linked to a
specific Zaptel (Dahdi) version.

As mentioned, you need to call IPCortex.

Regards,
Steve


On 11 July 2013 16:23, Xavier Singer - EcuTek xav...@ecutek.com wrote:

 Update:
 I can reproduce the error by putting the reception phone (call queue 0) in
 Do Not Disturb mode, then call in from outside using a mobile, then pick up
 the call from the 2nd phone in the queue. It will cause the error only if I
 hang up _before_ the mobile hangs up. The error doesn't seem to happen if
 the external call hangs up, or if the call is answered by the reception
 phone (first call in the queue).

 Thanks again,
 Xavier


 -Original Message-
 From: Xavier Singer - EcuTek
 Sent: 11 July 2013 12:02
 To: 'asterisk-users@lists.digium.com'
 Subject: IPcortex: ERROR[23444] chan_zap.c: You cannot use cause 17 number
 when in state 6

 We use an IPcortex PABX running Asterisk 1.2.39-BRIstuffed-0.3.0-PRE-1y-y.
 We have recently implemented Call Queuing for our main incoming line with
 hold music. The call queue type is: Ring all - One call at a time (no
 position announcement).

 Since implementing this feature we've been receiving the below error daily
 in the mornings and lunchtime when the queue will jump to the next
 available phone, as the main reception phone is in Do Not Disturb mode:

 Jul 11 08:30:54 WARNING[23444] chan_zap.c: 1 Cause code 17 not
 allowed when disconnecting an active call. Changing to cause 16.
 Jul 11 08:30:54 ERROR[23444] chan_zap.c: You cannot use cause 17
 number when in state 6! Corrected.
 Jul 11 08:30:54 WARNING[7133] chan_zap.c: Call specified, but not
 found?
 Jul 11 08:30:54 NOTICE[7133] chan_zap.c: Hangup, did not find cref
 1, tei 127
 Jul 11 08:30:54 WARNING[7133] chan_zap.c: Hangup on bad channel
 0/1 on span 1
 Jul 11 08:30:58 WARNING[7133] chan_zap.c: Call specified, but not
 found?
 Jul 11 08:30:58 NOTICE[7133] chan_zap.c: Hangup, did not find cref
 1, tei 127
 Jul 11 08:30:58 WARNING[7133] chan_zap.c: Hangup on bad channel
 0/1 on span 1
 Jul 11 08:47:04 WARNING[7133] chan_zap.c: 1 received SETUP message
 for call that is not a new call (retransmission), peercallstate 19
 ourcallstate 0 cr 1,
 Jul 11 08:47:08 WARNING[7133] chan_zap.c: 1 received SETUP message
 for call that is not a new call (retransmission), peercallstate 19
 ourcallstate 0 cr 1,
 Jul 11 08:47:19 WARNING[7133] chan_zap.c: 1 received SETUP message
 for call that is not a new call (retransmission), peercallstate 19
 ourcallstate 0 cr 1,
 Jul 11 08:47:23 WARNING[7133] chan_zap.c: 1 received SETUP message
 for call that is not a new call (retransmission), peercallstate 19
 ourcallstate 0 cr 1,

 The ERROR happens when the call is ended. I can't replicate the error
 either...

 I suspect that the chan_zap driver has a bug and is possibly trying to
 hang up the call on the first phone in the queue, rather than the phone
 that answered the call.

 I have investigated the different state and causes listed in the above log
 file, and this is what I think they mean (please correct me if I got it
 wrong):
 ISDN State 6 = not initialised
 Cause 16 = normal call clearing
 Cause 17 = user busy
 TEI 127 = reserved as the broadcast TEI


 So my questions are:
 1. What could be causing the error and any suggestions on how to
 troubleshoot this issue?
 2. Can I upgrade the chan_zap driver for the ISDN card without breaking
 the IPcortex frontend (we have root access)?
 3. Should I supply any config files?


 Thanks!
 Xavier



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Re: [asterisk-users] Fundemental changes to CDR within single asterisk family

[Steve Davies]

On 4 April 2013 09:05, Ishfaq Malik i...@pack-net.co.uk wrote:

 On Tue, 2013-03-26 at 07:26 -0500, Matthew Jordan wrote:
  On 03/26/2013 05:22 AM, Ishfaq Malik wrote:
   Hi
  
   In asterisk 1.8.7.0, an inbound call that was transferred to another
   peer would have 2 cdr entries.
  
   In asterisk 1.8.18.0 this same activity has a single cdr entry.
  
   This is a rather large and fundamental change to be enacting halfway
   through a single family branch, was there any reason why this happened?
   It means we can't upgrade without doing significant extra development
   and testing.
  
 
  This was most likely an unintended consequence of some other change
  (most likely dealing with masquerades). Is 1.8.18.0 the exact version
  when the behaviour changed?
 
  Just so I'm clear on the scenario, what are the channel technologies
  involved? Is the transfer initiated via a protocol message or via a DTMF
  feature?
 
  Thanks,
 
  Matt
 

 Hi Matt

 Did you ever spot/recreate the change I was referring to?



me too - I can confirm a behaviour change and will go try and pin down at
what point it happened.

This may take me a while :(

Steve
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Re: [asterisk-users] Congestion() forcing PRI channels to be not available

[Steve Davies]

On 19 December 2012 21:54, Christopher Harrington ch...@acsdi.com wrote:

 You probably already know this, but 1.4x is very old (released in 2006)
 and is officially end-of-life.

 https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

 You might get more help or better behavior by updating to a newer more
 current version of Asterisk, such as 1.8 which will be receiving bug fixes
 into October 2014.


 On Wed, Dec 19, 2012 at 3:47 PM, James Lamanna jlama...@gmail.com wrote:

 Hi,
 I have a PSTN Asterisk box that's connected to other dialplan PBXes
 through IAX2.

 Recently this box was upgraded to 1.4.44 with the latest DAHDI version.
 I've noticed that if one of the dialplan PBXes calls Congestion(), the PRI
 will return ISDN code 34 (as its supposed to do).
 However, the issue is that subsequent calls into that PRI channel are
 immediately responded by a Code 44 (channel not available) even though
 there is no live call on the channel.

 Has anyone else experienced this behavior? Its a pretty crippling
 behavior since all of our channels eventually become unresponsive until a
 'dahdi restart' is issued.

 Thanks.

 -- James


I believe that what you are describing is a very old bug, which is fixed
somewhere in the 1.8 timeline when the interface between DAHDI and Asterisk
is changed slightly. I encountered the same issue some time ago. I do not
recall the exact conditions under which the issue happens, but I believe it
is the attempt to cancel an unanswered inbound call with a specific subset
of cause codes.

If you are using an older Asterisk version, the only workaround is to use
Playtones + Hangup() instead of sending the Congestion() or Busy() cause
codes.

Regards,
Steve
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Re: [asterisk-users] SIP Question - Early audio one-way or 2-way?

[Steve Davies]

On 1 September 2012 09:08, Olle E. Johansson o...@edvina.net wrote:

 31 aug 2012 kl. 13:13 skrev Steve Davies davies...@gmail.com:

 On 31 August 2012 07:49, Olle E. Johansson o...@edvina.net wrote:

 24 aug 2012 kl. 16:18 skrev Steve Davies davies...@gmail.com:

 Hi SIP Gurus,

 I've tried to find the relevant RFCs, but am struggling. I can find
 the odd opinion online, but was wondering if anyone could give a
 definitive answer.

 If a SIP call is initiated (INVITE) and receives either a 180 with
 SDP, or a 183 with SDP, then the remote party will start to send
 audio for the inband-ringing. Asterisk then passes this audio, and it
 is correctly heard by the caller.

 At present, Asterisk will also start to pass back any handset audio in
 return, in theory allowing a conversation to occur on an unanswered
 channel if an endpoint were designed to allow this (free phonecalls
 here we come!).

 My question:

 Should:
 1) Asterisk block outbound audio between the 183 Progress and the 200
 OK packets?
 2) Replace any outbound audio with silence?
 3) Replace outbound audio with a special NULL RTP of some sort? Does that 
 exist?
 4) Allow any audio to be sent regardless?

 I have implemented 1) at present on our test rig, but the lack of
 outbound RTP causes issues with firewall state not being set-up to
 allow the inbound audio. I am not sure how hard/easy it would be to do
 2) as you'd need to create silence of the correct duration to replace
 each audio frame.

 Thoughts please?

 First, because of Asterisk's RTP implementation we have to send some RTP 
 packets at this point. You could mute the calling channel before calling 
 and unmute the channel at answer if needed, but normally sending audio 
 won't hurt. A normal user should not be able to send early media on a 
 pstn-like installation where you bill the calls, so there should be little 
 risc of two-way conversations before an answer.

 In some cases you have to let the caller send DTMF (the famous fed ex 
 example) in
 early media, so we can't block any media by default in Asterisk.

 Using the r option in dial causes a lot of issues, since you can still 
 get busy or congestion when you have early media, so that is not a good 
 solution.

 /Olle


 Excellent information as always Olle. Many thanks.

 My intention is to make the early-audio prevention in SIP a little
 more harsh, such that if SIP receives audio before a 183 or 200 is
 received, it is dropped.

 This fixes the case where useless early-audio is received from a
 non-SIP (eg ISDN) technology, and can cause an onward node to
 auto-enable early audio mode, causing silent ringing and other broken
 behaviours.

 This is one of my pet issues. THe problem today is that many gateway vendors 
 ALWAYS send 183 with sdp,
 regardless if it's a ring tone or a service provider message. If you kill the 
 183, service provider messages
 will disappear. My recommendation (which I've mentioned in tons of mails and 
 blog entries) is to send
 180 ringing with SDP for ring tones and 183 for other messages. That way I 
 could kill the 180 SDP in a Kamailio
 proxy before it hits the Asterisk server. In reality today, by killing 183 
 you will also block important information
 for the caller, like this subscriber has a new number.

I have 2 similar scenarios. One like yours is a SIP provider who
interchangeably uses 180 with sdp, 183 with sdp or 180 no sdp  to
indicate ringing. There seems to be no logic, so we have to honour all
of those cases when they happen. Asterisk handles this already.

The case I am now killing can be caused by a SIP to ISDN call where
the ISDN provides a RINGING response which becomes 180 no sdp, but
the ISDN then proceeds to send early audio regardless. Asterisk will
currently pass that audio even though the SDP exchange is incomplete,
by using the SDP in the initial INVITE to send one way audio. This is
then propagated along the call chain and is most inconvenient.

I think I have a solution, currently in 1.6.2 and 1.8, and if you
think it is legitimate as a change to block that unexpected early
audio, I can put the patch on jira.

Regards,
Steve

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[asterisk-users] SIP Question - Early audio one-way or 2-way?

[Steve Davies]

Hi SIP Gurus,

I've tried to find the relevant RFCs, but am struggling. I can find
the odd opinion online, but was wondering if anyone could give a
definitive answer.

If a SIP call is initiated (INVITE) and receives either a 180 with
SDP, or a 183 with SDP, then the remote party will start to send
audio for the inband-ringing. Asterisk then passes this audio, and it
is correctly heard by the caller.

At present, Asterisk will also start to pass back any handset audio in
return, in theory allowing a conversation to occur on an unanswered
channel if an endpoint were designed to allow this (free phonecalls
here we come!).

My question:

Should:
1) Asterisk block outbound audio between the 183 Progress and the 200
OK packets?
2) Replace any outbound audio with silence?
3) Replace outbound audio with a special NULL RTP of some sort? Does that exist?
4) Allow any audio to be sent regardless?

I have implemented 1) at present on our test rig, but the lack of
outbound RTP causes issues with firewall state not being set-up to
allow the inbound audio. I am not sure how hard/easy it would be to do
2) as you'd need to create silence of the correct duration to replace
each audio frame.

Thoughts please?

Many thanks,
Steve

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Re: [asterisk-users] SIP Question - Early audio one-way or 2-way?

[Steve Davies]

On 24 August 2012 15:34, Faisal Hanif fai...@vopium.com wrote:
 Steve Davies davies...@gmail.com wrote:
Hi SIP Gurus,

I've tried to find the relevant RFCs, but am struggling. I can find
the odd opinion online, but was wondering if anyone could give a
definitive answer.

If a SIP call is initiated (INVITE) and receives either a 180 with
SDP, or a 183 with SDP, then the remote party will start to send
audio for the inband-ringing. Asterisk then passes this audio, and it
is correctly heard by the caller.

At present, Asterisk will also start to pass back any handset audio in
return, in theory allowing a conversation to occur on an unanswered
channel if an endpoint were designed to allow this (free phonecalls
here we come!).

My question:

Should:
1) Asterisk block outbound audio between the 183 Progress and the 200
OK packets?
2) Replace any outbound audio with silence?
3) Replace outbound audio with a special NULL RTP of some sort? Does that 
exist?
4) Allow any audio to be sent regardless?

I have implemented 1) at present on our test rig, but the lack of
outbound RTP causes issues with firewall state not being set-up to
allow the inbound audio. I am not sure how hard/easy it would be to do
2) as you'd need to create silence of the correct duration to replace
each audio frame.

Thoughts please?

Many thanks,
Steve

 hi,

 you can simply avoid this by using local ring r option in dial command. 
 azterisk pass local ring voice to caller and will not bridge b leg audio 
 until b leg is answered.iin
 Regards,

 Faisal Hanif
 (sent from phone)

Nice thought, but what if there is a useful reason for the progress
audio? If it is sent we want to hono[u]r it, and if it is not, we
expect a 180 ringing, and let the SIP device generate the tone,
rather than send an unwanted audio stream to use up bandwidth :)

For example, some UK ISDN services will give a useful call failure
message by sending a progress-frame, followed by some audio. DAHDI and
SIP handle this nicely  with a 183 progress message, and pass on the
message on the un-answered SIP channel.

Regards,
Steve

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Re: [asterisk-users] CONNECTEDLINE() updated during SIP events?

[Steve Davies]

On 25 April 2012 18:05, Kevin P. Fleming kpflem...@digium.com wrote:
 On 04/25/2012 11:54 AM, Steve Davies wrote:

 A further question... It appears that for SIP endpoints, this facility
 only updates RPID and PAI headers? I have found that there appear to
 be 4 different SIP CID-update mechanisms out there as follows:

 - Update RPID and PAI (ITSP and trunks often understand this)
 - Update Contact: header (Aastra handsets use this)
 - A SIP INFO packet if Supported: callerid is specified (Older snom
 firmware uses this)
 - Update From: header if Supported: from-change is specified (RFC
 4916, snom, Yealink)

 Are there existing plans to support any of these other methods? If
 not, I will almost certainly add them for my own use, and submit the
 code.


 No, we have no plans at this time to go beyond RPID and PAI support. Those
 two appear to cover all the current endpoints that we have been able to test
 with, and many community members have also used with other endpoints and had
 success.

Thanks for that, I'll have to test further and see whether all the
devices we use support RPID/PAI. It would certainly be easier than
messing about with headers that should not really be changed!

 Changing the Contact header seems quite wrong; the display-name in a URI in
 the Contact header is pretty much irrelevant. Changing the From header also
 seems wrong; that should indicate who sent the initial INVITE, not who
 redirected it. I don't think we'd want to merge patches that added support
 for either of those mechanisms.

The From: header change is a relatively recent RFC, but I've seen
several handsets supporting it, and several non-Asterisk SIP stacks
using this to achieve COLP updates. I completely agree that changing
the Contact: header is daft, and I have no idea why Aastra use this
method.

Regards,
Steve

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[asterisk-users] CONNECTEDLINE() updated during SIP events?

[Steve Davies]

Hi,

I have read the excellent information here:
https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information
and believe I have an understanding of what is offered. I have a
couple of questions:

- Is it possible to update COLP/COLR when a SIP redirect occurs, or
when a SIP divert is in place? If so, how?

- Is it possible to have the COLP/COLR information updated when a SIP
attended transfer is completed? If so how?

In both of the above cases, there is no obvious dialplan execution
when the calls are redirected, diverted or masqueraded, so we cannot
update the CONNECTEDLINE() information trivially. Or am I missing an
obvious trick?

Thanks,
Steve

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Re: [asterisk-users] CONNECTEDLINE() updated during SIP events?

[Steve Davies]

On 25 April 2012 16:55, Richard Mudgett rmudg...@digium.com wrote:
[snip]

 - Is it possible to have the COLP/COLR information updated when a SIP
 attended transfer is completed? If so how?

 Transfers generate connected line update events automatically.  The connected
 line interception macros give you a chance to alter the connected line
 information as it is passed between the connected endpoints of the bridge.

 In both of the above cases, there is no obvious dialplan execution
 when the calls are redirected, diverted or masqueraded, so we cannot
 update the CONNECTEDLINE() information trivially. Or am I missing an
 obvious trick?

 This is the purpose of the interception macros.

Ah, thank you. I was looking at it back-to-front.

The key bit is Transfers generate connected line update events
automatically. - I can now see this in the source code in
ast_do_masquerade() and elsewhere. This then lets you use the macros
as you describe.

A further question... It appears that for SIP endpoints, this facility
only updates RPID and PAI headers? I have found that there appear to
be 4 different SIP CID-update mechanisms out there as follows:

- Update RPID and PAI (ITSP and trunks often understand this)
- Update Contact: header (Aastra handsets use this)
- A SIP INFO packet if Supported: callerid is specified (Older snom
firmware uses this)
- Update From: header if Supported: from-change is specified (RFC
4916, snom, Yealink)

Are there existing plans to support any of these other methods? If
not, I will almost certainly add them for my own use, and submit the
code.

Regards,
Steve

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Re: [asterisk-users] Asterisk ACL

[Steve Davies]

On 2 April 2012 14:06, Mark Farmer mark.far...@gagenetworks.com wrote:


 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leandro
 Dardini
 Sent: 02 April 2012 13:53
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Asterisk ACL



 Your understanding of the problem seems incorrect. The problem seems due to
 the extension not available in your dialplan. Please check carefully in
 which context the call is placed and if the extension is defined in that
 context.



 Maybe it can be useful to define a _X. extension to catch all not defined
 extensions.



 Leandro

 [Mark Farmer]

 The problem is that the inbound call is not being matched by the correct
 peer and as such falls through to the default context which is not supposed
 to match.

 The problem is around the matching of a range of IP addresses to one peer.

 Thanks

 Mark.

Mark,

This is a problem I have encountered regularly. Your mistake is
thinking that setting deny/permit will cause a peer to be matched if
it falls in the permitted range. It will not. The peer will only match
if the source IP address matches the host= value, and in the case of
dynamic it must match the IP address of the party that registered.

deny/permit will also restrict a 'type=user' or 'type=friend' so that
the username can only be attempted from specified IP ranges.

IAX does what you expect, and I have thought regularly of implementing
in SIP what you expected to be the normal behaviour, but in fact, the
deny/permit will limit where the original registration can come from,
but AFAIK does not get used for subsequent (INVITE) matches until
after the host IP match is completed.

At present, the best solution is to change type to 'friend' and use
username/password based authentication.

Buyer beware - I believe the above to be true, and I hope it makes sense!

Regards,
Steve

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Re: [asterisk-users] Fwd: Re: Asterisk CLI unresponsive

[Steve Davies]

On 6 February 2012 10:45, Jonas Kellens jonas.kell...@telenet.be wrote:

 **
 Hello,

 is there anyone that can give me some more information on these
 deadlocks ?!

 How can these deadlocks occur and what is good practise to avoid these
 problems ??


 Jonas.


The only way to avoid deadlocks is to report them when they happen, provide
the requested debug, and hope for a fix.

Deadlocks are just a type of bug. Nothing you can do to avoid it unless you
can work out the events that lead up to it, and avoid doing that.

I am still using 1.6.2.22 in many places, and if I see a deadlock show
locks trace on the mailing list - I will certainly be looking into it.

Regards,
Steve.
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Re: [asterisk-users] Asterisk CLI unresponsive

[Steve Davies]

On 3 February 2012 12:12, Jonas Kellens jonas.kell...@telenet.be wrote:
 On 02/03/2012 01:05 PM, Mikhail Lischuk wrote:

 Jonas Kellens писал 03.02.2012 12:09:


 using asterisk 1.6.2.22

 What is wrong with Asterisk when the CLI becomes unresponsive ?!


 Greetings. I am using the same version, Asterisk 1.6.2.22

 IDK is your problem is same as mine. My CLI also becomes unresponsive
 sometimes, however it does not affect calls at all. And all commands I type
 get processed and output given after some time - 3 or 5 minutes.

 As I determined, this happens when Asterisk is rotating logs.

 You can see it by issuing `ps aux | grep asterisk` at that time.

 IDK if this is normal (due to logfile locks or whatever) or not.


 Hello,

 my logs get rotated at 8h15, 12h15 and 17h15. The deadlock occurred at
 10h34. Don't think this is the relation between problem and deadlock.

 There is no more output on the CLI, so I don't see if the calls which are
 going on are affected and are hang up.



I think you are best to rebuild your asterisk with lock tracking
enabled (it is one of the debug options) and then when it locks, do a
core show locks and post it here.

Normally I would suggest posting it to the bugtracker, but 1.6.2 is
no-longer supported, so that would be a waste of time.

Regards,
Steve

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[asterisk-users] SIP and NAT best practices since recent changes?

[Steve Davies]

Hi,

Since the recent update to the NAT configuration options and defaults
in chan_sip.so, I am interested in any SIP/NAT best practices advice.

What I've always done in the past is:

Global: nat=no
SIP handsets that are local: nat=no
SIP handsets that are remote: nat=yes
ITSP SIP trunks: nat=yes

I will then set externip and localnet to reflect the local setup,
UNLESS there is a functional SIP ALG doing the work in the gateway
device. I make this statement because I've found one or two firewalls
where it is best to disable the SIP ALG, and one or two where it is
best to leave it enabled.

The above always worked very well, but I now find my asterisk logs
being spammed with warnings containing lots of !! and I'd like to
know the best way to operate to achieve what I've always had while
following the new rules in order to be as secure as possible with
clean logs. I should add that we do not accept unsolicited
connections, and 99% of attempts to connect will be stopped at the
firewall.

Thanks,
Steve

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Re: [asterisk-users] SIP and NAT best practices since recent changes?

[Steve Davies]

On 11 January 2012 15:43, Kevin P. Fleming kpflem...@digium.com wrote:
 On 01/11/2012 05:29 AM, Steve Davies wrote:

 Hi,

 Since the recent update to the NAT configuration options and defaults
 in chan_sip.so, I am interested in any SIP/NAT best practices advice.

 What I've always done in the past is:

 Global: nat=no
 SIP handsets that are local: nat=no
 SIP handsets that are remote: nat=yes
 ITSP SIP trunks: nat=yes

 I will then set externip and localnet to reflect the local setup,
 UNLESS there is a functional SIP ALG doing the work in the gateway
 device. I make this statement because I've found one or two firewalls
 where it is best to disable the SIP ALG, and one or two where it is
 best to leave it enabled.

 The above always worked very well, but I now find my asterisk logs
 being spammed with warnings containing lots of !! and I'd like to
 know the best way to operate to achieve what I've always had while
 following the new rules in order to be as secure as possible with
 clean logs. I should add that we do not accept unsolicited
 connections, and 99% of attempts to connect will be stopped at the
 firewall.


 The simplest answer is to always use 'nat=yes' (or at least
 'nat=force_rport' in recent versions of Asterisk that support it), until you
 come across a SIP endpoint that fails to work properly with that setting. If
 you do come across such an endpoint, try hard to get it to work with that
 setting; if you can't, then set 'nat=no' for that endpoint, and understand
 that the endpoint's name could be discoverable using the attack methods
 previously disclosed. If the endpoint's configuration is suitably locked
 down (permit/deny, for example) this may not be a concern for you. If it's
 not locked down (for example, if it has to register to your Asterisk server
 from random locations), then the next step would be to seriously consider
 requesting that the user of that endpoint consider switching to some other
 SIP endpoint.

 To date, the only endpoints that have been identified that do *not* work
 with Asterisk's 'rport' handling forced upon them are Cisco phones.


Excellent. Thanks as always Kevin.

(Why am I not surprised about Cisco!)

Regards,
Steve

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Re: [asterisk-users] Is Asterisk 1.4 compatible with 1.8.7 ?

[Steve Davies]

On 28 December 2011 03:02, Joseph syscon...@gmail.com wrote:
 No, it makes no difference, on the other end is asterisk 1.4.39

 and 1.8.8 is still giving me:

  Executing [4@internal:1] Dial(SIP/11-0003,
 IAX2/home_server:@192.168.141.1/4,30,rw) in new stack
    -- Called IAX2/home_server:@192.168.141.1/4
 [Dec 27 20:00:16] WARNING[16398]: chan_iax2.c:10672 socket_process: Call
 rejected by 192.168.141.1: Unable to negotiate codec
    -- Hungup 'IAX2/192.168.141.1:4569-5678'
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [4@internal:2] Hangup(SIP/11-0003, ) in new stack
  == Spawn extension (internal, 4, 2) exited non-zero on 'SIP/11-0003'

 --
 Joseph

[snip]

Have you tried enabling IAX2 debug at both ends to see if the packet
decode provides any more clues?

Regards,
Steve

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Re: [asterisk-users] Asterisk 1.4.x segfaulting daily

[Steve Davies]

On 14 December 2011 12:56, Paulo Santos paulo.r.san...@sapo.pt wrote:
 Hello list,

 An Asterisk installation that was doing fine suddenly stared segfaulting a
 couple of times per day. I enabled all the logging and debugging to try to
 find a pattern but there was too much information to see exactly where it
 broke. So I enabled core dump and did backtraces and all of them seem to
 break on ast_setstate, setting the state to AST_STATE_DOWN. That's pretty
 much the only thing I can make of it, don't even know if that's correct.

 Does anyone have any ideas on why this is happening? The backtrace is
 attached.

 P.S.: I've switched the whole hardware already, including the BRI card
 (B400P, OpenVox). Also tried different versions of Asterisk, Dahdi and
 mISDN. I'm stuck with 1.4 Asterisk branch and mISDN v1.


If I was guessing, I'd say that the channel structure that is being
modified by the ast_setstate() call is incomplete, and contains some
garbage pointers.

If I was guessing further, I'd say that the callerID pointers are the
most likely candidate - Does the issue happen when a caller-id
withheld call is hung-up? hung-up before being answered perhaps?

You'd need to add some debug reporting into ast_setstate() to know for sure.

Just my 2p - 1.4.42 is an old version, so the chance of a solid answer
is fairly low.

Steve

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Re: [asterisk-users] one way voice with IVR

[Steve Davies]

On 17 October 2011 11:01, gincantalupo gincantal...@fgasoftware.com wrote:
 Hi,

 found where the problem is.I tried with a Grandstream phone and it
 works!!!

 The problem is my (crappy) Snom phone.

 I'm investigating the probhope to find the cause asap.


FYI: snom firmware 7.3.30 is very old. I remember it being quite
reliable, but it is still worth looking at the 8.4.32 release or
newer.

Regards,
Steve

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Re: [asterisk-users] Reinvite dialplan application [Was: OT - SIP - Toggle to autoanswer after ringing]

[Steve Davies]

On 5 October 2011 10:21, Nasir Iqbal na...@ictinnovations.com wrote:
 You can do this by an AMI based transfer (Redirect) to Local channel, and
 then in local channel's dialplan you need to add your desired custom sip
 header followed by a dial command.
 Nasir Iqbal

 ICT Innovations
 http://www.ictinnovations.com/


Broadcom invented some SIP NOTIFY extensions to cover this case -
Several open SIP handsets support the Broadcom extensions, which
revolve around sending a NOTIFY Event: hold or Event: talk

Asterisk does not support these NOTIFY messages at present, though I
expect they could be added reasonably simply.

Regards,
Steve

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[asterisk-users] Possibly odd sip.conf security requirements. Possible?

[Steve Davies]

Hi,

Is the following possible in some way? I want to have several SIP
providers able to send me calls, each provider may send calls into
many possible DDIs. Each provider has a cluster of servers, but is
unable to authenticate with me, so the following would be a sort of
pseudo-code sip.conf example.


[general]
context = barred  ; Unknown/other source of calls

[provider 1]
type = peer
context = provider1-context   ; deal with provider's calls 1
deny = 0.0.0.0/0.0.0.0
permit = 12.13.14.0/24  ; This provider has servers in this range

[provider 2]
type = peer
context = provider2-context   ; deal with provider's calls 2
deny = 0.0.0.0/0.0.0.0
permit = 22.23.24.0/24  ; This provider has servers in this range

[provider 3]
type = peer
context = provider3-context   ; deal with provider's calls 3
deny = 0.0.0.0/0.0.0.0
permit = 32.33.34.0/24  ; This provider has servers in this range


Normally a call into SIP has one of 3 paths:
1) Unauthenticated, so use the default
2) Identifiable username
3) Identifiable IP address

In the above example, we have a BLOCK of IP addresses instead of a
single address. Can this be made to work?

Thanks for any pointers.

Regards,
Steve

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Re: [asterisk-users] 1.6.2.20 ${DIALSTATUS} disagrees with CDR(answered)

[Steve Davies]

On 14 August 2011 08:36, Eric Wieling ewiel...@nyigc.com wrote:

 I am having a problem with ${DIALSTATUS} and )=CDR(disposition) disagreeing.  
 Below is a dialplan snippet and the resulting CLI output.  This is running in 
 an 'h' extension.

        Noop(DIALSTATUS=${DIALSTATUS})
        Noop(CDR(disposition)=${CDR(disposition)})

    -- Executing [h@pbxmax-dial-simple:1] NoOp(SIP/msx_01-005b, 
 DIALSTATUS=ANSWER) in new stack
    -- Executing [h@pbxmax-dial-simple:2] NoOp(SIP/msx_01-005b, 
 CDR(disposition)=NO ANSWER) in new stack

 Unless I seriously misunderstand the CDR(disposition) function, this looks 
 like a bug to me.  Does any else have this issue?


What call events lead up to that? There is a case I encountered some
time ago where a blind-transfer can mess-up the CDR disposition. I
started to re-factor all of the CDR code to fix all of these issues,
but with 1.6 EOL, 1.8 etc etc, it needs revisiting.

As far as I know, 1.8/1.10 has the same issues.

Regards,
Steve

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Re: [asterisk-users] Seg Faults with 1.6.2.19

[Steve Davies]

On 18 July 2011 12:03, Lee Archer lee.arc...@thebigword.com wrote:
 Hi, is anyone else having problems with the reload command crashing Asterisk
 1.6.2.19?  I’ve run a few tests and 1.6.2.18.2 doesn’t have this problem but
 1.6.2.19 after a few reloads is just dumping and restarting.

 Thanks

 Lee


I've not had a problem here with 1.6.2.19.

What are you reloading that causes the issue, and can you post the
usual gdb backtrace somewhere? Perhaps on the bug tracker.

Regards,
Steve

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Re: [asterisk-users] FAX with SIP

[Steve Davies]

On 18 July 2011 12:20, Eduardo Carpes car...@bsd.com.br wrote:
 Hello guys
 I need some help to do works FAX using SIP, anybody know the secret to
 this? Have asterisk 1.6.
 Thanks!!

 --
 Enviado do meu celular

 Eduardo Carpes
 E-mail: car...@bsd.com.br
 www.freebsd.org

The magic sauce that you need is T.38 - Asterisk 1.6 supports this
to a limited degree, and your ITSP will need to support it.

The sip.conf.sample file and the voip-info wiki has all the
information you need to try it out.

Regards,
Steve

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Re: [asterisk-users] Seg Faults with 1.6.2.19

[Steve Davies]

On 18 July 2011 13:00, Lee Archer lee.arc...@thebigword.com wrote:
 Hi Steve, I think it's related to my ODBC connection.  I started with a 
 random hang where it looked ODBC related which led me to try a few things.  
 Reloading the config a few times is causing core dumps which 1.6.2.18.2 just 
 doesn't have, however my main reason for using 1.6.2.19 is a fix to ODBC so I 
 don't really want to downgrade.  I will try and get some traces from one of 
 my test boxes.

 Thanks

 Lee


I can confirm that we are NOT using ODBC, and that our box does NOT
crash, so your theory is still holding up.

:)
Steve

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Re: [asterisk-users] Seg Faults with 1.6.2.19

[Steve Davies]

On 18 July 2011 14:05, Lee Archer lee.arc...@thebigword.com wrote:
 Seems to be an already reported problem but since no more fixes for 1.6
 it's back to 1.6.2.18.2 for me.

 https://issues.asterisk.org/jira/browse/ASTERISK-18103

 Regards

 Lee


If it is a regression introduced in 1.6.2.19, then it should still be fixed.

At least I believe that's the rules.

Regards,
Steve

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Re: [asterisk-users] OT: Google Plus

[Steve Davies]

On Saturday, 9 July 2011, Gordon Henderson gordon+aster...@drogon.net wrote:
 On Sat, 9 Jul 2011, Steve Davies wrote:


 On 9 July 2011 12:34, randulo rand...@randulo.com wrote:

 Go ahead and lambast me for this post, it isn't specific to Asterisk, but:

 G+ has only been open at all for a week and I already am chatting with
 over 200 people who are into VoIP, Asterisk and all the rest of the
 stuff we here care about. If you don't care or are anti-social, fine.
 But you owe it to yourself to check it, because a lot of cool VoIP
 people are there and after all, Google themselves  are doing some
 great stuff with VoIP, XMPP and video, and steadily moving towards
 open source. Come drink the Kool-Aid!



 Can you suggest a good way of finding/following appropriate
 VoIP/Asterisk people once on Google+? How do you then group them? Just
 in a Circle, or some other mechanism?


 I've just created a VoIPy circle - So I can then invite people I know into 
 the circle by email address, and/or looking at someone else's circles and 
 seing if they have something relevant in their summary tag and adding them 
 into your own circle... (Or using their people search - e.g. for 'randulo' :)

 You can have people in more than one circle. Right now, it's a bit like a 
 media-rich version of twitter with excellent filtering (the circles). I don't 
 have camera/microphone/speakers on my PC, (got real desk SIP phones!) so 
 haven't tried the audio/video chat yet, but the typing instant messaging 
 type chat works just fine.

 I think Google are still slowly gating people into + though. I did have some 
 invites, but seem to have used them all up now (google didn't tell me how 
 many, the invite button just went away after a while!)

 I'd love to see SIP integration into it, so I can use my existing SIP toys 
 with it.

 Gordon

Thanks for that Gordon. What appears to be missing at the moment is
the ability to interface or collaborate with a group of 'strangers'.

It would be good if there were a way to broadcast a 'we're here, come
join us' to bring a group of VoIP people together, a bit like an IRC
channel name can do, or a Facebook fan page.

I thought that sparks might cover that, but I'm not entirely sure how
sparks work yet.

I agree that SIP integration would be great. I think it'll be a while
yet but if anyone will allow it, it'll be Google.

Cheers,
Steve

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Re: [asterisk-users] OT: Google Plus

[Steve Davies]

On 9 July 2011 12:34, randulo rand...@randulo.com wrote:
 Go ahead and lambast me for this post, it isn't specific to Asterisk, but:

 G+ has only been open at all for a week and I already am chatting with
 over 200 people who are into VoIP, Asterisk and all the rest of the
 stuff we here care about. If you don't care or are anti-social, fine.
 But you owe it to yourself to check it, because a lot of cool VoIP
 people are there and after all, Google themselves  are doing some
 great stuff with VoIP, XMPP and video, and steadily moving towards
 open source. Come drink the Kool-Aid!


Can you suggest a good way of finding/following appropriate
VoIP/Asterisk people once on Google+? How do you then group them? Just
in a Circle, or some other mechanism?

Thanks,
Steve

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[asterisk-users] iLBC re-licence

[Steve Davies]

Does anybody know if the updated licence on iLBC makes it safe to
include in Asterisk when used in a commercial environment again?

  https://sites.google.com/site/webrtc/ilbc-freeware

It seems to require that the Google iLBC licence document is on the
box, but that otherwise it is free-to use by all in any way (BSD
licence style). I believe that prior to that there was a requirement
to register every commercial use of the codec with the licence holder,
or some-such thing?

Regards,
Steve

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Re: [asterisk-users] iLBC re-licence

[Steve Davies]

On 22 June 2011 17:14, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote:
 On 06/22/2011 03:32 PM, Steve Davies wrote:

 Does anybody know if the updated licence on iLBC makes it safe to
 include in Asterisk when used in a commercial environment again?

   https://sites.google.com/site/webrtc/ilbc-freeware

 It seems to require that the Google iLBC licence document is on the
 box, but that otherwise it is free-to use by all in any way (BSD
 licence style). I believe that prior to that there was a requirement
 to register every commercial use of the codec with the licence holder,
 or some-such thing?

 IANAL but that's what I also understand from reading the software license.
 Seems you just need to ship the license with the binaries and all should be
 well. Please also note that the IP Rights Grant seems to note some
 limitations that you may want to study before shipping to your customers.

 Regards,
 Patrick

Yes, it seems to grant the rights to use the iLBC implementation that
comes with WebRTC. I'll have to take a look and see how that compares
(if at-all) with the code that can be plugged into Asterisk.

Cheers,
Steve

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Re: [asterisk-users] Aastra phone # key in dialplan

[Steve Davies]

On 22 June 2011 17:09, marvin horst fivehor...@gmail.com wrote:
 I want to use extension numbers that begin with the # key in my dialplan,
 but I can't get my Aastra phone (6731i) to transmit the # key to asterisk.
 It works fine for the * key.

 I've tried numerous Local Dial Plan patterns in the aastra web configuration
 but none of them worked. My current Local dial Plan pattern is
 x+#|xx+*|#x+. Any help would be appreciated.


The following works here:

sip dial plan terminator: 1
sip digit timeout: 99
sip dial plan: X+^

I believe that '#' acts as a send button by default.

Regards,
Steve

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Re: [asterisk-users] Asterisk 1.8 broken MWI

[Steve Davies]

On 9 June 2011 15:49, satish patel satish...@hotmail.com wrote:
 Date: Wed, 8 Jun 2011 18:15:14 +0100
 From: davies...@gmail.com
 To: asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] Asterisk 1.8 broken MWI

 On 8 June 2011 17:20, satish patel satish...@hotmail.com wrote:
  Interesting thing is when i reload sip.conf  i got MWI lamp working on
  polycom 501
 
  But its not working when anyone leave voicemail. Do you know its some
  timeout or polling setting in sip.conf ?
 
  Still my question is my my asterisk not sending NOTIFY message ? Do i
  need
  to subscribe my phone to asterisk ?
 

 Does this help?

 https://issues.asterisk.org/jira/browse/ASTERISK-17866

 Regards,
 Steve

 Thanks steve,

 But you know if i connect X-lite softphone my asterisk sending NOTIFY .

 But its not sending NOTIFY to polycom 501 phone ? Do you think i need to
 subscribe my phone to asterisk ?

 -S


X-Lite automatically SUBSCRIBEs for MWI indication. Polycom and snom
do not do this by default, instead they assume that the REGISTER will
automatically cause MWI notifications.

chan_sip changed behaviour (by accident I suspect) somewhere between
version 1.2 and 1.6, and the patch basically puts back what went
missing. It is crude, but has not caused me any problems so far.

Regards,
Steve

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Re: [asterisk-users] Asterisk 1.8 broken MWI

[Steve Davies]

On 8 June 2011 17:20, satish patel satish...@hotmail.com wrote:
 Interesting thing is when i reload sip.conf  i got MWI lamp working on
 polycom 501

 But its not working when anyone leave voicemail. Do you know its some
 timeout or polling setting in sip.conf ?

 Still my question is my my asterisk not sending NOTIFY message ? Do i need
 to subscribe my phone to asterisk ?


Does this help?

https://issues.asterisk.org/jira/browse/ASTERISK-17866

Regards,
Steve

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[asterisk-users] Refactor of CDR - Comments please.

[Steve Davies]

Hi,

Since raising this ticket about broken CDR data:
https://issues.asterisk.org/jira/browse/ASTERISK-17826
I have been researching how CDR records work in various circumstances.
CEL will do most things that people want, but that does not change
that CDR records are likely to persist into future versions, and
should be as correct as possible.

Is this something people want? Is it worth the risk to change CDRs
again? I am working on a patch within 1.6.2, and if it is considered
worthwhile I will clean-up and migrate the patch to 1.8 and put it on
the review-board

It struck me that the fundamental issue was the bridge_cdr construct
that appears to have been added in version 1.4. While this solved the
problems it was aimed at solving, it appears to have caused other
problems, and perhaps even worsened support of CDR records when
channels are transferred/masqueraded. The special cdr reset feature
seems to be able to clear out valuable CDRs that are no-longer related
to the bridge when it ends, and copying bridge_cdr back into chan-cdr
often overwrites other valuable CDR data!

A classic example of the current failing is:
- A UK based receptionist calls Australia.
- Then a 2nd call leg is placed UK to America.
- The receptionist bridges the 2 calls.
The desired outcome would be 2 CDR records detailing the full duration
of both calls for billing purposes. Sadly, at present, one of the 2
CDRs stops when the transfer happens because Asterisk sees only one
bridge in progress.

Here is the outline of my solution:

1) There are 2 basic types of CDR
a) A CDR that tracks a running PBX/dialplan.
b) A CDR that tracks an outbound dialled channel.
2) Channels can be bridged and re-bridged. When bridging, the existing
code provides a way to merge caller (PBX) and callee (Dialed) CDRs,
and for consistency this will be maintained as closely as possible.
3) A CDR on a dialled channel should track that channel for its lifetime.
4) A CDR that tracks a running PBX should track that PBX for its
lifetime or until merged into a dialled call.
Side-note: 3) and 4) Affect the way cdr data is masqueraded.
5) Aim: Valid CDR output should be changed minimally over the current system.
6) Aim: NoCDR, ResetCDR and ForkCDR etc will remain untouched and
function the same.
7) A bridge CDR will be un-needed, instead, the bridged CDR will be
stored the dialed channel.
8) It is intended that all call legs will have a CDR, which will
accurately reflect the duration of that call leg, but no attempt will
be made to record transfers/masquerades beyond current mechanisms. CEL
data should provide that additional information if needed.

The changes to achieve the above are surprisingly minor (relatively
speaking). I am testing all of the cases that I can think of:

- Simple call in.
- Simple call out.
- Call-out, then blind transferred by caller and callee.
- Call-out, then att. transferred by caller and callee.
- AMP or call-file originated call.
- Masq-away of Local channel.
- Bridge() channels in the dialplan.
- Feature park of calls.
- Local/ channel calls.
- Feature transfer of calls.
- Transfer call to IVR/Playback
- Transfer IVR/Playback to a handset
- Feature Pickup and Pickup()
- SIP blind xfer
- SIP att xfer
- SIP xfer to ringing channel

Thoughts?

Many thanks,
Steve

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Re: [asterisk-users] busy hangup HDLC Bad FCS (8) on Primary D-channel

[Steve Davies]

On 1 June 2011 15:10, randall rand...@songshu.org wrote:
 On 06/01/2011 03:55 PM, randall wrote:
 On 06/01/2011 03:41 PM, Tzafrir Cohen wrote:
 On Wed, Jun 01, 2011 at 08:06:02AM +0200, randall wrote:
 Hi all,

 After running fine for a few months now asterisk seems to hang
 frequently , still functioning but the DAHDI channels seem busy  (users
 report a busy signal when calling or being called)

 A reboot will allow it to run for another day or maybe 2  or 3 till the
 problem occurs again.


 running stock Asterisk 1.6.2.9-2+squeeze2 on Debian with stock kernel
 2.6.32-5-686

 i get the following errors:
 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 
 2

 (happens on all 4 spans)

 and the following in dmesg:
 [ 9004.635323] NOTICE-xpd_bri: XBUS-00/XPD-01: D-Chan RX DROP: BADFCS: 252
 [ 9004.635332] NOTICE-xpd_bri: XBUS-00/XPD-01: D-Chan RX:    current
 packet[0..2]: 55 55 FC
 [ 9004.635340] NOTICE-xpd_bri: XBUS-00/XPD-01: Multibyte Drop: errno=-71


 Channel 0/1, span 1 got hangup, cause 18

 Is this happening in the middle of a call? Or only a while after the
 call ended?


 the bad fcs messages seem to happen random
 there seems to be a relation indeed, have seen them happen randomly
 quite spurious, but they indeed tend to happen a while after the call is
 made.

 the hangup happens when a call through DAHDI is attempted,
 (usually after it has been working fine for a while a day or 2)

In my experience, FCS errors are caused by line quality issues, and
usually (not always) are in the telco's equipment. If they are only
happening occasionally, it may be a marginal, but mostly-OK signal on
the wire.

Do you also get occasional poor-quality audio on calls? The issue will
happen more when a call is being setup, or is progressing because
there are more frames being exchanged when a call is in progress.

I have also seen a bad component or dry solder on a voice card cause
this, and even a badly made ISDN cable can be part of the problem. If
none of that helps, I would ask the telco to put a trace on the line.

Hope that helps,
Steve

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Re: [asterisk-users] AstManProxy

[Steve Davies]

On 20 May 2011 16:16, Ishfaq Malik i...@pack-net.co.uk wrote:
 On Fri, 2011-05-20 at 10:58 -0400, Leif Madsen wrote:
 On 11-05-20 09:37 AM, Ishfaq Malik wrote:
  Do many people use this?
  Is it reliable and safe?

 It may still work, but that code is quite old, and I'm not even sure it's
 necessary any more.

 Leif.


I use astmanproxy with Asterisk 1.6.2.18 - It works fine. The most
recent version is on Github, and is not that old. In fact that reminds
me that I really must upload my latest changes to my github fork of
the project!

 The reasons I'm considering it are as follows:
 Building a web page which uses AJAX to get information from the AMI
 every 10-30 seconds or so and not wanting to log on and off via AMI that
 many times.

 We'll soon be using multiple asterisk servers so having a single point
 of access would be very useful.

Astmanproxy does seem to offer that, but it is not a feature I've used.

 I'd love for you to elaborate on why it's not necessary any more, is
 there something simple I've overlooked?

- I run asterisk in high-priority mode, meaning that every AMI
connection gets a high priority socket - Astmanproxy runs at a
normal priority - That just feels like a nicer thing to be doing :)

- If a client fails to read its net socket fast enough, it affects
astmanproxy, and not asterisk with the backlog.. Again, that just
feels like a better way to do it.

- Astmanproxy offers filtering, which can be used to either reduce
load on the client, to present information about a single
device/channel only, or as a sanity layer to block or fix-up invalid
requests by a client.

Hope that helps.
Steve

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Re: [asterisk-users] AstManProxy

[Steve Davies]

On 24 May 2011 10:43, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote:
 On 05/24/2011 11:02 AM, Steve Davies wrote:
 [snip]

 I use astmanproxy with Asterisk 1.6.2.18 - It works fine. The most
 recent version is on Github, and is not that old. In fact that reminds
 me that I really must upload my latest changes to my github fork of
 the project!

 The last update to Dave Troy's version was in December 2008 and the last
 update to your repo was more than a year ago. Not very old but not crispy
 fresh either :)

 For the archives, Dave Troy's astmanproxy repo can be found here on github:
 https://github.com/davetroy/astmanproxy

 And Steve Davies' fork: https://github.com/davies147/astmanproxy

 I look forward to seeing your repo updated with your latest changes.


Repo updated. I have tried to merge all of the other changes that are
out there also.

Regards,
Steve

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Re: [asterisk-users] Asterisk 1.6.2.18, Cisco 79XX not registering

[Steve Davies]

On 6 May 2011 16:30, Eric Wieling ewiel...@nyigc.com wrote:
 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
 Cassius Smith
 Sent: Friday, May 06, 2011 11:23 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] Asterisk 1.6.2.18, Cisco 79XX not
 registering

 Hi all,
 I have a production server running with about 90 Cisco
 79[46]1's and SIP release 8.5(2)SR1 from last year. I was
 running Asterisk 1.6.2.9 and upgraded last night after hours.
 (Seemed low risk to me!)

 Much to my surprise, not a single one of the Cisco 79XX
 phones would register. Since it's a production server, I
 rolled back to 1.6.2.9 and everything was fine. All my
 Linksys SPA phones and Polycom speaker phones registered just fine.

 I am now setting up  test servers with both 1.6.2.18 and
 1.8.3.3 to collect some debug.

 I am just curious - has anyone else had SIP issues with these
 phones and updating Asterisk broke them?

 I will post results of my findings after I have time to collect them.

 Cassius Smitha


 I seem to recall this issue mentioned on asterisk-dev.  Check 
 issues.digium.com and see if there is anything similar to your issue.


I also remember this being mentioned - I believe it was fixed in the
chan_sip Via: header handling code. The fix is in branches/1.6.2
already, so you should be able to grab the patch without too much
trouble.

Regards,
Steve


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[asterisk-users] IAX2 codec selection and video

[Steve Davies]

Hi,

Can anyone let me know how I can enable video (h.263) on SIP, but if a
video call is passed over IAX, it will remove the video and pass the
audio only.

What I tried was:

SIP - videosupport=yes
  - disallow=all
  - allow=alaw
  - allow=h263

IAX - disallow=all
  - allow=alaw


What appears to occur is that the SIP call negotiates h263 video, and
when passed over IAX, the h263  frames are passed, and are also
accepted at the far end which also does not have a video codec
allowed. Should that be happening? This is with 1.6.2.18-rc1. Am I
missing a setting somewhere?

Thanks,
Steve

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Re: [asterisk-users] Possible bug in Hangup() (Asterisk 1.4.x)

[Steve Davies]

On 15 April 2011 13:02, Vlasis Hatzistavrou vh...@kinetix.gr wrote:
 Hello,

 On an Asterisk 1.4.33.1 in a simple scenario:

 [test]
 exten = _X.,1,Dial(SIP/12345@peer01,,,)

 exten = i,1,Hangup(${HANGUPCAUSE})
 exten = t,1,Hangup(${HANGUPCAUSE})
 exten = h,1,Hangup(${HANGUPCAUSE})


 I have noticed that no matter what value we set in the Hangup(cause code)
  commands, if the call is not answered by peer01 for any reason, the actual
 cause code returned to the calling party is a 503, no matter what the
 ${HANGUPCAUSE} is.

 Even if we set a fixed value like Hangup(1) (which should give a 404) or
 Hangup(17) (which should give a 486), the cause code returned is always a
 503.

 Has anyone else noticed this? I went through the issue tracker but I
 couldn't find any relevant bug posted in the past. I am certain that in
 previous versions I could set the reply message to the desired value, so I
 wonder if this is a bug in this particular version (1.4.33.1).


Strictly speaking you can only Hangup (BYE) an answered and fully
established call. In SIP terms, a hangup that occurs before an answer
is a CANCEL, and I believe a CANCEL is always represented by a 503
code in chan_sip.

Regards,
Steve

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Re: [asterisk-users] No ringback even though progressinband=yes is set

[Steve Davies]

On 7 April 2011 23:04, Douglas Mortensen d...@impalanetworks.com wrote:
 Steve. Thanks for the insight. I won't pretend to know what early-audio is, 
 but I guess I'm about to find out :-).

 Also, I believe that I have a nearly identical setup like this with the exact 
 same SIP provider w/o any trouble. However, I think that system must be 
 running asterisk 1.4 or 1.2 (my guess is 1.4, but I'll have to check to 
 confirm). Is there a significant difference between 1.2/1.4  1.6 in this 
 scenario?

 Thanks a million!! :-)

 -
 Doug Mortensen
 Network Consultant
 Impala Networks
 P: 505.327.7300
 .


 -Original Message-
 From: Steve Davies [mailto:davies...@gmail.com]
 Sent: Thursday, April 07, 2011 10:49 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] No ringback even though progressinband=yes is 
 set

 On 7 April 2011 17:02, Douglas Mortensen d...@impalanetworks.com wrote:
 Any ideas on why callers who call into my customer's SIP trunk are not 
 hearing a ringback tone? I had this on one other asterisk system, and wound 
 up needing to set progressinband=yes in the SIP trunk config.

 I have set this on the current system  restarted asterisk, but to no avail.

 I am using:

 AsteriskNOW distro
 Asterisk build is 1.6 from AsteriskNOW repository:
 asterisk16-1.6.2.17.2-1_centos5 FreePBX 2.9

 Any help would be greatly appreciated! :-)

 -
 Doug Mortensen


 In my personal experience with SIP and 1.6.x, that mostly depends on where 
 you are sending the call to. It depends on whether the next or subsequent leg 
 tries to use early-audio for the ring tone, or uses a Ringing event to signal 
 that is what is happening. It then depends on whether the originating 
 caller's equipment can understand early-audio ringing.

 We have a setup here where all our trunks support early-audio ringing except 
 one (an ISDN30 circuit) and we have to juggle things a bit sometimes to 
 ensure ringing occurs.

 Perhaps provide more details? Or you may find that tracing the SIP gives you 
 the clue that you need.

 Hope that helps,
 Steve



Early audio is audio that is sent before the call is answered,
usually in the form of a custom ring-tone or perhaps a cannot
connect, try later message. Some systems do not support it as it can
be abused to communicate at least basic information for free.

We had a problem with this when connecting Asterisk 1.2 to Asterisk
1.6 via IAX. A 1.2 SIP system will automatically switch into early
audio if it sees an early audio frame. 1.6 defaults to not doing this,
but there is a parameter to re-enable it. In this case we solved the
problem by upgrading to 1.6 everywhere :)

Regards,
Steve

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Re: [asterisk-users] No ringback even though progressinband=yes is set

[Steve Davies]

On 7 April 2011 17:02, Douglas Mortensen d...@impalanetworks.com wrote:
 Any ideas on why callers who call into my customer's SIP trunk are not 
 hearing a ringback tone? I had this on one other asterisk system, and wound 
 up needing to set progressinband=yes in the SIP trunk config.

 I have set this on the current system  restarted asterisk, but to no avail.

 I am using:

 AsteriskNOW distro
 Asterisk build is 1.6 from AsteriskNOW repository: 
 asterisk16-1.6.2.17.2-1_centos5
 FreePBX 2.9

 Any help would be greatly appreciated! :-)

 -
 Doug Mortensen


In my personal experience with SIP and 1.6.x, that mostly depends on
where you are sending the call to. It depends on whether the next or
subsequent leg tries to use early-audio for the ring tone, or uses a
Ringing event to signal that is what is happening. It then depends on
whether the originating caller's equipment can understand early-audio
ringing.

We have a setup here where all our trunks support early-audio ringing
except one (an ISDN30 circuit) and we have to juggle things a bit
sometimes to ensure ringing occurs.

Perhaps provide more details? Or you may find that tracing the SIP
gives you the clue that you need.

Hope that helps,
Steve

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[asterisk-users] SIP channel able to add codecs once up and running?

[Steve Davies]

From my observations, if a video capable device starts the call in
non-video mode, it is never able to add video to the channel? Is this
correct, or am I missing something?

It looks as if the codec 'jointcapability' is calculated at the start
of the call, and can never be added to (with exceptions for T.38 fax)
as any SDP update is masked using the existing 'jointcapability' and
knocks out the newly requested codec.

Is that right?

Thanks,
Steve

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Re: [asterisk-users] Registration from '000000 x 1000

[Steve Davies]

On 2 April 2011 09:46, Jonas Kellens jonas.kell...@telenet.be wrote:
 Hello list,

 I often see the following in my message log :

 [Apr  2 08:15:01] NOTICE[22988] chan_sip.c: Registration from '00
 sip:00@MY-IP' failed for '184.106.109.168' - No matching peer found
 [Apr  2 08:15:01] NOTICE[22988] chan_sip.c: Registration from '00
 sip:00@MY-IP' failed for '184.106.109.168' - No matching peer found
 [Apr  2 08:15:01] NOTICE[22988] chan_sip.c: Registration from '00
 sip:00@MY-IP' failed for '184.106.109.168' - No matching peer found
 [Apr  2 08:15:01] NOTICE[22988] chan_sip.c: Registration from '00
 sip:00@MY-IP' failed for '184.106.109.168' - No matching peer found
 [Apr  2 08:15:01] NOTICE[22988] chan_sip.c: Registration from '00
 sip:00@MY-IP' failed for '184.106.109.168' - No matching peer found

 And there are hundreds of them...


 Is there a setting so I can make Asterisk not respond to SIP PEER
 registrations which are not in my sip.conf or my realtime MySQL DB ??

Yes, you add a rule to your firewall! Even better, get it filtered
further out so that it does not waste your inbound Internet bandwidth,
because in my experience, once those SIP spammers start, they continue
for weeks at the very least.

IIRC, the way SIP registrations works basically requires than an
failed/un-authorised attempt is responded to, so that the other party
knows to authenticate. If you stop sending that response, no-one can
authenticate.

Hope that helps.
Steve

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[asterisk-users] DAHDI, IAX2 and SIP considerations for Early-Media / Alerting

[Steve Davies]

Hi,

Short version:

Is it possible or even legal to convert an IAX2 PROGRESS/EARLY-MEDIA
indication into a DAHDI/q.931 ALERTING signal when your ISDN provider
does not pass early media on receipt of an PROGRESS(8) indication?

Long version:

I have an Asterisk 1.6.2.18-rc1 based system with a DAHDI trunk (UK E1
line), also, the system has IAX2 trunks, and several SIP handsets.

All 3 protocols (q.931/IAX2/SIP) have a mechanism to indicate either
ALERTING/RINGING, or to specify PROGRESS/EARLY-MEDIA. Based on this
you'd think call setup would all work happily all of the time :) What
happens based on the call direction is as follows:

SIP - DAHDIISDN returns ALERTING, SIP uses 180 Ringing, all OK
SIP - IAX2 IAX2 returns PROGRESS, SIP uses 183 Progress, early
audio works OK
IAX2 - DAHDI   ISDN returns ALERTING, IAX2 uses RINGING, all OK
IAX2 - SIP SIP returns 180 ringing, IAX2 uses RINGING, all OK
DAHDI - SIPSIP returns 180 ringing, ISDN uses ALERTING, all OK
DAHDI - IAX2   IAX2 returns PROGRESS, ISDN uses PROGRESS(8), but the
caller hears no ringing.

I believe that my issue is that my UK ISDN provider does not accept
early media, and will simply send silence instead of using the
provided early audio stream. DAHDI is configured with:
       priindication=outofband
The IAX2 trunk provider is using early-media to send the ringing tone,
and as above, this mostly seems to work okay. The exception is when
the call is bridged to ISDN, where I believe the ISDN provider does
not pass on early media.

I checked the IAX2 RFCs 5456/5457, but cannot find a definition of how
RINGING/PROGRESS is meant to work. Is my IAX2 trunk provider doing
something wring by not also sending RINGING?

Is there a workaround that converts either IAX2 PROGRESS into RINGING,
or allows DAHDI to send ALERTING if it receives an early media
indication? I suspect the code to do the latter would be reasonably
simple, but would appreciate pointers for any badness that it may
cause.

Thanks in advance for any suggestions.

Regards,
Steve

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Re: [asterisk-users] One Way Audio

[Steve Davies]

On 10 March 2011 11:17, Ishfaq Malik i...@pack-net.co.uk wrote:
 Just fixed our problem with

 directmedia=no

 but this only applies if your extensions are behind a nat

 Ish


There are several reasons why directmedia=no might be the correct
configuration.

1) NAT - probably the most common reason
2) Routing - Sometimes devices cannot route to each other directly
3) ITSP calls. Many SIP providers will not accept a redirect

and I am sure there are many more...

Cheers,
Steve

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Re: [asterisk-users] IAX channel name incorrect - Found in 1.2 still happens in 1.6

[Steve Davies]

*Bump* No takers? Perhaps no-one else thinks this is a bug?

Regards,
Steve

On 7 February 2011 16:45, Steve Davies davies...@gmail.com wrote:
 Hi,

 The following IAX config (slightly edited) causes an issue for me in
 version 1.6.2.16.1, where my CDR data is unreliable.

 [user1]
 type=friend
 auth=md5
 accountcode=user1
 notransfer=yes
 context=context1
 host=10.0.0.250
 username=user1
 secret=secret1
 disallow=all
 allow=alaw

 [user2]
 type=friend
 auth=md5
 accountcode=user2
 notransfer=yes
 context=context2
 host=dynamic
 deny=0.0.0.0/0.0.0.0
 permit=10.0.0.0/24
 username=user2
 secret=
 disallow=all
 allow=alaw

 If a call comes in from 10.0.0.250, using username user2 and with no
 password, then it is correctly authenticated against the [user2]
 section.
    Accountcode is set to user2
    Context is set to context2
 and the call mostly proceeds correctly, BUT the source channel name is
 set to IAX2/user1-, which is then seen both in the dialplan debug
 output, and in the CDR. I would expect the channel name to reflect the
 section name that was used to authenticate the call ie.
 IAX2/user2-; I specifically put a password onto [user1] so there
 is no possibility that the call is authenticating there.

 Am I missing something? Or is this a bug?

 Thanks,
 Steve


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Re: [asterisk-users] Aastra phones cannot transfer calls?

[Steve Davies]

On 16 February 2011 00:22, Ernie Dunbar maill...@lightspeed.ca wrote:
 At 12:12 PM 2/15/2011, you wrote:
I have two Aastra phones, a 6730 and a 6757, both connected to Asterisk
v1.6.2.1. They can call each other's extensions (and make and receive
calls otherwise), but they cannot transfer calls, not even to outside

 I'm running 1.6.2.16.1 and have three Aastra 480i phones and have had
 no problem at all with transfers. Have you considered trying a newer
 version?


 Nope. Upgraded to 1.6.2.16.1, and I still see the same effect.

 It may be a setting on the phone or a SIP setting. I'll investigate this
 elsewhere but report back about the solution.



I also tried this with a 6757i and a 6753i with no problems (blind and
attended) on Asterisk 1.6.2.16.1. Have you updated the handset
firmware to 2.6.0.2010?

Cheers,
Steve

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Re: [asterisk-users] Barge in.

[Steve Davies]

On 16 February 2011 10:13, Peter den Hartog peterdenhar...@gmail.com wrote:
 I'm running Asterisk 1.6 and was wondering if anybody have a workig barge
 in solution running.
 I was thinking of using chanspy, but i would like that the original call
 would be dropped, and the new call would be the only one there.

What you are describing looks to me like a third party controlled
transfer, and not a barge-in at all.

I suspect that the Asterisk Manager API action Redirect will be your friend.

Regards,
Steve

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[asterisk-users] IAX channel name incorrect - Found in 1.2 still happens in 1.6

[Steve Davies]

Hi,

The following IAX config (slightly edited) causes an issue for me in
version 1.6.2.16.1, where my CDR data is unreliable.

[user1]
type=friend
auth=md5
accountcode=user1
notransfer=yes
context=context1
host=10.0.0.250
username=user1
secret=secret1
disallow=all
allow=alaw

[user2]
type=friend
auth=md5
accountcode=user2
notransfer=yes
context=context2
host=dynamic
deny=0.0.0.0/0.0.0.0
permit=10.0.0.0/24
username=user2
secret=
disallow=all
allow=alaw

If a call comes in from 10.0.0.250, using username user2 and with no
password, then it is correctly authenticated against the [user2]
section.
Accountcode is set to user2
Context is set to context2
and the call mostly proceeds correctly, BUT the source channel name is
set to IAX2/user1-, which is then seen both in the dialplan debug
output, and in the CDR. I would expect the channel name to reflect the
section name that was used to authenticate the call ie.
IAX2/user2-; I specifically put a password onto [user1] so there
is no possibility that the call is authenticating there.

Am I missing something? Or is this a bug?

Thanks,
Steve

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Re: [asterisk-users] queue_log in MySQL database

[Steve Davies]

On 13 January 2011 16:28, Jonas Kellens jonas.kell...@telenet.be wrote:


 I actually found this :
 http://www.voip-info.org/wiki/view/Asterisk+queue_log+on+MySQL

 But a second question :

 how can I know how long a caller stayed inside the queue untill it was
 answered by a member ??



The queue_log table contains exactly that information - Along with a
few other events, it indicates when a caller joined a queue, and when
an agent gets given the call. Take the difference between the 2 times
and you have the number that you need.

Cheers,
Steve

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Re: [asterisk-users] No MOH with parked call

[Steve Davies]

On 24 December 2010 15:44, Steve Davies davies...@gmail.com wrote:
 On 23 December 2010 18:01, Steve Davies davies...@gmail.com wrote:
 Hi Again,

 I thought I had this sorted, but it appears that in a clean
 environment I did not in fact fix it. There appears to be a bit of a
 contradiction.

 1) In 1.6.2.x, musiconhold requires DAHDI (which we have)
 2) In 1.6.2.x parked calls get MOH only if res_timing_dahdi is not loaded...

 I am confused. MOH in general terms works just fine, but if I park a
 call with res_timing_dahdi loaded, I get silence after the orbit
 announcement. If I unload res_timing_dahdi, all works fine, but my
 timing sources are less reliable.

 I have backported res_musiconhold.c from 1.8 to 1.6.2.16-rc1, but this
 does not seem to fix things - is the problem elsewhere? Is there a fix
 that I can try, or perhaps backport?

 Further to this, I have been slowly tracing through the codepath for a
 parked call - ast_settimer is called correctly for the MOH generator,
 and seems to set up the DAHDI timer exactly the same way as it does
 for the alternative timing modules, and all of the setup calls return
 success. I don't have a verbose output trace to hand, but the sequence
 seems to be as follows:

 -- Start moh_files_generator successfully using ast_settimer()
   Chunks of MOH file are generated here
 -- Stop moh_files_generator successfully
 -- Play the parking position numbers
 -- Start moh_files_generator using ast_settimer()
 Generator is never called

 Any thoughts where to look next? It is odd that it appears to be
 working immediately before the position is read out, but then fails to
 restart afterwards...


Thanks to Lee pointing me in the right direction, I discovered

https://issues.asterisk.org/view.php?id=18262

Which seems to fix this issue.

Many thanks,
Steve

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Re: [asterisk-users] One way crappy audio in iax call - Asterisk 1.6.2.15

[Steve Davies]

On 24 December 2010 14:40, Administrator TOOTAI ad...@tootai.net wrote:
 Hi,

 We had 2 asterisk 1.4 connected together in iax, all was fine. One of them
 was upgraded (server and Asterisk) in 1.6.2.15, the other end is in 1.4.38

 When calling to 1.4 to 1.6.2 -remember, it's iax- all is good. But calling
 from 1.6.2 to 1.4 give a bad audio to calling party (words are cutted, you
 can't understand the words). On callee party it's still good.

 We replace 1.6.2 version with 1.4.38 and everything is going back to normal,
 good audio on both side does'nt matter who call.

 I already opened another thread about problem with iax and Asterisk 1.6.2
 (rsa auth not working anymore). Are there some known problems with iax and
 1.6 version of Asterisk?

 Thanks for any hint


Not 100% sure, but I think there was a fix for IAX audio in
1.6.2.16-rc1 - Perhaps try that?

Regards,
Steve

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Re: [asterisk-users] No MOH with parked call

[Steve Davies]

On 23 December 2010 18:01, Steve Davies davies...@gmail.com wrote:
 Hi Again,

 I thought I had this sorted, but it appears that in a clean
 environment I did not in fact fix it. There appears to be a bit of a
 contradiction.

 1) In 1.6.2.x, musiconhold requires DAHDI (which we have)
 2) In 1.6.2.x parked calls get MOH only if res_timing_dahdi is not loaded...

 I am confused. MOH in general terms works just fine, but if I park a
 call with res_timing_dahdi loaded, I get silence after the orbit
 announcement. If I unload res_timing_dahdi, all works fine, but my
 timing sources are less reliable.

 I have backported res_musiconhold.c from 1.8 to 1.6.2.16-rc1, but this
 does not seem to fix things - is the problem elsewhere? Is there a fix
 that I can try, or perhaps backport?

Further to this, I have been slowly tracing through the codepath for a
parked call - ast_settimer is called correctly for the MOH generator,
and seems to set up the DAHDI timer exactly the same way as it does
for the alternative timing modules, and all of the setup calls return
success. I don't have a verbose output trace to hand, but the sequence
seems to be as follows:

-- Start moh_files_generator successfully using ast_settimer()
   Chunks of MOH file are generated here
-- Stop moh_files_generator successfully
-- Play the parking position numbers
-- Start moh_files_generator using ast_settimer()
Generator is never called

Any thoughts where to look next? It is odd that it appears to be
working immediately before the position is read out, but then fails to
restart afterwards...

Thanks,
Steve

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Re: [asterisk-users] No MOH with parked call

[Steve Davies]

On 7 December 2010 17:47, Steve Davies davies...@gmail.com wrote:
 On 7 December 2010 15:00, Steve Davies davies...@gmail.com wrote:
 On 7 December 2010 14:17, Lee Archer lee.arc...@thebigword.com wrote:
 Hi, try unloading res_timing_dahdi.so then trying again.

 Lee

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
 Davies
 Sent: 07 December 2010 12:54
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] No MOH with parked call

 Hi,

 Has anybody else noticed that MOH does not play on parked calls in
 1.6.2.14? Or is it just my setup? MOH seems to work in every other
 respect (Call Held or in-Queue), but when a call is parked, the logs
 show MOH being started, but the parked party hears nothing.

 The verbose logs show the following. Any thoughts on whet to check next?

 Thanks,
 Steve


 ### Call comes in here and is answered
    -- SIP/snom360-0d6f answered DAHDI/2-1
    -- Executing [...@macro-set-moh-call:1] GotoIf(SIP/snom360-0d6f,
 0?done) in new stack
    -- Executing [...@macro-set-moh-call:2] Set(SIP/snom360-0d6f,
 CHANNEL(musicclass)=m-default) in new stack
    -- Executing [...@macro-set-moh-call:3] NoOp(SIP/snom360-0d6f,
 ) in new stack

 ### Here the call is being blind transferred to the Park number
    -- Started music on hold, class 'default', on DAHDI/2-1
    -- Stopped music on hold on DAHDI/2-1
  == Spawn extension (local, 210, 1) exited non-zero on 'DAHDI/2-1'
    -- Executing [...@local:1] ForkCDR(DAHDI/2-1, ) in new stack
    -- Executing [...@local:2] Set(DAHDI/2-1, CDR(userfield)=) in
 new stack

 ### Not sure why I send Ringing here, but I tried NoOP() and
 Answer() too just in case
    -- Executing [...@local:3] Ringing(DAHDI/2-1, ) in new stack
    -- Executing [...@local:4] Set(DAHDI/2-1,
 CHANNEL(musicclass)=default) in new stack
    -- Executing [...@local:5] Set(DAHDI/2-1,
 CHANNEL(parkinglot)=default) in new stack
    -- Executing [...@local:6] Goto(DAHDI/2-1,
 parkedcalls_default,park,1) in new stack
    -- Goto (parkedcalls_default,park,1)
    -- Executing [p...@parkedcalls_default:1] Park(DAHDI/2-1, ) in
 new stack
  == Parked DAHDI/2-1 on 211 (lot default). Will timeout back to
 extension [parkedcalls_default] s, 1 in 90 seconds
    -- Added extension '211' priority 1 to parkedcalls_default
 (0xbe2e528)

 # The 211 announcement is heard perfectly
    -- DAHDI/2-1 Playing 'digits/2.alaw' (language 'en')
  == Extension Changed 211[extensions] new state InUse for Notify User
 steve
    -- DAHDI/2-1 Playing 'digits/1.alaw' (language 'en')
    -- DAHDI/2-1 Playing 'digits/1.alaw' (language 'en')

 # The system claims to start MOH default which works elsewhere, but
 the caller gets silence
    -- Started music on hold, class 'default', on DAHDI/2-1
  == Spawn extension (parkedcalls_default, s, 1) exited non-zero on
 'Parked/DAHDI/2-1ZOMBIE'


 Unloading res_timing_dahdi.so worked to fix MOH for Parked calls, but
 it has killed call quality on ISDN calls - I think it interferes with
 the software echo canceller somehow.

 Is there a ticket open on this? A patch to try?

 Thanks,
 Steve

 For anyone searching/finding this issue, the patch here:

  https://issues.asterisk.org/view.php?id=17726

 Applies to 1.6.2 with only a trivial tweak, and with minimal testing
 is working here. We now get music on hold when a call is parked, even
 when we are using res_timing_dahdi.so - Call quality remains high
 under these circumstances too.


Hi Again,

I thought I had this sorted, but it appears that in a clean
environment I did not in fact fix it. There appears to be a bit of a
contradiction.

1) In 1.6.2.x, musiconhold requires DAHDI (which we have)
2) In 1.6.2.x parked calls get MOH only if res_timing_dahdi is not loaded...

I am confused. MOH in general terms works just fine, but if I park a
call with res_timing_dahdi loaded, I get silence after the orbit
announcement. If I unload res_timing_dahdi, all works fine, but my
timing sources are less reliable.

I have backported res_musiconhold.c from 1.8 to 1.6.2.16-rc1, but this
does not seem to fix things - is the problem elsewhere? Is there a fix
that I can try, or perhaps backport?

Thanks for any pointers.

Regards,
Steve

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Re: [asterisk-users] Asterisk 1.6 produces *many* zombie processes on Debian.

[Steve Davies]

On 21 December 2010 22:06, Tilghman Lesher tilgh...@meg.abyt.es wrote:
 On Monday 20 December 2010 14:39:36 Ernie Dunbar wrote:
 We have an issue with our Asterisk install where Asterisk produces many
 Zombie processes (on the order of several hundred per minute) until
 either the Asterisk server is restarted (and the zombies die a natural
 death), or the kernel runs out of PID space (happens within hours) and
 brings the system to a halt.

 This problem only happens when the server is under some non-trivial
 load. We were testing this server with 8 SCCP phones, making up to five
 simultaneous calls through the DAHDI interface (a Digium Wildcard
 TE410P/TE405P (1st Gen)). Once our customers (nearly all SIP clients)
 start logging on and we get around 7 or 8 simultaneous DAHDI calls,
 Asterisk starts producing zombie processes at a high rate.

 I know what the issue is.  Please open a report on
 https://issues.asterisk.org and I'll get a patch uploaded pronto.


Please let us know the issue number once raised - I'd like to follow this one.

Regards,
Steve

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Re: [asterisk-users] Asterisk hangs up call after 20s

[Steve Davies]

On 22 December 2010 12:44, Gilles codecompl...@free.fr wrote:
 Hello

        I have an Asterisk 1.4 server and two XLite softphones, where
 Asterisk and the local XLite phone are located in a LAN behind a NAT
 router, and the remote XLite phone is located elsewhere on the Net
 behind its own NAT router:

 http://img252.imageshack.us/img252/3749/asterisknat.png

 I'm having the following issue: When the _local_ XLite calls out the
 remote XLite, everything works fine; However, when the _remote_ XLite
 calls the local XLite, things work OK until precisely 20s, where
 Asterisk decides to hang up, and displays the following error message
 in the console:

 ==
 WARNING[593]: chan_sip.c:1948 retrans_pkt: Maximum retries exceeded on
 transmission
 e45ed578253b9f3dMTRiYTg2OTI0YjExYjUzZWFiNDk3ZjZjMmRlMTQ4NjM. for seqno
 2 (Critical Response)

 WARNING[593]: chan_sip.c:1972 retrans_pkt: Hanging up call
 e45ed578253b9f3dMTRiYTg2OTI0YjExYjUzZWFiNDk3ZjZjMmRlMTQ4NjM. - no
 reply to our critical packet.
  == Spawn extension (my-phones, local-xlite-extension, 1) exited
 non-zero on 'SIP/unused-008008e4'
 ==

 I'm no SIP expert, but based on the debug session, before deciding to
 hang up, Asterisk tries 6 times to send an OK message to the remote
 XLite, and doesn't seem to get an answer. FWIW, after Asterisk has
 hung up, the remote XLite remains off-hook, oblivious to this error
 and keeps displaying Call established:

 www.pastebin.com/x6MgnrpG

 There's also this oddity on line 50: Scheduling destruction of SIP
 dialog.

 FWIW, in sip.conf, for the remote XLite user, I tried nat=no and
 nat=yes, with no difference. I'm actually not sure how to configure
 a remote user which happens to be listed in sip.conf (it's behind a
 NAT router but it registers with Asterisk, so... is it NATed or not?),
 and am surprised it actually rings and sends/receives voice with no
 problem, regardless of this parameter.

 I found discussions about using t1min=500 in sip.conf, but it made
 no difference either.

 Has someone already experienced this and knows what can be done?

 Any hint much appreciated.


Look in the XLite advanced network settings and disable the 2 timeout
settings (RTP and RTCP?). This is not always necessary, but there are
sufficient cases where the packets XLite expects appear early on, but
do not persist, thus causing a hangup. I think the default timeout is
20 seconds.

Cheers,
Steve

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[asterisk-users] UDP buffer overflows?

[Steve Davies]

Hi,

On one of our asterisk systems that is quite busy, we are seeing the
following from 'netstat -s':

Udp:
17725210 packets received
36547 packets to unknown port received.
44017 packet receive errors
17101174 packets sent
RcvbufErrors: 44017  --- this

When this number increases, we see SIP errors, and in particular
Qualify packets are lost, and temporarily disable handsets, causing
all sorts of minor chaos.

I have already tuned from the defaults of:

 net.core.rmem_max = 131071
 net.core.wmem_max = 131071
 net.core.rmem_default = 111616
 net.core.wmem_default = 111616
 net.core.optmem_max = 10240
 net.core.netdev_max_backlog = 1000

up to:

 net.core.rmem_max = 1048575
 net.core.wmem_max = 1048575
 net.core.rmem_default = 1048575
 net.core.wmem_default = 1048575
 net.core.optmem_max = 1048575
 net.core.netdev_max_backlog = 1

with no luck.

Any more suggestions?

Many thanks,
Steve

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Re: [asterisk-users] UDP buffer overflows?

[Steve Davies]

On 10 December 2010 16:45, Steve Davies davies...@gmail.com wrote:
 Hi,

 On one of our asterisk systems that is quite busy, we are seeing the
 following from 'netstat -s':

 Udp:
    17725210 packets received
    36547 packets to unknown port received.
    44017 packet receive errors
    17101174 packets sent
    RcvbufErrors: 44017                  --- this

[snip]

  net.core.rmem_max = 1048575
  net.core.wmem_max = 1048575
  net.core.rmem_default = 1048575
  net.core.wmem_default = 1048575
  net.core.optmem_max = 1048575
  net.core.netdev_max_backlog = 1


Additional question - Do I need to restart Asterisk for these settings
to apply to SIP? I have not done so yet, and further reading suggests
that this is a per-process buffer and may be assigned when the
listener is created.

All pointers gratefully received :)
Steve

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Re: [asterisk-users] UDP buffer overflows?

[Steve Davies]

On 10 December 2010 17:21, Shaun Ruffell sruff...@digium.com wrote:
 On 12/10/2010 11:02 AM, Steve Davies wrote:
 On 10 December 2010 16:45, Steve Davies davies...@gmail.com wrote:
 Hi,

 On one of our asterisk systems that is quite busy, we are seeing the
 following from 'netstat -s':

 Udp:
    17725210 packets received
    36547 packets to unknown port received.
    44017 packet receive errors
    17101174 packets sent
    RcvbufErrors: 44017                  --- this

 [snip]

  net.core.rmem_max = 1048575
  net.core.wmem_max = 1048575
  net.core.rmem_default = 1048575
  net.core.wmem_default = 1048575
  net.core.optmem_max = 1048575
  net.core.netdev_max_backlog = 1


 Additional question - Do I need to restart Asterisk for these settings
 to apply to SIP? I have not done so yet, and further reading suggests
 that this is a per-process buffer and may be assigned when the
 listener is created.


 I had to double check myself in the kernel sources (in
 net/core/sock.c:sock_init_data), but yes you will need to recreate the
 socket for the rmem_default option to take effect.

 I was just about to ask if changing the sysctl changed the frequency
 that you see the error counter increment when I saw your follow on.

 Cheers,
 Shaun

Thanks for making the extra effort Shaun :)  I will restart Asterisk
when I get a chance, and re-check.

Regards,
Steve

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Re: [asterisk-users] UDP buffer overflows?

[Steve Davies]

On 10 December 2010 17:33, Steve Davies davies...@gmail.com wrote:
 On 10 December 2010 17:21, Shaun Ruffell sruff...@digium.com wrote:
 On 12/10/2010 11:02 AM, Steve Davies wrote:
 On 10 December 2010 16:45, Steve Davies davies...@gmail.com wrote:
 Hi,

 On one of our asterisk systems that is quite busy, we are seeing the
 following from 'netstat -s':

 Udp:
    17725210 packets received
    36547 packets to unknown port received.
    44017 packet receive errors
    17101174 packets sent
    RcvbufErrors: 44017                  --- this

 [snip]

  net.core.rmem_max = 1048575
  net.core.wmem_max = 1048575
  net.core.rmem_default = 1048575
  net.core.wmem_default = 1048575
  net.core.optmem_max = 1048575
  net.core.netdev_max_backlog = 1


 Additional question - Do I need to restart Asterisk for these settings
 to apply to SIP? I have not done so yet, and further reading suggests
 that this is a per-process buffer and may be assigned when the
 listener is created.


 I had to double check myself in the kernel sources (in
 net/core/sock.c:sock_init_data), but yes you will need to recreate the
 socket for the rmem_default option to take effect.

 I was just about to ask if changing the sysctl changed the frequency
 that you see the error counter increment when I saw your follow on.

 Cheers,
 Shaun

 Thanks for making the extra effort Shaun :)  I will restart Asterisk
 when I get a chance, and re-check.

 Regards,
 Steve


False alarm?

FYI, it seems that the issue is not at-all what we thought.

It transpires that the unit is an old Asterisk 1.2 system, and that
the DNS server that the unit had been pointed to had been taken out of
service. This caused lots of threads to back-up while attempting to
resolve things, and this dragged down the whole of asterisk, seemingly
to the point where it was not reading UDP data out of its buffers fast
enough!

Thanks,
Steve

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[asterisk-users] No MOH with parked call

[Steve Davies]

Hi,

Has anybody else noticed that MOH does not play on parked calls in
1.6.2.14? Or is it just my setup? MOH seems to work in every other
respect (Call Held or in-Queue), but when a call is parked, the logs
show MOH being started, but the parked party hears nothing.

The verbose logs show the following. Any thoughts on whet to check next?

Thanks,
Steve


### Call comes in here and is answered
-- SIP/snom360-0d6f answered DAHDI/2-1
-- Executing [...@macro-set-moh-call:1]
GotoIf(SIP/snom360-0d6f, 0?done) in new stack
-- Executing [...@macro-set-moh-call:2] Set(SIP/snom360-0d6f,
CHANNEL(musicclass)=m-default) in new stack
-- Executing [...@macro-set-moh-call:3] NoOp(SIP/snom360-0d6f,
) in new stack

### Here the call is being blind transferred to the Park number
-- Started music on hold, class 'default', on DAHDI/2-1
-- Stopped music on hold on DAHDI/2-1
  == Spawn extension (local, 210, 1) exited non-zero on 'DAHDI/2-1'
-- Executing [...@local:1] ForkCDR(DAHDI/2-1, ) in new stack
-- Executing [...@local:2] Set(DAHDI/2-1, CDR(userfield)=) in
new stack

### Not sure why I send Ringing here, but I tried NoOP() and
Answer() too just in case
-- Executing [...@local:3] Ringing(DAHDI/2-1, ) in new stack
-- Executing [...@local:4] Set(DAHDI/2-1,
CHANNEL(musicclass)=default) in new stack
-- Executing [...@local:5] Set(DAHDI/2-1,
CHANNEL(parkinglot)=default) in new stack
-- Executing [...@local:6] Goto(DAHDI/2-1,
parkedcalls_default,park,1) in new stack
-- Goto (parkedcalls_default,park,1)
-- Executing [p...@parkedcalls_default:1] Park(DAHDI/2-1, ) in new stack
  == Parked DAHDI/2-1 on 211 (lot default). Will timeout back to
extension [parkedcalls_default] s, 1 in 90 seconds
-- Added extension '211' priority 1 to parkedcalls_default (0xbe2e528)

# The 211 announcement is heard perfectly
-- DAHDI/2-1 Playing 'digits/2.alaw' (language 'en')
  == Extension Changed 211[extensions] new state InUse for Notify User steve
-- DAHDI/2-1 Playing 'digits/1.alaw' (language 'en')
-- DAHDI/2-1 Playing 'digits/1.alaw' (language 'en')

# The system claims to start MOH default which works elsewhere, but
the caller gets silence
-- Started music on hold, class 'default', on DAHDI/2-1
  == Spawn extension (parkedcalls_default, s, 1) exited non-zero on
'Parked/DAHDI/2-1ZOMBIE'

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Re: [asterisk-users] No MOH with parked call

[Steve Davies]

On 7 December 2010 14:17, Lee Archer lee.arc...@thebigword.com wrote:
 Hi, try unloading res_timing_dahdi.so then trying again.

 Lee

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
 Davies
 Sent: 07 December 2010 12:54
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] No MOH with parked call

 Hi,

 Has anybody else noticed that MOH does not play on parked calls in
 1.6.2.14? Or is it just my setup? MOH seems to work in every other
 respect (Call Held or in-Queue), but when a call is parked, the logs
 show MOH being started, but the parked party hears nothing.

 The verbose logs show the following. Any thoughts on whet to check next?

 Thanks,
 Steve


 ### Call comes in here and is answered
    -- SIP/snom360-0d6f answered DAHDI/2-1
    -- Executing [...@macro-set-moh-call:1] GotoIf(SIP/snom360-0d6f,
 0?done) in new stack
    -- Executing [...@macro-set-moh-call:2] Set(SIP/snom360-0d6f,
 CHANNEL(musicclass)=m-default) in new stack
    -- Executing [...@macro-set-moh-call:3] NoOp(SIP/snom360-0d6f,
 ) in new stack

 ### Here the call is being blind transferred to the Park number
    -- Started music on hold, class 'default', on DAHDI/2-1
    -- Stopped music on hold on DAHDI/2-1
  == Spawn extension (local, 210, 1) exited non-zero on 'DAHDI/2-1'
    -- Executing [...@local:1] ForkCDR(DAHDI/2-1, ) in new stack
    -- Executing [...@local:2] Set(DAHDI/2-1, CDR(userfield)=) in
 new stack

 ### Not sure why I send Ringing here, but I tried NoOP() and
 Answer() too just in case
    -- Executing [...@local:3] Ringing(DAHDI/2-1, ) in new stack
    -- Executing [...@local:4] Set(DAHDI/2-1,
 CHANNEL(musicclass)=default) in new stack
    -- Executing [...@local:5] Set(DAHDI/2-1,
 CHANNEL(parkinglot)=default) in new stack
    -- Executing [...@local:6] Goto(DAHDI/2-1,
 parkedcalls_default,park,1) in new stack
    -- Goto (parkedcalls_default,park,1)
    -- Executing [p...@parkedcalls_default:1] Park(DAHDI/2-1, ) in
 new stack
  == Parked DAHDI/2-1 on 211 (lot default). Will timeout back to
 extension [parkedcalls_default] s, 1 in 90 seconds
    -- Added extension '211' priority 1 to parkedcalls_default
 (0xbe2e528)

 # The 211 announcement is heard perfectly
    -- DAHDI/2-1 Playing 'digits/2.alaw' (language 'en')
  == Extension Changed 211[extensions] new state InUse for Notify User
 steve
    -- DAHDI/2-1 Playing 'digits/1.alaw' (language 'en')
    -- DAHDI/2-1 Playing 'digits/1.alaw' (language 'en')

 # The system claims to start MOH default which works elsewhere, but
 the caller gets silence
    -- Started music on hold, class 'default', on DAHDI/2-1
  == Spawn extension (parkedcalls_default, s, 1) exited non-zero on
 'Parked/DAHDI/2-1ZOMBIE'


Unloading res_timing_dahdi.so worked to fix MOH for Parked calls, but
it has killed call quality on ISDN calls - I think it interferes with
the software echo canceller somehow.

Is there a ticket open on this? A patch to try?

Thanks,
Steve

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Re: [asterisk-users] No MOH with parked call

[Steve Davies]

On 7 December 2010 15:00, Steve Davies davies...@gmail.com wrote:
 On 7 December 2010 14:17, Lee Archer lee.arc...@thebigword.com wrote:
 Hi, try unloading res_timing_dahdi.so then trying again.

 Lee

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
 Davies
 Sent: 07 December 2010 12:54
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] No MOH with parked call

 Hi,

 Has anybody else noticed that MOH does not play on parked calls in
 1.6.2.14? Or is it just my setup? MOH seems to work in every other
 respect (Call Held or in-Queue), but when a call is parked, the logs
 show MOH being started, but the parked party hears nothing.

 The verbose logs show the following. Any thoughts on whet to check next?

 Thanks,
 Steve


 ### Call comes in here and is answered
    -- SIP/snom360-0d6f answered DAHDI/2-1
    -- Executing [...@macro-set-moh-call:1] GotoIf(SIP/snom360-0d6f,
 0?done) in new stack
    -- Executing [...@macro-set-moh-call:2] Set(SIP/snom360-0d6f,
 CHANNEL(musicclass)=m-default) in new stack
    -- Executing [...@macro-set-moh-call:3] NoOp(SIP/snom360-0d6f,
 ) in new stack

 ### Here the call is being blind transferred to the Park number
    -- Started music on hold, class 'default', on DAHDI/2-1
    -- Stopped music on hold on DAHDI/2-1
  == Spawn extension (local, 210, 1) exited non-zero on 'DAHDI/2-1'
    -- Executing [...@local:1] ForkCDR(DAHDI/2-1, ) in new stack
    -- Executing [...@local:2] Set(DAHDI/2-1, CDR(userfield)=) in
 new stack

 ### Not sure why I send Ringing here, but I tried NoOP() and
 Answer() too just in case
    -- Executing [...@local:3] Ringing(DAHDI/2-1, ) in new stack
    -- Executing [...@local:4] Set(DAHDI/2-1,
 CHANNEL(musicclass)=default) in new stack
    -- Executing [...@local:5] Set(DAHDI/2-1,
 CHANNEL(parkinglot)=default) in new stack
    -- Executing [...@local:6] Goto(DAHDI/2-1,
 parkedcalls_default,park,1) in new stack
    -- Goto (parkedcalls_default,park,1)
    -- Executing [p...@parkedcalls_default:1] Park(DAHDI/2-1, ) in
 new stack
  == Parked DAHDI/2-1 on 211 (lot default). Will timeout back to
 extension [parkedcalls_default] s, 1 in 90 seconds
    -- Added extension '211' priority 1 to parkedcalls_default
 (0xbe2e528)

 # The 211 announcement is heard perfectly
    -- DAHDI/2-1 Playing 'digits/2.alaw' (language 'en')
  == Extension Changed 211[extensions] new state InUse for Notify User
 steve
    -- DAHDI/2-1 Playing 'digits/1.alaw' (language 'en')
    -- DAHDI/2-1 Playing 'digits/1.alaw' (language 'en')

 # The system claims to start MOH default which works elsewhere, but
 the caller gets silence
    -- Started music on hold, class 'default', on DAHDI/2-1
  == Spawn extension (parkedcalls_default, s, 1) exited non-zero on
 'Parked/DAHDI/2-1ZOMBIE'


 Unloading res_timing_dahdi.so worked to fix MOH for Parked calls, but
 it has killed call quality on ISDN calls - I think it interferes with
 the software echo canceller somehow.

 Is there a ticket open on this? A patch to try?

 Thanks,
 Steve

For anyone searching/finding this issue, the patch here:

  https://issues.asterisk.org/view.php?id=17726

Applies to 1.6.2 with only a trivial tweak, and with minimal testing
is working here. We now get music on hold when a call is parked, even
when we are using res_timing_dahdi.so - Call quality remains high
under these circumstances too.

Thanks for the pointers.

Regards,
Steve

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Re: [asterisk-users] Avoided deadlock Error(solved)

[Steve Davies]

On 25 November 2010 13:02,  bayardo.sanc...@gmail.com wrote:
 The proble is dialplan I configure fine
 --
 Sent from my BlackBerry®
 VoIP, Windows/Linux Administration and Network Management
 US Numbers: 561-886-0664
 Nicaragua Mobile: +505.8488.6876

 -Original Message-
 From: Stefan Schmidt s...@sil.at
 Sender: asterisk-users-boun...@lists.digium.com
 Date: Wed, 24 Nov 2010 22:59:56
 To: Asterisk Users Mailing List - Non-Commercial 
 Discussionasterisk-users@lists.digium.com
 Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
        asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] Avoided deadlock Error

 Am 24.11.2010 13:48, schrieb Bayardo Sanchez:
 My Asterisk is Asterisk 1.2.30.2 currently running on viciserver the problem
 is :

 Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided
 deadlock for '0x861f6d8', 9 retries!
 Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided
 deadlock for '0x85a6420', 9 retries!
 Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided
 deadlock for '0x85bc510', 9 retries!
 Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided
 deadlock for '0x85f9e68', 9 retries!
 Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided
 deadlock for '0x85e1db0', 9 retries!

 this error comes only when I call spain saturated my CLI with the message
 error


 hello,

 as tilghman noticed 1.2 is EOL, but i still use it too and i see a bunch
 of this messages on different servers and they dont cause any problem at
 all.

 if you have some problems with this (except the warning message) you
 should upgrade.


AFAIK, this message is still there in 1.6, but has been downgraded to
a DEBUG message, so it is no longer visible most of the time.

Regards,
Steve

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[asterisk-users] Contradiction in GROUP() function

[Steve Davies]

I am confused. In Asterisk 1.2 and 1.4, in the code there is an error:
Setting a group requires an argument (group name)

But the syntax is shown as: Syntax: GROUP([category])

The [category] square brackets indicate to me an optional parameter,
which contradicts the error.

Verison 1.6 is non-committal in its definition, but I always assumed
that an empty string was still a valid category-name, so GROUP()=123
is as valid as GROUP(X)=123.

Could this be clarified?

Many thanks,
Steve

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Re: [asterisk-users] Contradiction in GROUP() function

[Steve Davies]

On 24 November 2010 10:12, Steve Davies davies...@gmail.com wrote:
 I am confused. In Asterisk 1.2 and 1.4, in the code there is an error:
 Setting a group requires an argument (group name)

 But the syntax is shown as: Syntax: GROUP([category])

 The [category] square brackets indicate to me an optional parameter,
 which contradicts the error.

 Verison 1.6 is non-committal in its definition, but I always assumed
 that an empty string was still a valid category-name, so GROUP()=123
 is as valid as GROUP(X)=123.

 Could this be clarified?

I suspect from further reading the code that this might just be a
misleading error message.

Regards
Steve

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Re: [asterisk-users] Asterisk parking question

[Steve Davies]

On 18 November 2010 17:43, Mike l...@net-wall.com wrote:
 I tried thator I think I did something similar, but that may or may not 
 apply (depending on my understanding of parking lots)


 Here is my relevant contexts.  The SIP phones are registered under this 
 context:


 [some_context]
 include = parkinglotA
 include = outboundcalls

 exten = t,1,Verbose(1|parking timeout!!!)

 Here, in features.conf, here is parkinglotA's definition

 [parkinglotA]
 context = parkinglotA
 parkpos = 101-120
 findslot = next
 parkingtime=60


 The thing is, I never hit the Verbose command.

 So my questions:
 1) Why won't this work?
 And more importantly:
 2) what's this park-dial context you speak of ?  Is this a hardcoded context 
 calls go back to?  Can I set one per parkilots (remember: I use multiple 
 parking lots)


Your call is in the [parkinglotA] context, but you are adding the 't'
to your [some_context] context, perhaps the following will work. I
have not tried it:


[some_context]
include = parkinglotA
include = outboundcalls

[parkinglotA]
exten = t,1,Verbose(1|parking timeout!!!)


Regards,
Steve

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Re: [asterisk-users] Licensing of Default MOH

[Steve Davies]

On 22 October 2010 14:24, Miguel Molina mmol...@millenium.com.co wrote:
 I think the OP is asking for the old MoH sound (fpm-world-mix, etc) that
 came with asterisk. I wonder why the change from the fpm sounds to the
 opsound ones, it was a licensing issue?


I think the original 'fpm' files were not as freely licenced as
originally thought, so there was a need to change. The details and
discussion will be in the mailing list archives, so I won't bother
repeating it here.

The licences mentioned earlier in this thread all relate to the newer MoH files.

Regards,
Steve

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[asterisk-users] SIP authentication - Thoughts please

[Steve Davies]

Hi,

We have a scenario where we need multiple discrete SIP trunks (peers)
from/to a single endpoint. Because the authentication system starts by
matching IP address, it only ever matches on one of the SIP peer
entries, and ignores the others. This is documented behaviour and as
such is correct. I would like to propose an extension to how SIP
peers are authenticated in this case:

1) Initial INVITE arrives, scan the list of all matching peer IP addresses.
   If a peer with no password is found, proceed with that peer immediately.

2) ...otherwise, a password is required, so send 407 challenge

3) INVITE arrives with Basic-Auth.
   Scan for /all/ matching peers based on IP address. If one of the
matching peers has a section name matching the Basic-Auth username,
use it to proceed.

4) I am not sure whether it is worth dropping through and testing auth
against other peers if there is no username match. Can auth ever
succeed under those circumstances (password matches, but not
username?)

Thanks for any feedback.

Regards,
Steve

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Re: [asterisk-users] SIP authentication - Thoughts please

[Steve Davies]

On 7 October 2010 10:10, Stefan Schmidt s...@sil.at wrote:
 Am 07.10.10 10:52, schrieb Steve Davies:
 Hi,

 snipped

 Hello,

 i just want to say something about point 4 which comes to my mind about
 security.


 4) I am not sure whether it is worth dropping through and testing auth
 against other peers if there is no username match. Can auth ever
 succeed under those circumstances (password matches, but not
 username?)

 If you use UDP its very easy to fake the source ip of a call so do you
 really want to open a door to an attacker by authenticate only by ip and
 passwort which can match to any peer with the same ip adress? To
 bruteforce this would be much easier than to bruteforce against sending
 IP, right username and right password.

I was not clear. By option 4) I intended that you test the password
against other peers with a matching IP address. I am not sure whether
the username is included in the SIP password hash, so do not know
whether there is even any point in doing so. As far as I can tell, in
the EXISTING sip stack, digest username is not used to determine which
peer to authenticate with, it just uses the first peer with a matching
IP.

 Have you tried to use different ports to register? i think this could help.

AFAIK, Asterisk will only operate on one port, and the remote end is a
major ITSP who will not be wanting to listen to me making odd requests
:)

Thanks for the feedback!

Regards,
Steve

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Re: [asterisk-users] High volume BLF - Suggestions?

[Steve Davies]

On 13 September 2010 19:12, Cassius Smith cass...@cassius.org wrote:
 Steve
 I have 64 channels being monitored with an SPA962 with two SPA932
 sidecars. It works perfectly with Asterisk 1.6.2.9; my users are very
 happy with this. Latest firmware is a must.

 HTH
 Cassius Smith


Any chance you could send me the output of
http://1.2.3.4/admin/spacfg.xml from that phone? Of course, remember
to remove any passwords :)

I would like to compare it to mine in case I am doing something stupid!

Thanks,
Steve

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Re: [asterisk-users] Queue/Dial Recording - Capture answering channel name.

[Steve Davies]

On 11 September 2010 20:36, Antonio Berrios
anto...@sheffieldcitytaxis.com wrote:
 On 09/09/10 17:59, Steve Davies wrote:
 On 9 September 2010 17:52, Antonio Berrios
 anto...@sheffieldcitytaxis.com  wrote:

 Steve Davies wrote:

 Hi,

 I am using 1.6.2.11, and I need to be able to include the name of the
 channel that answered a call in the call-recording filename.

 At a guess we need to use the Queue(name,,macro) or
 Dial(chan1chan2,,M(macro)) and use the macro to update the call
 recording filename. But, the macro runs on the calling channel, and I
 need the called channel - Is this accessible?

 Thanks,
 Steve

 Where ever the MixMonitor recording is done add in the  ${CHANNEL}
 variable to the filename parameter. Or even add in the line below to the
 context that contains Dial(QueueName).

 For example:

 exten =
 s,n,MixMonitor(*${CHANNEL}-${TIMESTAMP}-${UNIQUEID}-${CALLERID(num)}.wav49)



[snip]

 Why is it you require the answering channel in the recording filename?
 There may be an easier way to get you what you need. And a quick copy
 pasta of the dial plan you're using could be handy.


I will look at putting together an example of the dialplan.

We need the answering channel so the we can identify which handset was
recorded (which agent took the call). At present we have caller-id and
dialled extension, but an extension might be ringing any one of 10
phones, and the recording needs to be associated with that
handset/agent.

Regards,
Steve

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[asterisk-users] High volume BLF - Suggestions?

[Steve Davies]

Hi,

We have a user who is putting large call volumes through Asterisk, and
wants to BLF monitor up to 90 extensions. We are struggling to find a
handset that can keep up with Asterisk :)

1) Is there a handset that will do this?
2) Is there a different (standard) way to send BLF and allow directed pickups?
2a) Or even a handset specific way?

Asterisk handles the BLF volume fine, even on quite low-end hardware,
but we cannot find any handsets that can cope with it longer term.

Our test involves about 10 BLF-NOTIFY messages per second to each
handset with a 5-second pause every 5 seconds. This will either crash
or render unusable all of the following combinations:

snom360 + 1 x sidecar
Yealink T28 + 1 x sidecar
Yealink T28 + 2 x sidecar
Cisco SPA504g + 1 x sidecar
Cisco SPA504g + 2 x sidecar
Cisco SPA525g + 1 x sidecar (reboots often)
Cisco SPA525g + 2 x sidecar (reboots quickly)
Aastra 55i + non-LCD sidecar

Did not try Polycom as they do not do directed pickup and only small sidecars.
Linksys SPA962 with one sidecar is OK but is discontinued hardware.

Help?

Thanks,
Steve

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Re: [asterisk-users] Queue/Dial Recording - Capture answering channel name.

[Steve Davies]

On 13 September 2010 11:07, Antonio Berrios
anto...@sheffieldcitytaxis.com wrote:
 Gotcha. Yeah, I'm looking at implementing that (searching call
 recordings by agent that took the call) here but since our asterisk call
 recording is a separate server to the ones dealing with queues I'll be
 looking at tie-ing the agent to the call recording via a unique call ID
 in a database rather than in the filename itself. I'll post my
 findings/solution to the list if you would like?


Sounds like a plan :)

I may try to modify Asterisk to send more parameters to the
MONITOR_EXEC script if requested to do so. UniqueID, CDR(channel) and
CDR(dstchannel) seem like obvious choices.

I'll post results here if I manage it.

Regards,
Steve

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Re: [asterisk-users] High volume BLF - Suggestions?

[Steve Davies]

On 13 September 2010 11:43, Olivier oza_4...@yahoo.fr wrote:


 2010/9/13 Steve Davies davies...@gmail.com
[snip]
 Our test involves about 10 BLF-NOTIFY messages per second to each
 handset with a 5-second pause every 5 seconds. This will either crash
 or render unusable all of the following combinations:

 snom360 + 1 x sidecar

 As Snom phones have a parameter to express a time period during which BLF's
 SUBSCRIBE messages are spread and sent to Asterisk, I thought Snom phones
 would handle this load more easily.


[snip]

The SUBSCRIBE is handled fine, it is the NOTIFY messages that cause a problem.

Thanks,
Steve

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Re: [asterisk-users] High volume BLF - Suggestions?

[Steve Davies]

On 13 September 2010 12:16, Stefan Schmidt s...@sil.at wrote:
 Hello,

 Am 13.09.10 11:56, schrieb Steve Davies:
 Hi,

 We have a user who is putting large call volumes through Asterisk, and
 wants to BLF monitor up to 90 extensions. We are struggling to find a
 handset that can keep up with Asterisk :)

 1) Is there a handset that will do this?
 we only use the spa962 and spa525 for this, but didnt have any problems
 so far.

We find that the spa962 works pretty well with one sidecar (32
buttons) we only have one spa932 sidecar to hand though.

The spa525 seems to not like our sidecar configuration, and allows
only 59 subscriptions, so the last few lights do not work, and after a
while it decides to reboot regularly. Not quite sure how to debug that
one :(

The spa504 works pretty well with a sidecar, but crashed under load
over a weekend. I have updated 7.4.3 to 7.4.4 firmware and we're
trying again.

 2) Is there a different (standard) way to send BLF and allow directed 
 pickups?

 have a look at www.fop2.com maybe this could be interesting for you? its
 a web based solution to see the state of extensions and do a pickup or
 transfer calls.


Yes, we have a web-based BLF solution just like FOP, but some
end-users have very specific requirements *sigh* :)

Thanks for the input.

Cheers,
Steve

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Re: [asterisk-users] Queue/Dial Recording - Capture answering channel name.

[Steve Davies]

On 13 September 2010 16:58, Carlos Chavez cur...@telecomabmex.com wrote:
 On Mon, 2010-09-13 at 11:22 +0100, Steve Davies wrote:
 On 13 September 2010 11:07, Antonio Berrios
 anto...@sheffieldcitytaxis.com wrote:
  Gotcha. Yeah, I'm looking at implementing that (searching call
  recordings by agent that took the call) here but since our asterisk call
  recording is a separate server to the ones dealing with queues I'll be
  looking at tie-ing the agent to the call recording via a unique call ID
  in a database rather than in the filename itself. I'll post my
  findings/solution to the list if you would like?
 

 Sounds like a plan :)

 I may try to modify Asterisk to send more parameters to the
 MONITOR_EXEC script if requested to do so. UniqueID, CDR(channel) and
 CDR(dstchannel) seem like obvious choices.

 I'll post results here if I manage it.

        What we do is get the ${MEMBERINTERFACE} variable from the Queue and
 then rename the file in the h extension.  To get this variable you need
 to set setinterfacevar=yes in your queue definition.  We then run an
 AGI from the h extension to rename and move the file.


Excellent. thank you.
:)
Steve

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Re: [asterisk-users] Queue/Dial Recording - Capture answering channel name.

[Steve Davies]

On 9 September 2010 17:52, Antonio Berrios
anto...@sheffieldcitytaxis.com wrote:
 Steve Davies wrote:
 Hi,

 I am using 1.6.2.11, and I need to be able to include the name of the
 channel that answered a call in the call-recording filename.

 At a guess we need to use the Queue(name,,macro) or
 Dial(chan1chan2,,M(macro)) and use the macro to update the call
 recording filename. But, the macro runs on the calling channel, and I
 need the called channel - Is this accessible?

 Thanks,
 Steve

 Where ever the MixMonitor recording is done add in the  ${CHANNEL}
 variable to the filename parameter. Or even add in the line below to the
 context that contains Dial(QueueName).

 For example:

 exten =
 s,n,MixMonitor(*${CHANNEL}-${TIMESTAMP}-${UNIQUEID}-${CALLERID(num)}.wav49)


I was under the impression from the documentation that the ${CHANNEL}
variable held the _Calling_ channel, I need the _Called_ channel (the
channel that eventually picks up the call)

Am I mistaken? Perhaps I should give it a try :)

Thanks,
Steve

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[asterisk-users] Queue/Dial Recording - Capture answering channel name.

[Steve Davies]

Hi,

I am using 1.6.2.11, and I need to be able to include the name of the
channel that answered a call in the call-recording filename.

At a guess we need to use the Queue(name,,macro) or
Dial(chan1chan2,,M(macro)) and use the macro to update the call
recording filename. But, the macro runs on the calling channel, and I
need the called channel - Is this accessible?

Thanks,
Steve

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Re: [asterisk-users] How to debug this specific issue?

[Steve Davies]

On 25 August 2010 08:22, Matt Riddell li...@venturevoip.com wrote:
 On 25/08/10 7:20 PM, Tilghman Lesher wrote:
 I really thought that the canary should have sounded if Asterisk got in
 a loop - or maybe that only happens with high priority?

 The canary only runs in high priority mode, and it's only able to do anything
 if high priority scheduling is the culprit.  If it's something else, like
 memory swapping, there's nothing the canary can do to fix that.

 Aha, explains why I've never seen the canary die :D


...and yes, I was running in high priority mode - I thought I was
turning it off for testing, but looks like I left the setting in
asterisk.conf so leaving '-p' off the command line was making no
difference *sigh*

I believe the issue was actually with the devicestate thread. It was
trying to update state on a locked channel, and was trying to grab the
lock so regularly that it caused asterisk to grab lots of CPU cycles
(because of -p mode) The lock was not released because it was waiting
on a database write, which was being done by a lower priority external
process that was getting no time scheduled to it.

The database write is a local hack to record some extra call data - I
changed it to occur after the locks are released as I should have done
in the first place.

1.6.2.11 does not seem to have quite the same issue - It recovers
after the usual 200 lock attempts and gets on with life much more
happily. I cannot see any changes between 1.6.2.10 and 1.6.2.11 that
would have improved this behaviour.

Cheers,
Steve

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Re: [asterisk-users] How to debug this specific issue?

[Steve Davies]

On 24 August 2010 08:07, Stefan Schmidt s...@sil.at wrote:
 Steve Davies schrieb:
 On 23 August 2010 18:32, Stefan Schmidt s...@sil.at wrote:

 hello,

 have you allready tried strace ?
 you could just easily start asterisk with this command:

 strace asterisk -


 Yes, I tried this. Output just stops along with everything else and
 there are no clues.

 if you know in which function this happens you could also patch some
 ast_verbose rows into this function to see where it happens.

 another thing you could try is using gprof to see which functions waste
 much time. for this you need to compile asterisk like this:
 make ASTCFLAGS=-pg ASTLDFLAGS=-pg
 make install

 after this you will see a gmon.out file in the directory from where you
 have started asterisk (your home) and then you could use gprof with this
 gmon.out file.

 maybe you will find something with this.

I found that doing a build with debug symbols included and running
under gdb slowed down asterisk enough for me to get debug output.

Thanks for the pointers. I'll start a separate thread on working out
what the hell is going on :)

Cheers,
Steve

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Re: [asterisk-users] How to debug this specific issue?

[Steve Davies]

On 24 August 2010 14:34, Steve Davies davies...@gmail.com wrote:
 On 24 August 2010 08:07, Stefan Schmidt s...@sil.at wrote:
 Steve Davies schrieb:
 On 23 August 2010 18:32, Stefan Schmidt s...@sil.at wrote:

 hello,

 have you allready tried strace ?
 you could just easily start asterisk with this command:

 strace asterisk -


 Yes, I tried this. Output just stops along with everything else and
 there are no clues.

 if you know in which function this happens you could also patch some
 ast_verbose rows into this function to see where it happens.

 another thing you could try is using gprof to see which functions waste
 much time. for this you need to compile asterisk like this:
 make ASTCFLAGS=-pg ASTLDFLAGS=-pg
 make install

 after this you will see a gmon.out file in the directory from where you
 have started asterisk (your home) and then you could use gprof with this
 gmon.out file.

 maybe you will find something with this.

 I found that doing a build with debug symbols included and running
 under gdb slowed down asterisk enough for me to get debug output.

 Thanks for the pointers. I'll start a separate thread on working out
 what the hell is going on :)

Turns out that 1.6.2.11 fixes the symptom, and the cause is in my own
hack to the code. Holding locks too long == verybad.

Steve

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[asterisk-users] How to debug this specific issue?

[Steve Davies]

Hi,

I am happy with the usual GDB backtrace methods and so forth, but have
an issue that I cannot work out how to trace on 1.6.2.10.

If I use either the Bridge() app, or the manager Action: Bridge() in a
certain scenario (Basically to bridge 2 SIP channels, like an attended
transfer, resulting in 2 other SIP channels being discarded) then the
whole server locks solid. The console stops, the network stops,
something is hammering the box and nothing (including debug tools)
seem to be able to do anything about it.

If I 'nice' asterisk to lowest priority, and 'nice' a copy of 'top' to
highest priority, everything still locks. After a short period, the
box recovers, seemingly due to the 60 second RTP timer. Anything that
was being logged is lost.

My theory is the I am somehow causing a frame loop internal to
Asterisk by setting up some type of illegal bridge, but I used the
same code on 1.2 (I backported Bridge()) and it works just fine.

I need suggestions please on how to determine where it is locking, and why.

Many thanks,
Steve

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