Re: [asterisk-users] Asterisk 11 and H.323 trunk using OOH323 - is it stable?

[Pat Collins]

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vladimir
Mikhelson
Sent: Thursday, January 16, 2014 8:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 11 and H.323 trunk using OOH323 - is
it stable?


On 1/16/2014 6:57 PM, Dan Austin wrote:
> Patrick Lists wrote:
>> On 16-01-14 21:37, Gergely Kiss wrote:
>>> Dear List,
>>>
>>> I'm about to build an Asterisk 11.7 based PBX from scratch for our 
>>> company. I'm in the middle of the planning phase and it turned out 
>>> that our VoIP provider prefers H.323 protocol for handling voice 
>>> calls (while SIP is also supported as "plan B").
>> It's SIP everywhere and anyone who requires you, in 2014, to use 
>> H.323 should get a clue. Avoid them or at least demand SIP
> Bah.  There is nothing wrong with a working H.323 stack.  Just 
> assuming that they will have a working SIP stack because of the date 
> can lead to heartache.
>
>>> As I never worked with H.323 channels in Asterisk earlier, I'm not 
>>> sure if it's stable enough to be used in production.
>> No idea. Maybe someone else with H.323 experience will respond. AFAIK 
>> it's a dead-end.
> The ooh323 channel has been fairly reliable in our use case, which 
> involve connecting to a commercial IP PBX with crud SIP support.  Only 
> you can tell if it will work for you however, as sadly many times new 
> core features only get tested against the SIP channel(s), or worse 
> only implemented there as well.  Our current Asterisk version is 
> 11.5.1
>
> Dan
>
>
>
Sorry, have nothing to say of 11.5 but OOH323 works great in 1.8.  I use it
as an Avaya IP Office trunk.  No problems.

As you observed for yourself when you researched the topic there is not a
lot of help available, and Asterisk team prefers to make everybody think
that SIP is the only viable call setup protocol around.  They kind of not
talking a lot about their own IAX any more.

The official H.323 is abandoned.  OOH323 is being supported by a very
capable and responsive guy.  He does not frequent the user list as he
subscribes to the developer list, so I normally transfer the help inquiries
to him if there is no traction here.

-Vladimir

Hey Vladimir, can you share a bit about the ooH323 trunk to IPO
configuration that's stable?
I've tried a few different setups on both sides and wound up using a PRI to
do it.
A call from the IPO to Asterisk (1.6.2 at the time) would crash the Asterisk
box or not work at all.
I'd love to be able to offer this to my IPO and CM customers!
Thank you!
Pat...


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Re: [asterisk-users] Asterisk 11 and H.323 trunk using OOH323 - is it stable?

[Patrick Lists]

On 17-01-14 01:57, Dan Austin wrote:

Patrick Lists wrote:

On 16-01-14 21:37, Gergely Kiss wrote:

Dear List,

I'm about to build an Asterisk 11.7 based PBX from scratch for our
company. I'm in the middle of the planning phase and it turned out that
our VoIP provider prefers H.323 protocol for handling voice calls (while
SIP is also supported as "plan B").



It's SIP everywhere and anyone who requires you, in 2014, to use H.323
should get a clue. Avoid them or at least demand SIP

Bah.  There is nothing wrong with a working H.323 stack.  Just assuming
that they will have a working SIP stack because of the date can lead to
heartache.


By itself there is nothing wrong with a working H.323 stack. I just 
would not use it :-) Using H.323 for one provider while any backup or 
alternative providers probably use SIP results in needing two stacks in 
testing & production. It also requires the admins to gain knowledge of a 
legacy protocol. Maybe there are some incumbents or service providers 
with legacy H.323 equipment continuing to offer H.323 service. I get 
that. But for a business building a VoIP PBX from scratch H.323 does not 
make sense from a cost and operations point of view.



As I never worked with H.323 channels in Asterisk earlier, I'm not sure
if it's stable enough to be used in production.



No idea. Maybe someone else with H.323 experience will respond. AFAIK
it's a dead-end.

The ooh323 channel has been fairly reliable in our use case, which involve
connecting to a commercial IP PBX with crud SIP support.  Only you can tell
if it will work for you however, as sadly many times new core features only
get tested against the SIP channel(s), or worse only implemented there as
well.  Our current Asterisk version is 11.5.1


The OP mentioned that his VoIP provider prefers H.323 so it seems to be 
about trunking. IMHO "fairly reliable" is not something that is 
acceptable for trunking phone service.


H.323 is what Gopher is to HTTP/webservers. When was the last time you 
used a Gopher service? Would you today still buy Gopher based service 
because the service provider prefers it? :-)


Regards,
Patrick

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[asterisk-users] how can I get authenticate from my own server?

[Sean Darcy]

I'm  used to seeing fraudulent attempts to authenticate, But now I'm 
getting them from the server itself.


I have an asterisk server behind a firewalled router. The local subnet 
is 10.10.10.0/24, the server is 10.10.10.100.


Now I'm seeing in the log lots of:

Failed to authenticate device 
<*>@10.10.10.100:5060>;tag=9c565e6e


How can this happen?

sean


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Re: [asterisk-users] Asterisk 11 and H.323 trunk using OOH323 - is it stable?

[Vladimir Mikhelson]

On 1/16/2014 6:57 PM, Dan Austin wrote:
> Patrick Lists wrote:
>> On 16-01-14 21:37, Gergely Kiss wrote:
>>> Dear List,
>>>
>>> I'm about to build an Asterisk 11.7 based PBX from scratch for our
>>> company. I'm in the middle of the planning phase and it turned out that
>>> our VoIP provider prefers H.323 protocol for handling voice calls (while
>>> SIP is also supported as "plan B").
>> It's SIP everywhere and anyone who requires you, in 2014, to use H.323 
>> should get a clue. Avoid them or at least demand SIP
> Bah.  There is nothing wrong with a working H.323 stack.  Just assuming
> that they will have a working SIP stack because of the date can lead to
> heartache.  
>
>>> As I never worked with H.323 channels in Asterisk earlier, I'm not sure
>>> if it's stable enough to be used in production.
>> No idea. Maybe someone else with H.323 experience will respond. AFAIK 
>> it's a dead-end.
> The ooh323 channel has been fairly reliable in our use case, which involve
> connecting to a commercial IP PBX with crud SIP support.  Only you can tell
> if it will work for you however, as sadly many times new core features only
> get tested against the SIP channel(s), or worse only implemented there as
> well.  Our current Asterisk version is 11.5.1 
>
> Dan
>
>
>
Sorry, have nothing to say of 11.5 but OOH323 works great in 1.8.  I use
it as an Avaya IP Office trunk.  No problems.

As you observed for yourself when you researched the topic there is not
a lot of help available, and Asterisk team prefers to make everybody
think that SIP is the only viable call setup protocol around.  They kind
of not talking a lot about their own IAX any more.

The official H.323 is abandoned.  OOH323 is being supported by a very
capable and responsive guy.  He does not frequent the user list as he
subscribes to the developer list, so I normally transfer the help
inquiries to him if there is no traction here.

-Vladimir




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Re: [asterisk-users] Asterisk 11 and H.323 trunk using OOH323 - is it stable?

[Dan Austin]

Patrick Lists wrote:
> On 16-01-14 21:37, Gergely Kiss wrote:
>> Dear List,
>>
>> I'm about to build an Asterisk 11.7 based PBX from scratch for our
>> company. I'm in the middle of the planning phase and it turned out that
>> our VoIP provider prefers H.323 protocol for handling voice calls (while
>> SIP is also supported as "plan B").

> It's SIP everywhere and anyone who requires you, in 2014, to use H.323 
> should get a clue. Avoid them or at least demand SIP
Bah.  There is nothing wrong with a working H.323 stack.  Just assuming
that they will have a working SIP stack because of the date can lead to
heartache.  

>> As I never worked with H.323 channels in Asterisk earlier, I'm not sure
>> if it's stable enough to be used in production.

> No idea. Maybe someone else with H.323 experience will respond. AFAIK 
> it's a dead-end.
The ooh323 channel has been fairly reliable in our use case, which involve
connecting to a commercial IP PBX with crud SIP support.  Only you can tell
if it will work for you however, as sadly many times new core features only
get tested against the SIP channel(s), or worse only implemented there as
well.  Our current Asterisk version is 11.5.1 

Dan



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Re: [asterisk-users] Asterisk 11 and H.323 trunk using OOH323 - is it stable?

[Patrick Lists]

On 16-01-14 21:37, Gergely Kiss wrote:

Dear List,

I'm about to build an Asterisk 11.7 based PBX from scratch for our
company. I'm in the middle of the planning phase and it turned out that
our VoIP provider prefers H.323 protocol for handling voice calls (while
SIP is also supported as "plan B").


It's SIP everywhere and anyone who requires you, in 2014, to use H.323 
should get a clue. Avoid them or at least demand SIP.



As I never worked with H.323 channels in Asterisk earlier, I'm not sure
if it's stable enough to be used in production.


No idea. Maybe someone else with H.323 experience will respond. AFAIK 
it's a dead-end.


Regards,
Patrick

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Re: [asterisk-users] Asterisk 11 and H.323 trunk using OOH323 - is it stable?

[Paul Belanger]

On 14-01-16 03:37 PM, Gergely Kiss wrote:

Dear List,

I'm about to build an Asterisk 11.7 based PBX from scratch for our company.
I'm in the middle of the planning phase and it turned out that our VoIP
provider prefers H.323 protocol for handling voice calls (while SIP is also
supported as "plan B").

As I never worked with H.323 channels in Asterisk earlier, I'm not sure if
it's stable enough to be used in production.

Googling about the subject didn't help much, I could only find some old and
probably outdated information which I don't want to rely on.

Can you please confirm if the OOH323 module in Asterisk 11 is stable enough
to use for voice calls? No extra functionality is needed, just to be able
to create a H.323 trunk towards the provider and make and receive a maximum
of 30 simultaneous voice calls through the trunk.

Thanks for your kind response!

Save yourself time / energy and insist using SIP. If your ITSP cannot 
accommodate your request, thank them and look for another provider.


H323 is Asterisk is basically dead, sure there is a module, sure it 
might compile, but you'll be going down the path of zero help.


--
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Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
Github: https://github.com/pabelanger | Twitter: 
https://twitter.com/pabelanger


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Re: [asterisk-users] Asterisk ignoring nat settings

[Richard Mudgett]

On Thu, Jan 16, 2014 at 3:17 PM, Andres  wrote:

> On 1/16/14, 2:23 PM, Michael L. Young wrote:
>
>> - Original Message -
>>
>>  From: "Andres" 
>>> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>>> 
>>> Sent: Wednesday, January 15, 2014 7:51:28 PM
>>> Subject: Re: [asterisk-users] Asterisk ignoring nat settings
>>>
>>
>>
>>> Why don't you try with nat=yes. It should be equivalent to what you
>>> have but who knows. It might just work.
>>>
>> I am curious why you would say that "nat=yes" might work over
>> "nat=force_rport,comedia"?  As you stated, they are the same.  "nat=yes" is
>> deprecated and should be discouraged from being used.
>>
> I had no idea it was deprecated.  I have never seen such a warning in
> Asterisk 1.8.X
>

nat=yes is deprecated starting with Asterisk v11 as documented in the
CHANGES file.
Asterisk v11 also adds the comma separated list of nat options
(nat=force_rport,comedia).  Earlier versions do not support comma separated
nat
settings.

The OP did not specify the Asterisk version and may be using a syntax
inappropriate
to the Asterisk version.

Richard
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Re: [asterisk-users] Asterisk ignoring nat settings

[Michael L. Young]

- Original Message -
> From: "Andres" 
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
> 
> Sent: Thursday, January 16, 2014 4:17:53 PM
> Subject: Re: [asterisk-users] Asterisk ignoring nat settings
> 
> > I am curious why you would say that "nat=yes" might work over
> > "nat=force_rport,comedia"?  As you stated, they are the same.
> >  "nat=yes" is deprecated and should be discouraged from being
> > used.
> I had no idea it was deprecated.  I have never seen such a warning in
> Asterisk 1.8.X

The OP didn't specify which version of Asterisk he was using.  In Asterisk 1.8, 
"nat" was not a combinable list of options.  In Asterisk 11 it was.  So, I 
figured that since he was asking about "nat=force_rport,comedia" that he was on 
Asterisk 11 and in that version, "nat=yes" is deprecated.  I apologize about 
not clarifying the version that I was talking about.

Michael

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Re: [asterisk-users] Asterisk ignoring nat settings

[Andres]

On 1/16/14, 2:23 PM, Michael L. Young wrote:

- Original Message -


From: "Andres" 
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Sent: Wednesday, January 15, 2014 7:51:28 PM
Subject: Re: [asterisk-users] Asterisk ignoring nat settings
  

Why don't you try with nat=yes. It should be equivalent to what you
have but who knows. It might just work.

I am curious why you would say that "nat=yes" might work over "nat=force_rport,comedia"?  
As you stated, they are the same.  "nat=yes" is deprecated and should be discouraged from being 
used.
I had no idea it was deprecated.  I have never seen such a warning in 
Asterisk 1.8.X


Michael




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[asterisk-users] Asterisk 11 and H.323 trunk using OOH323 - is it stable?

[Gergely Kiss]

Dear List,

I'm about to build an Asterisk 11.7 based PBX from scratch for our company.
I'm in the middle of the planning phase and it turned out that our VoIP
provider prefers H.323 protocol for handling voice calls (while SIP is also
supported as "plan B").

As I never worked with H.323 channels in Asterisk earlier, I'm not sure if
it's stable enough to be used in production.

Googling about the subject didn't help much, I could only find some old and
probably outdated information which I don't want to rely on.

Can you please confirm if the OOH323 module in Asterisk 11 is stable enough
to use for voice calls? No extra functionality is needed, just to be able
to create a H.323 trunk towards the provider and make and receive a maximum
of 30 simultaneous voice calls through the trunk.

Thanks for your kind response!


Regards,
Gergely Kiss
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Re: [asterisk-users] Asterisk ignoring nat settings

[Michael L. Young]

- Original Message - 

> From: "Andres" 
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> 
> Sent: Wednesday, January 15, 2014 7:51:28 PM
> Subject: Re: [asterisk-users] Asterisk ignoring nat settings

 
> Why don't you try with nat=yes. It should be equivalent to what you
> have but who knows. It might just work.

I am curious why you would say that "nat=yes" might work over 
"nat=force_rport,comedia"?  As you stated, they are the same.  "nat=yes" is 
deprecated and should be discouraged from being used.

Michael

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Re: [asterisk-users] Starpy and Asterisk on different machines ? [SOLVED]

[Tzafrir Cohen]

On Thu, Jan 16, 2014 at 04:58:14PM +0100, Olivier wrote:
> Thanks for replying.
> 
> So as python-starpy requires asterisk in Debian Wheezy repo, for a Debian
> setup the alternatives are either :
> - to install it from source
> - tto build my own custom package removing this asterisk dependency (is it
> easy or even possible ?)

Should be simple.

> - to use another solution such as pyst.

- To "provide" Asterisk by a dummy package such as one built by equivs.

See, e.g. https://wiki.debian.org/CreateDummyPackage

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[asterisk-users] Cisco SPA504G, transfer asterisk page()

[Adam Moffett]

exten => 179,1,SIPAddHeader(Call-Info:\;answer-after=0)
exten => 179,2,Page(SIP/180&SIP/181&SIP/182&SIP/184)

The asterisk11 page() application works great, but I've just learned 
that the person who initiated the page can transfer or conference the 
page if they don't hang it up before using those functions.  It never 
would have occurred to me to try it, but a user did it accidentally 
today and it caused quite a stir when somebody's conversation with a 
caller was being broadcast from every phone.


They're using the conf and xfer buttons on the phone to make this 
happen, so I'm not sure if asterisk can even prevent them from doing it 
or if I have to figure out a way to stop it from happening on the 
phone.  The "i" option for Page didn't help.


Anybody dealt with this before?

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Re: [asterisk-users] Weird issue with Set(CALLERID(name)=string);

[Tiago Geada]

And we just figured that sound quality issues were not due to tcpdump ..
anyway sorry to troll this feed, and thank you for your sugestion



On 16 January 2014 16:57, Tiago Geada  wrote:

> Gareth,
>
> I had to disable the tcpdump process, has we were having sound quality
> issues.
>
> :-(
>
>
> On 16 January 2014 15:35, Gareth Blades 
> wrote:
>
>>  On 16/01/14 15:29, Kevin Larsen wrote:
>>
>> Not to derail the conversation, Gareth, but do you leave this running
>> full time on your asterisk boxes or just turn it on when you are trying to
>> track problems?
>>
>> On average, how far back can you go for looking at problems?
>>
>>
>> Its normally running full time so if someone reports a problem with a
>> call we can look at the logs and find out exactly what happened. We keep
>> asterisk verbose logs for 3 months, sip traces currently for about a month,
>> and uk-isup traces for a couple of weeks.
>>
>> Most carriers will do something similar. BT for example keep all of their
>> SS7 signalling for 48 hours.
>>
>> Regards
>> Gareth
>>
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>
>
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Re: [asterisk-users] Weird issue with Set(CALLERID(name)=string);

[Tiago Geada]

Gareth,

I had to disable the tcpdump process, has we were having sound quality
issues.

:-(


On 16 January 2014 15:35, Gareth Blades wrote:

>  On 16/01/14 15:29, Kevin Larsen wrote:
>
> Not to derail the conversation, Gareth, but do you leave this running full
> time on your asterisk boxes or just turn it on when you are trying to track
> problems?
>
> On average, how far back can you go for looking at problems?
>
>
> Its normally running full time so if someone reports a problem with a call
> we can look at the logs and find out exactly what happened. We keep
> asterisk verbose logs for 3 months, sip traces currently for about a month,
> and uk-isup traces for a couple of weeks.
>
> Most carriers will do something similar. BT for example keep all of their
> SS7 signalling for 48 hours.
>
> Regards
> Gareth
>
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Re: [asterisk-users] Starpy and Asterisk on different machines ? [SOLVED]

[A J Stiles]

On Thursday 16 January 2014, Olivier wrote:
> Thanks for replying.
> 
> So as python-starpy requires asterisk in Debian Wheezy repo, for a Debian
> setup the alternatives are either :
> - to install it from source
> - tto build my own custom package removing this asterisk dependency (is it
> easy or even possible ?)
> - to use another solution such as pyst.

Installing from Source Code might well be the simplest solution.


If you need to install something on several boxen, you can make your own .deb 
package -fairly- easily -- although it probably will have too many little 
technicalities to be accepted by the Debian project.

What I have done with homebrew .debs is have the package depend on `build-
essential` and the necessary `*-dev` .debs, install the Source Code files under 
/usr/src/, then do the build process in the postinst script.  This allows you 
to install the same .deb on 32 bit, 64 bit or Raspberry Pi architectures.


-- 
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Answers come *after* questions.

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Re: [asterisk-users] Starpy and Asterisk on different machines ? [SOLVED]

[Olivier]

Thanks for replying.

So as python-starpy requires asterisk in Debian Wheezy repo, for a Debian
setup the alternatives are either :
- to install it from source
- tto build my own custom package removing this asterisk dependency (is it
easy or even possible ?)
- to use another solution such as pyst.


Regards


2014/1/16 Adolphe Cher-Aime 

> Yes you can. This what starpy is for. It's build around Python twisted
> which allow you to write non blocked socket servers. You  can use starpy as
> a fastagi server.
> Both AMI and FASTAGI can be configured from a .conf file as follow:
>
> [AMI]
> username=ami_user
> secret=ami_pass
> server=asterisk_ami_ip
> port=ami_port
>
> [FastAGI]
> port=listen_port
> interface=listen_ip
>
>
> Hope that will help.
>
>
>
>
> On Thu, Jan 16, 2014 at 10:02 AM, Olivier  wrote:
>
>> Hello,
>>
>> Is it possible to run Starpy and Asterisk on different machines ?
>>
>> A quick glance at http://www.vrplumber.com/programming/starpy/ seems to
>> tell it is possible but Debian's python-starpy package installs Asterisk.
>>
>> What do you think ?
>>
>>
>> Regards
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>>http://www.asterisk.org/hello
>>
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>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
>
>
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Re: [asterisk-users] Weird issue with Set(CALLERID(name)=string);

[Gareth Blades]

On 16/01/14 15:29, Kevin Larsen wrote:
Not to derail the conversation, Gareth, but do you leave this running 
full time on your asterisk boxes or just turn it on when you are 
trying to track problems?


On average, how far back can you go for looking at problems?


Its normally running full time so if someone reports a problem with a 
call we can look at the logs and find out exactly what happened. We keep 
asterisk verbose logs for 3 months, sip traces currently for about a 
month, and uk-isup traces for a couple of weeks.


Most carriers will do something similar. BT for example keep all of 
their SS7 signalling for 48 hours.


Regards
Gareth
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Re: [asterisk-users] Weird issue with Set(CALLERID(name)=string);

[Tiago Geada]

Looking at his tcpdump command it keeps 500 files of 10 MB each? (not sure)


On 16 January 2014 15:29, Kevin Larsen wrote:

> asterisk-users-boun...@lists.digium.com wrote on 01/16/2014 08:55:31 AM:
>
> > From: Gareth Blades 
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > ,
> > Date: 01/16/2014 08:55 AM
> > Subject: Re: [asterisk-users] Weird issue with
> Set(CALLERID(name)=string);
> > Sent by: asterisk-users-boun...@lists.digium.com
> >
> > Very little as the amount of data being captured is quite small. We
> > have it running on our production servers which routinely handle a
> > couple of hundred concurrent calls.
> >
> > This is the script we use to start off the capture. It uses rolling
> > capture files so we will always have the last X number of capture
> > logs. It works very well and we have a custom system which enables
> > us to search for calls and request traces for them for when we have
> > to diagnose problems.
> >
> > #!/bin/bash
> > cd /var/lib/asterisk/siptraces
> > DATE=`date +%Y%m%d%H%M%S`
> > TRACEFILE=/var/lib/asterisk/siptraces/$DATE-siptrace.pcap
> > nohup /usr/sbin/tcpdump -p -i eth0 -s 0 port 5060 -w $TRACEFILE -C 10 -W
> 500 &
> >
> >
>
> Not to derail the conversation, Gareth, but do you leave this running full
> time on your asterisk boxes or just turn it on when you are trying to track
> problems?
>
> On average, how far back can you go for looking at problems?
> --
> _
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Re: [asterisk-users] Weird issue with Set(CALLERID(name)=string);

[Tiago Geada]

Hi,

I transfered the capture to my local machine and opened it in wireshark, I
can search from there:
--> SIP Display info:
"Sapo:0:243709253:1389884558.292163:SIP/covilha-pstn-000201f3"

but I will add your comment to my notes.


I've already searched the asterisk FULL log, and seen the Set() line ..
shows the correct string, that should have been displayed on softphone ...




On 16 January 2014 15:25, Gareth Blades wrote:

>  The SIP trace will give you an idea is perhaps something is becoming
> corrupted. If you keep a log of the asterisk console output (asterisk
> -rvvv) then you will see what it attempts to set the callerid to and any
> errors.
>
> Another tip. When you have a look at the sip trace you will see the
> call-id. If you make a note of this and run the following replacing the
> call-id and the trace file with the appropriate values it will display the
> sip trace in a very nice human readable format. tshark comes with the
> wireshark pakage and ngrep is part of epel repository if you are running
> centos.
>
> tshark -t ad -r '$tracefile' -R 'sip.Call-ID contains $callID' -w - |
> ngrep -I - -W byline -t
>
>
>
> On 16/01/14 14:57, Tiago Geada wrote:
>
>  Second thought, that would only allow me to know if there is a problem
> on asterisk or softphone.
>
>  Because the old callerid(name) that was presented on the softphone,
> belonged to a call made to a different peer, I doubt that it would be a
> softphone problem.
>
>  Our softphones are fixed with the same peer/extension .. if the wrong
> callerid were originally called to the same peer.. I guess that would be
> worth it..
>
>  even so, I will try and measure the impact on performance, however if
> asterisk did send the wrong string, how could I debug that??
>
>
> On 16 January 2014 14:27, Tiago Geada  wrote:
>
>>  You're right, seems like a nice way to debug. Regarding that, how would
>> the impact be affected running it on asterisk box? I guess only port 5060
>> is not too bad
>>
>>
>>  On 16 January 2014 14:09, Gareth Blades <
>> mailinglist+aster...@dns99.co.uk> wrote:
>>
>>>On 16/01/14 10:47, Tiago Geada wrote:
>>>
>>>  Hi folks,
>>>
>>>  We've been having a weird issue... It is happening more often in the
>>> last few months...
>>>
>>>  Most inbound calls, we have in our dialplan before Queue():
>>>
>>>
>>> Set(CALLERID(name)=${PARTNER}:0:${CALLERID(num)}:${UNIQUEID}:${CHANNEL})
>>> ;
>>>
>>>  So when the call rings a member, softphone will show this string 
>>>
>>>  The issue is that sometimes the string showing in the softphone is not
>>> the same. Its a string from a past call, in the latest case I've seen, from
>>> about 40 days ago!!
>>>
>>>  User took a screenshot, I've searched for that uniqueid showing in
>>> softphone in cdr, and that string was valid for a different call 40 days
>>> ago!!
>>>
>>>
>>>  I searched full log, and Set() sets the correct string... I can't
>>> figure why softphone shows a string from a past call !!
>>>
>>>  :(
>>>
>>>  Any hints ?
>>>
>>>
>>>   I would leave tcpdump running capturing port 5060 so you can load it
>>> onto wireshark and have a look at the sip headers. That will tell you if
>>> the SIP is incorrect or if its a problem with the client.
>>>
>>>  --
>>> _
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>>http://www.asterisk.org/hello
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>>
>
>
>
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Re: [asterisk-users] Weird issue with Set(CALLERID(name)=string);

[Kevin Larsen]

asterisk-users-boun...@lists.digium.com wrote on 01/16/2014 08:55:31 AM:

> From: Gareth Blades 
> To: Asterisk Users Mailing List - Non-Commercial Discussion 
> , 
> Date: 01/16/2014 08:55 AM
> Subject: Re: [asterisk-users] Weird issue with 
Set(CALLERID(name)=string);
> Sent by: asterisk-users-boun...@lists.digium.com
> 
> Very little as the amount of data being captured is quite small. We 
> have it running on our production servers which routinely handle a 
> couple of hundred concurrent calls.
> 
> This is the script we use to start off the capture. It uses rolling 
> capture files so we will always have the last X number of capture 
> logs. It works very well and we have a custom system which enables 
> us to search for calls and request traces for them for when we have 
> to diagnose problems.
> 
> #!/bin/bash
> cd /var/lib/asterisk/siptraces
> DATE=`date +%Y%m%d%H%M%S`
> TRACEFILE=/var/lib/asterisk/siptraces/$DATE-siptrace.pcap
> nohup /usr/sbin/tcpdump -p -i eth0 -s 0 port 5060 -w $TRACEFILE -C 10 -W 
500 &
> 
> 

Not to derail the conversation, Gareth, but do you leave this running full 
time on your asterisk boxes or just turn it on when you are trying to 
track problems?

On average, how far back can you go for looking at problems?-- 
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Re: [asterisk-users] Weird issue with Set(CALLERID(name)=string);

[Gareth Blades]

The SIP trace will give you an idea is perhaps something is becoming 
corrupted. If you keep a log of the asterisk console output (asterisk 
-rvvv) then you will see what it attempts to set the callerid to and any 
errors.


Another tip. When you have a look at the sip trace you will see the 
call-id. If you make a note of this and run the following replacing the 
call-id and the trace file with the appropriate values it will display 
the sip trace in a very nice human readable format. tshark comes with 
the wireshark pakage and ngrep is part of epel repository if you are 
running centos.


tshark -t ad -r '$tracefile' -R 'sip.Call-ID contains $callID' -w - | 
ngrep -I - -W byline -t



On 16/01/14 14:57, Tiago Geada wrote:
Second thought, that would only allow me to know if there is a problem 
on asterisk or softphone.


Because the old callerid(name) that was presented on the softphone, 
belonged to a call made to a different peer, I doubt that it would be 
a softphone problem.


Our softphones are fixed with the same peer/extension .. if the wrong 
callerid were originally called to the same peer.. I guess that would 
be worth it..


even so, I will try and measure the impact on performance, however if 
asterisk did send the wrong string, how could I debug that??



On 16 January 2014 14:27, Tiago Geada > wrote:


You're right, seems like a nice way to debug. Regarding that, how
would the impact be affected running it on asterisk box? I guess
only port 5060 is not too bad


On 16 January 2014 14:09, Gareth Blades
mailto:mailinglist+aster...@dns99.co.uk>> wrote:

On 16/01/14 10:47, Tiago Geada wrote:

Hi folks,

We've been having a weird issue... It is happening more often
in the last few months...

Most inbound calls, we have in our dialplan before Queue():


Set(CALLERID(name)=${PARTNER}:0:${CALLERID(num)}:${UNIQUEID}:${CHANNEL});

So when the call rings a member, softphone will show this
string 

The issue is that sometimes the string showing in the
softphone is not the same. Its a string from a past call, in
the latest case I've seen, from about 40 days ago!!

User took a screenshot, I've searched for that uniqueid
showing in softphone in cdr, and that string was valid for a
different call 40 days ago!!


I searched full log, and Set() sets the correct string... I
can't figure why softphone shows a string from a past call !!

:(

Any hints ?



I would leave tcpdump running capturing port 5060 so you can
load it onto wireshark and have a look at the sip headers.
That will tell you if the SIP is incorrect or if its a problem
with the client.

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Re: [asterisk-users] Starpy and Asterisk on different machines ?

[Adolphe Cher-Aime]

Yes you can. This what starpy is for. It's build around Python twisted
which allow you to write non blocked socket servers. You  can use starpy as
a fastagi server.
Both AMI and FASTAGI can be configured from a .conf file as follow:

[AMI]
username=ami_user
secret=ami_pass
server=asterisk_ami_ip
port=ami_port

[FastAGI]
port=listen_port
interface=listen_ip


Hope that will help.




On Thu, Jan 16, 2014 at 10:02 AM, Olivier  wrote:

> Hello,
>
> Is it possible to run Starpy and Asterisk on different machines ?
>
> A quick glance at http://www.vrplumber.com/programming/starpy/ seems to
> tell it is possible but Debian's python-starpy package installs Asterisk.
>
> What do you think ?
>
>
> Regards
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
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>http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] Weird issue with Set(CALLERID(name)=string);

[Tiago Geada]

Thank you Gareth

I will try that :)


On 16 January 2014 14:55, Gareth Blades wrote:

>  Very little as the amount of data being captured is quite small. We have
> it running on our production servers which routinely handle a couple of
> hundred concurrent calls.
>
> This is the script we use to start off the capture. It uses rolling
> capture files so we will always have the last X number of capture logs. It
> works very well and we have a custom system which enables us to search for
> calls and request traces for them for when we have to diagnose problems.
>
> #!/bin/bash
> cd /var/lib/asterisk/siptraces
> DATE=`date +%Y%m%d%H%M%S`
> TRACEFILE=/var/lib/asterisk/siptraces/$DATE-siptrace.pcap
> nohup /usr/sbin/tcpdump -p -i eth0 -s 0 port 5060 -w $TRACEFILE -C 10 -W
> 500 &
>
>
>
> On 16/01/14 14:27, Tiago Geada wrote:
>
>  You're right, seems like a nice way to debug. Regarding that, how would
> the impact be affected running it on asterisk box? I guess only port 5060
> is not too bad
>
>
> On 16 January 2014 14:09, Gareth Blades 
> wrote:
>
>>   On 16/01/14 10:47, Tiago Geada wrote:
>>
>>  Hi folks,
>>
>>  We've been having a weird issue... It is happening more often in the
>> last few months...
>>
>>  Most inbound calls, we have in our dialplan before Queue():
>>
>>  Set(CALLERID(name)=${PARTNER}:0:${CALLERID(num)}:${UNIQUEID}:${CHANNEL})
>> ;
>>
>>  So when the call rings a member, softphone will show this string 
>>
>>  The issue is that sometimes the string showing in the softphone is not
>> the same. Its a string from a past call, in the latest case I've seen, from
>> about 40 days ago!!
>>
>>  User took a screenshot, I've searched for that uniqueid showing in
>> softphone in cdr, and that string was valid for a different call 40 days
>> ago!!
>>
>>
>>  I searched full log, and Set() sets the correct string... I can't
>> figure why softphone shows a string from a past call !!
>>
>>  :(
>>
>>  Any hints ?
>>
>>
>>   I would leave tcpdump running capturing port 5060 so you can load it
>> onto wireshark and have a look at the sip headers. That will tell you if
>> the SIP is incorrect or if its a problem with the client.
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>http://www.asterisk.org/hello
>
> asterisk-users mailing list
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>
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[asterisk-users] Starpy and Asterisk on different machines ?

[Olivier]

Hello,

Is it possible to run Starpy and Asterisk on different machines ?

A quick glance at http://www.vrplumber.com/programming/starpy/ seems to
tell it is possible but Debian's python-starpy package installs Asterisk.

What do you think ?


Regards
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Re: [asterisk-users] Weird issue with Set(CALLERID(name)=string);

[Tiago Geada]

Second thought, that would only allow me to know if there is a problem on
asterisk or softphone.

Because the old callerid(name) that was presented on the softphone,
belonged to a call made to a different peer, I doubt that it would be a
softphone problem.

Our softphones are fixed with the same peer/extension .. if the wrong
callerid were originally called to the same peer.. I guess that would be
worth it..

even so, I will try and measure the impact on performance, however if
asterisk did send the wrong string, how could I debug that??


On 16 January 2014 14:27, Tiago Geada  wrote:

> You're right, seems like a nice way to debug. Regarding that, how would
> the impact be affected running it on asterisk box? I guess only port 5060
> is not too bad
>
>
> On 16 January 2014 14:09, Gareth Blades 
> wrote:
>
>>  On 16/01/14 10:47, Tiago Geada wrote:
>>
>>  Hi folks,
>>
>>  We've been having a weird issue... It is happening more often in the
>> last few months...
>>
>>  Most inbound calls, we have in our dialplan before Queue():
>>
>>  Set(CALLERID(name)=${PARTNER}:0:${CALLERID(num)}:${UNIQUEID}:${CHANNEL})
>> ;
>>
>>  So when the call rings a member, softphone will show this string 
>>
>>  The issue is that sometimes the string showing in the softphone is not
>> the same. Its a string from a past call, in the latest case I've seen, from
>> about 40 days ago!!
>>
>>  User took a screenshot, I've searched for that uniqueid showing in
>> softphone in cdr, and that string was valid for a different call 40 days
>> ago!!
>>
>>
>>  I searched full log, and Set() sets the correct string... I can't
>> figure why softphone shows a string from a past call !!
>>
>>  :(
>>
>>  Any hints ?
>>
>>
>>  I would leave tcpdump running capturing port 5060 so you can load it
>> onto wireshark and have a look at the sip headers. That will tell you if
>> the SIP is incorrect or if its a problem with the client.
>>
>> --
>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
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Re: [asterisk-users] Weird issue with Set(CALLERID(name)=string);

[Gareth Blades]

Very little as the amount of data being captured is quite small. We have 
it running on our production servers which routinely handle a couple of 
hundred concurrent calls.


This is the script we use to start off the capture. It uses rolling 
capture files so we will always have the last X number of capture logs. 
It works very well and we have a custom system which enables us to 
search for calls and request traces for them for when we have to 
diagnose problems.


#!/bin/bash
cd /var/lib/asterisk/siptraces
DATE=`date +%Y%m%d%H%M%S`
TRACEFILE=/var/lib/asterisk/siptraces/$DATE-siptrace.pcap
nohup /usr/sbin/tcpdump -p -i eth0 -s 0 port 5060 -w $TRACEFILE -C 10 -W 
500 &



On 16/01/14 14:27, Tiago Geada wrote:
You're right, seems like a nice way to debug. Regarding that, how 
would the impact be affected running it on asterisk box? I guess only 
port 5060 is not too bad



On 16 January 2014 14:09, Gareth Blades 
> wrote:


On 16/01/14 10:47, Tiago Geada wrote:

Hi folks,

We've been having a weird issue... It is happening more often in
the last few months...

Most inbound calls, we have in our dialplan before Queue():

Set(CALLERID(name)=${PARTNER}:0:${CALLERID(num)}:${UNIQUEID}:${CHANNEL});

So when the call rings a member, softphone will show this string 

The issue is that sometimes the string showing in the softphone
is not the same. Its a string from a past call, in the latest
case I've seen, from about 40 days ago!!

User took a screenshot, I've searched for that uniqueid showing
in softphone in cdr, and that string was valid for a different
call 40 days ago!!


I searched full log, and Set() sets the correct string... I can't
figure why softphone shows a string from a past call !!

:(

Any hints ?



I would leave tcpdump running capturing port 5060 so you can load
it onto wireshark and have a look at the sip headers. That will
tell you if the SIP is incorrect or if its a problem with the client.

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Re: [asterisk-users] Weird issue with Set(CALLERID(name)=string);

[Tiago Geada]

You're right, seems like a nice way to debug. Regarding that, how would the
impact be affected running it on asterisk box? I guess only port 5060 is
not too bad


On 16 January 2014 14:09, Gareth Blades wrote:

>  On 16/01/14 10:47, Tiago Geada wrote:
>
>  Hi folks,
>
>  We've been having a weird issue... It is happening more often in the
> last few months...
>
>  Most inbound calls, we have in our dialplan before Queue():
>
>  Set(CALLERID(name)=${PARTNER}:0:${CALLERID(num)}:${UNIQUEID}:${CHANNEL});
>
>  So when the call rings a member, softphone will show this string 
>
>  The issue is that sometimes the string showing in the softphone is not
> the same. Its a string from a past call, in the latest case I've seen, from
> about 40 days ago!!
>
>  User took a screenshot, I've searched for that uniqueid showing in
> softphone in cdr, and that string was valid for a different call 40 days
> ago!!
>
>
>  I searched full log, and Set() sets the correct string... I can't figure
> why softphone shows a string from a past call !!
>
>  :(
>
>  Any hints ?
>
>
>  I would leave tcpdump running capturing port 5060 so you can load it onto
> wireshark and have a look at the sip headers. That will tell you if the SIP
> is incorrect or if its a problem with the client.
>
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Re: [asterisk-users] Weird issue with Set(CALLERID(name)=string);

[Gareth Blades]

On 16/01/14 10:47, Tiago Geada wrote:

Hi folks,

We've been having a weird issue... It is happening more often in the 
last few months...


Most inbound calls, we have in our dialplan before Queue():

Set(CALLERID(name)=${PARTNER}:0:${CALLERID(num)}:${UNIQUEID}:${CHANNEL});

So when the call rings a member, softphone will show this string 

The issue is that sometimes the string showing in the softphone is not 
the same. Its a string from a past call, in the latest case I've seen, 
from about 40 days ago!!


User took a screenshot, I've searched for that uniqueid showing in 
softphone in cdr, and that string was valid for a different call 40 
days ago!!



I searched full log, and Set() sets the correct string... I can't 
figure why softphone shows a string from a past call !!


:(

Any hints ?


I would leave tcpdump running capturing port 5060 so you can load it 
onto wireshark and have a look at the sip headers. That will tell you if 
the SIP is incorrect or if its a problem with the client.
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Re: [asterisk-users] Solution to connect an audio system to MeetMe

[Darryl Moore]

Yup. That's what i do. The CLI version of linphone set to autoanswer, with
the audio jacks tied to our exernal sound system. Works well. The echo
cancellation in linphone helps a lot for speakerphones.
On Jan 16, 2014 7:51 AM, "Administrator TOOTAI"  wrote:

> Hi list,
>
> I have a customer which will organize a conference in a big meeting room
> which has a sound system. He would like to connect this sound system to a
> MeetMe room so participant in the MeetMe can act as if they where on site.
>
> My idea is to take a barbone or Notebook, connect it to the sound system
> using the soundcard and run a softphone on it.
>
> Does some of you already have success in such a setup? Which solution did
> you implement?
>
> Any ideas are welcome :-)
>
> --
> Daniel
>
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[asterisk-users] Solution to connect an audio system to MeetMe

[Administrator TOOTAI]

Hi list,

I have a customer which will organize a conference in a big meeting room 
which has a sound system. He would like to connect this sound system to 
a MeetMe room so participant in the MeetMe can act as if they where on site.


My idea is to take a barbone or Notebook, connect it to the sound system 
using the soundcard and run a softphone on it.


Does some of you already have success in such a setup? Which solution 
did you implement?


Any ideas are welcome :-)

--
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[asterisk-users] Weird issue with Set(CALLERID(name)=string);

[Tiago Geada]

Hi folks,

We've been having a weird issue... It is happening more often in the last
few months...

Most inbound calls, we have in our dialplan before Queue():

Set(CALLERID(name)=${PARTNER}:0:${CALLERID(num)}:${UNIQUEID}:${CHANNEL});

So when the call rings a member, softphone will show this string 

The issue is that sometimes the string showing in the softphone is not the
same. Its a string from a past call, in the latest case I've seen, from
about 40 days ago!!

User took a screenshot, I've searched for that uniqueid showing in
softphone in cdr, and that string was valid for a different call 40 days
ago!!


I searched full log, and Set() sets the correct string... I can't figure
why softphone shows a string from a past call !!

:(

Any hints ?
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[asterisk-users] Transfer call placed from console (with chan_alsa)

[Alex]

Hi everyone.

Having experimented a but with a prototype of a system I described in
an earlier thread (Reading DTMF sent by callee during a SIP call), I
decided to implement my requirement by transferring the call to
another extension. This way, the callee can open the door by pressing
#1, and the dial plan for extension 1 takes care of the rest.

This works when I make a typical SIP to SIP call, but it doesn't when
I call from the console, using chan_alsa. I can see that the transfer
feature is inactive:

rasterisk*CLI> core show channeltype console
-- Info about channel driver: Console --
  Device State: no
Indication: yes
 Transfer : no
  Capabilities: 0x40 (slin)
   Digit Begin: no
 Digit End: yes
Send HTML : no
 Image Support: no
  Text Support: yes



However, I am unable to find a way to activate it. How can I transfer
placed from the console? Is it possible, in principle?


Alex

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Re: [asterisk-users] Asterisk ignoring nat settings

[Leandro Dardini]

Yes, thank you. Maybe I have found the problem. The asterisk server is
behind a nat and the RTP port range was not redirected to the asterisk box,
so the Symmetric RTP cannot work because the asterisk is not receiving any
RTP packet from the remote phone.

Leandro


2014/1/16 Ishfaq Malik 

> Is directmedia set to no?
>
>
> On 15 January 2014 23:11, Leandro Dardini  wrote:
>
>> Hello,
>> I have an asterisk box with a peer configured with
>> nat=force_rport,comedia, but asterisk keeps sending the audio to the
>> private IP address and ignoring the client peer nat settings.
>>
>> If I check the "sip show peer extension", I see both symmetric RTP and
>> Force Rport are set to yes, but asterisk seems ignoring them.
>>
>>   Force rport  : Yes
>>   Symmetric RTP: Yes
>>
>> Asterisk is behind a nat the the externip and localnet has been
>> configured. The local net on the asterisk network is different from the
>> local net on phone.
>>
>> What else could I check?
>>
>> Leandro
>>
>>
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>
>
>
> --
>
> Ishfaq Malik
> Department: VOIP Support
> Company: Packnet Limited
> t: +44 (0)845 004 4994
> f: +44 (0)161 660 9825
> e: i...@pack-net.co.uk
> w: http://www.pack-net.co.uk
>
> Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
> 37 Ducie Street
> Manchester, M1 2JW
> COMPANY REG NO. 04920552
>
>
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Re: [asterisk-users] Asterisk ignoring nat settings

[Ishfaq Malik]

Is directmedia set to no?


On 15 January 2014 23:11, Leandro Dardini  wrote:

> Hello,
> I have an asterisk box with a peer configured with
> nat=force_rport,comedia, but asterisk keeps sending the audio to the
> private IP address and ignoring the client peer nat settings.
>
> If I check the "sip show peer extension", I see both symmetric RTP and
> Force Rport are set to yes, but asterisk seems ignoring them.
>
>   Force rport  : Yes
>   Symmetric RTP: Yes
>
> Asterisk is behind a nat the the externip and localnet has been
> configured. The local net on the asterisk network is different from the
> local net on phone.
>
> What else could I check?
>
> Leandro
>
>
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-- 

Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk

Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
37 Ducie Street
Manchester, M1 2JW
COMPANY REG NO. 04920552
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