Re: [asterisk-users] is DNS SRV enough for failover?

[CunningPike]

This is so wrong it's not funny. The caching of DNS SRV records acts in
its favor when it comes to failover - the UAs already have the
information they need in their resolver cache to perform the failover
without having to make another DNS query.

The TTL you need to worry about is that of the SIP registration - UAs
will typically renew their SIP registration at the half-life of the TTL.
A short SIP registration TTL will permit devices which are not actively
placing calls to failover more quickly than they otherwise would. Of
course, a balance must be struck between short SIP registration TTLs,
and the amount of SIP registration traffic this generates. YMMV.

CP

On Tue, 2008-09-30 at 22:07 -0400, Alex Balashov wrote:
> Nhadie wrote:
> > hi,
> > 
> > i'm using DNS SRV for failover, i tried to test shutting the server 
> > down, sip client should still register on the other server but it did 
> > not.  i'm using x-lite which i don't know if it's doing a srv query. 
> > does this mean SRV is not enough for failover? if a client has dns 
> > caching would this cause a problem?
> 
> SRV records are DNS.  DNS is cached.  Ergo, SRV records are cached. 
> Ergo, if they are cached excessively - either because the TTL is long, 
> or in defiance of the TTL - it can cause a problem.
> 
> No, DNS is not a good way to do real-time failover for anything.
> 


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Re: [asterisk-users] Improving the voice Quality,

[Steve Totaro]

BT3 (BackTrack) LiveCD is one of the best things out there, even has sipp
built right in, as well as other great apps, utilities, and security
"auditing".

I suggest everyone have a copy in their arsenal, and it is free of course.

Thanks,
Steve Totaro

On Fri, Oct 3, 2008 at 10:09 PM, Jai Rangi <[EMAIL PROTECTED]> wrote:

> All,
>
> Just an update on this. This turned out to be a bug in Cisco firewall. We
> ended up in upgrading the Firmware on the firewall.
>
> One thing I want to add, this was first time we used the fail over unit
> during peak time. In the whole process (failover, upgrade and failover back
> to active unit) was completely seamless. Did not had any down time, there
> was just a pause for just 1 second in the audio. I was very impressed.
>
> -Jai
>
>
>
> On Fri, Oct 3, 2008 at 1:21 PM, Jai Rangi <[EMAIL PROTECTED]> wrote:
>
>> Oh yes, how could I forgot about that?
>> Thank you,
>>
>> -Jai
>>
>>
>>
>> On Fri, Oct 3, 2008 at 1:08 PM, Alex Balashov <[EMAIL PROTECTED]>wrote:
>>
>>> sipp can simulate RTP traffic.
>>>
>>> Jai Rangi wrote:
>>>
>>> > Al and Alex,
>>> > Thank you for your input,
>>> > Sorry TDM is not the option at this time :( .
>>> > Everything has been great until last 2-3 days. Machine loads is not the
>>> > issue, we have multiple asterisk server to share the load. Not much
>>> > change in traffic.
>>> >
>>> > Now it been narrowed down to networking and we are trying to find out
>>> > where the issue is?  In our Firewall or SP's router. Does any one know
>>> > of any tool to simulate RTP traffic. Its pain to find out the bad calls
>>> > out of hundreds of calls.
>>> > BTW, What should be right value for tos in sip.conf.
>>> > We have
>>> > tos=0x68
>>> > Dont remember how did I come up with this value.
>>> >
>>> > I found this
>>> > http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+tos
>>> >
>>> > tos=0x10  low delay
>>> > tos=0x08  high throughput
>>> > tos=0x04  high reliability
>>> > tos=0x02  ECT bit set
>>> > tos=0x01  CE bit set
>>> >
>>> >
>>> > -Jai
>>> >
>>> >
>>> > On Fri, Oct 3, 2008 at 4:58 AM, Al Baker <[EMAIL PROTECTED]
>>> > > wrote:
>>> >
>>> > USE TDM Circuits - Voice Quality Good
>>> >
>>> > Alex Balashov wrote:
>>> >  > Jai Rangi wrote:
>>> >  >
>>> >  >
>>> >  >> All,
>>> >  >>
>>> >  >> I am having audio quality problem in some calls (1-2%) on
>>> > asterisk. I
>>> >  >> captured RTP traffic using ethereal and this is what I found
>>> > with the
>>> >  >> problematic calls. (Worst cases)
>>> >  >> Drop by Jitter buff: 25-75%
>>> >  >> Out of Seq: 50-100% (100 % means very very poor call quality).
>>> >  >>
>>> >  >> Has anyone had similar problem? If yes, can you please share
>>> your
>>> >  >> experience on how did you fix this?
>>> >  >>
>>> >  >
>>> >  > Such poor performance is not fixable.  The network, connectivity
>>> > issues,
>>> >  > machine load, etc. needs to be addressed - the underlying cause,
>>> in
>>> >  > other words.
>>> >  >
>>> >  > BTW, 100% out-of-sequence RTP packets leads to a lot more than
>>> just
>>> >  > "very very poor call quality."  I don't see how the conversation
>>> > could
>>> >  > even be coherent in that situation.
>>> >  >
>>> >  > What is more likely is that Wireshark's RTP stats are giving you
>>> some
>>> >  > distorted information.  I've found its stream analysis to be
>>> somewhat
>>> >  > buggy in that regard.
>>> >  >
>>> >  >
>>> >  >> I was wondering if I can decrease the MTU size to 250-500 on
>>> the
>>> > network
>>> >  >> card and use that card only for VoIP traffic. Will this have
>>> any bad
>>> >  >> effect on sip traffic/packets?
>>> >  >>
>>> >  >
>>> >  > No.  RTP packets are very small - much smaller than that MTU, or
>>> any
>>> >  > reasonable MTU you could set.
>>> >  >
>>> >  >
>>> >
>>> > ___
>>> > -- Bandwidth and Colocation Provided by http://www.api-digital.com--
>>> >
>>> > AstriCon 2008 - September 22 - 25 Phoenix, Arizona
>>> > Register Now: http://www.astricon.net
>>> >
>>> > 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 --
>>> >
>>> > AstriCon 2008 - September 22 - 25 Phoenix, Arizona
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>>>
>>>
>>> --
>>> Alex Balashov
>>> Evariste Systems
>>> Web: http://www.evaristesys.co

Re: [asterisk-users] Improving the voice Quality,

[Jai Rangi]

All,

Just an update on this. This turned out to be a bug in Cisco firewall. We
ended up in upgrading the Firmware on the firewall.

One thing I want to add, this was first time we used the fail over unit
during peak time. In the whole process (failover, upgrade and failover back
to active unit) was completely seamless. Did not had any down time, there
was just a pause for just 1 second in the audio. I was very impressed.

-Jai


On Fri, Oct 3, 2008 at 1:21 PM, Jai Rangi <[EMAIL PROTECTED]> wrote:

> Oh yes, how could I forgot about that?
> Thank you,
>
> -Jai
>
>
>
> On Fri, Oct 3, 2008 at 1:08 PM, Alex Balashov <[EMAIL PROTECTED]>wrote:
>
>> sipp can simulate RTP traffic.
>>
>> Jai Rangi wrote:
>>
>> > Al and Alex,
>> > Thank you for your input,
>> > Sorry TDM is not the option at this time :( .
>> > Everything has been great until last 2-3 days. Machine loads is not the
>> > issue, we have multiple asterisk server to share the load. Not much
>> > change in traffic.
>> >
>> > Now it been narrowed down to networking and we are trying to find out
>> > where the issue is?  In our Firewall or SP's router. Does any one know
>> > of any tool to simulate RTP traffic. Its pain to find out the bad calls
>> > out of hundreds of calls.
>> > BTW, What should be right value for tos in sip.conf.
>> > We have
>> > tos=0x68
>> > Dont remember how did I come up with this value.
>> >
>> > I found this
>> > http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+tos
>> >
>> > tos=0x10  low delay
>> > tos=0x08  high throughput
>> > tos=0x04  high reliability
>> > tos=0x02  ECT bit set
>> > tos=0x01  CE bit set
>> >
>> >
>> > -Jai
>> >
>> >
>> > On Fri, Oct 3, 2008 at 4:58 AM, Al Baker <[EMAIL PROTECTED]
>> > > wrote:
>> >
>> > USE TDM Circuits - Voice Quality Good
>> >
>> > Alex Balashov wrote:
>> >  > Jai Rangi wrote:
>> >  >
>> >  >
>> >  >> All,
>> >  >>
>> >  >> I am having audio quality problem in some calls (1-2%) on
>> > asterisk. I
>> >  >> captured RTP traffic using ethereal and this is what I found
>> > with the
>> >  >> problematic calls. (Worst cases)
>> >  >> Drop by Jitter buff: 25-75%
>> >  >> Out of Seq: 50-100% (100 % means very very poor call quality).
>> >  >>
>> >  >> Has anyone had similar problem? If yes, can you please share
>> your
>> >  >> experience on how did you fix this?
>> >  >>
>> >  >
>> >  > Such poor performance is not fixable.  The network, connectivity
>> > issues,
>> >  > machine load, etc. needs to be addressed - the underlying cause,
>> in
>> >  > other words.
>> >  >
>> >  > BTW, 100% out-of-sequence RTP packets leads to a lot more than
>> just
>> >  > "very very poor call quality."  I don't see how the conversation
>> > could
>> >  > even be coherent in that situation.
>> >  >
>> >  > What is more likely is that Wireshark's RTP stats are giving you
>> some
>> >  > distorted information.  I've found its stream analysis to be
>> somewhat
>> >  > buggy in that regard.
>> >  >
>> >  >
>> >  >> I was wondering if I can decrease the MTU size to 250-500 on the
>> > network
>> >  >> card and use that card only for VoIP traffic. Will this have any
>> bad
>> >  >> effect on sip traffic/packets?
>> >  >>
>> >  >
>> >  > No.  RTP packets are very small - much smaller than that MTU, or
>> any
>> >  > reasonable MTU you could set.
>> >  >
>> >  >
>> >
>> > ___
>> > -- Bandwidth and Colocation Provided by http://www.api-digital.com--
>> >
>> > AstriCon 2008 - September 22 - 25 Phoenix, Arizona
>> > Register Now: http://www.astricon.net
>> >
>> > asterisk-users mailing list
>> > To UNSUBSCRIBE or update options visit:
>> >   http://lists.digium.com/mailman/listinfo/asterisk-users
>> >
>> >
>> >
>> > 
>> >
>> > ___
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>> >
>> > AstriCon 2008 - September 22 - 25 Phoenix, Arizona
>> > Register Now: http://www.astricon.net
>> >
>> > asterisk-users mailing list
>> > To UNSUBSCRIBE or update options visit:
>> >http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
>> --
>> Alex Balashov
>> Evariste Systems
>> Web: http://www.evaristesys.com/
>> Tel: (+1) (678) 954-0670
>> Direct : (+1) (678) 954-0671
>> Mobile : (+1) (706) 338-8599
>>
>> ___
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
>> Register Now: http://www.astricon.net
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
_

Re: [asterisk-users] OT: Re: sip clients for smart phones?

[Philipp Kempgen]

Tariq .. schrieb:
> i'm using Hotmail webmail.. so what is wrong with it?  

See for yourself:
http://lists.digium.com/pipermail/asterisk-users/2008-October/219531.html
http://lists.digium.com/pipermail/asterisk-users/2008-October/219538.html
http://lists.digium.com/pipermail/asterisk-users/2008-October/219541.html

Ironically you would think that *my* email client is broken while
actually yours messes up the text/plain part.

And in addition to that
- You don't skip the irrelevant parts (e.g. my signature or the
  list footer)
- You top-post
- You violate email netiquette by not using a proper signature
  separator
- You send a footer telling me about Windows Live which is totally
  unrelated. Even for free email accounts that's not acceptable
  any longer since there are free accounts without advertising.

I could live with 1 or maybe 2 of these issues but 5 is a bit
much. You didn't even notice these problems, so, ok, sorry for
being rude. But for people who are used to email in ages it feels
like a punch in the face. It's a real culture clash.

   Philipp Kempgen

-- 
http://www.das-asterisk-buch.de  -  http://www.the-asterisk-book.com
Amooma GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
-- 

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Re: [asterisk-users] DTMF issues...

[Moises Silva]

Hey Carlos,

>What is the best method to debug DTMF issues?  Do I have to sniff the
> SIP packets?

The best method to debug DTMF issues depend on how you receive those
DTMF digits. Assuming you can use SIP INFO for the DTMF, that means
the DTMF digits are not really DTMF :-), that is, is not audio, with
SIP INFO the digits will be received by Asterisk as part of the SIP
signaling protocol and therefore you can easily spot them using "sip
debug peer myxpeer". If I were you I'd try that and see what I am
really receiving before drawing any other conclusion.

Moy

-- 
"I do not agree with what you have to say, but I'll defend to the
death your right to say it." Voltaire

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[asterisk-users] DTMF Talk Off

[Sam]

I am having some issues with dtmf beeping while in conversation.  This 
is a simple home setup using asterisk 1.4.21 and a grandsteam 286 ATA 
with a cordless phone.  I have played with the relaxdtmf setting in 
sip.conf.  Setting it to yes made it worse which makes me think that the 
issue is coming from asterisk and not the ata.  And all the dtmf modes 
are rfc.  Any one have any tips to further trouble shoot this?

Sam

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[asterisk-users] DTMF issues...

[Carlos Chavez]

I am having a big problem with DTMF.  I have a customer using an
Asterisk 1.4.20.1 system with ZTDUMMY as the timing source.  The problem
is that when they dial into a conference bridge or IVR where they have
to enter a code they always get an error.  Either some numbers are
duplicated or missing.

They use Teliax for calls to the USA and Protel in Mexico.  Both
carriers have the same problem so I think Asterisk could be at fault
here.  I have tried using dtmfmode as auto, inband, info and rfc2833 but
get the same result.  I have tried ulaw, alaw and g729 as the codec for
the calls but have the same problem.

Could this be a timing issue?  Since there are no zap channels we use
ZTDUMMY for timing.  The server usually runs only about 15 simultaneous
calls so the load is not heavy on the processor.

What is the best method to debug DTMF issues?  Do I have to sniff the
SIP packets?

-- 
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001


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[asterisk-users] Asterisk 1.6.1-beta1 Now Available

[Asterisk Development Team]

The Asterisk.org development team has released Asterisk 1.6.1-beta1.

While Asterisk 1.6.0 was in the beta and release candidate stage, the
development team was also working on merging new things for Asterisk
1.6.1.  Now that 1.6.0 has been released, the testing cycle for Asterisk
1.6.1 has begun.

To see the list of new features in Asterisk 1.6.1, see the CHANGES file:

http://svn.digium.com/view/asterisk/branches/1.6.1/CHANGES?view=markup

For a full list of changes that have gone into the development of
Asterisk 1.6.1 so far, see the ChangeLog:

http://svn.digium.com/view/asterisk/tags/1.6.1-beta1/ChangeLog?view=markup

To download Asterisk 1.6.1-beta1, visit the Digium downloads site:

http://downloads.digium.com/pub/telephony/asterisk/

Please report all issues to the issue tracker, http://bugs.digium.com/.

Thank you for your continued support of Asterisk!

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Re: [asterisk-users] network monitoring - triggering a phone call in asterisk

[Matt Gibson]

This may be what you're looking for:

http://www.linuxjournal.com/content/custom-checks-and-notifications-nagios


Thanks,
Matt G

: http://www.voipphreak.ca
: http://www.ratemydialplan.com
: http://www.asterisk-jobs.com


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex Balashov
Sent: Friday, October 03, 2008 4:13 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [asterisk-users] network monitoring - triggering a phone call
in asterisk

Use callfiles, monitor Manager for the appropriate connect or failure 
events, output those to a state file or report them via a domain socket 
service that a Nagios plugin can connect to and inquire.

Eric Fort wrote:

> I'd like to integrate asterisk into the notification portion of my 
> network monitoring where a call is generated and a message is played 
> when a host or service failed to be reachable (something where nagios 
> could trigger an event in asterisk would rock!)  How could this best be 
> accomplished?  Anyone know an existing solution?
> 
> Eric
> 
> 
> 
> 
> 
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-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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Re: [asterisk-users] sip clients for smart phones?

[Matt Gibson]

Ah, I don't use the touchflow crap :) 

On mine on the today screen (you'll have to go to settings, today, items) 

Set "internet telephony" to "on" and you should see it on the home screen. 

Thanks,
Matt G

: http://www.voipphreak.ca
: http://www.ratemydialplan.com
: http://www.asterisk-jobs.com


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton
Sent: Friday, October 03, 2008 4:03 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] sip clients for smart phones?

I don't even see it anywhere on TouchFlo3D. I don't see where to even use
it.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt Gibson
Sent: October 3, 2008 3:34 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] sip clients for smart phones?

I use the TytnII with Win Mob 6.1, customized ROM and it's working for me -
through the back speaker though. 


Thanks,
Matt G

: http://www.voipphreak.ca
: http://www.ratemydialplan.com
: http://www.asterisk-jobs.com


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton
Sent: Friday, October 03, 2008 2:57 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] sip clients for smart phones?

This doesn't work in WinMo6.1 for some reason. Especially on touchscreen
phones. Touch Diamond for instance.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt Gibson
Sent: October 3, 2008 11:52 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] sip clients for smart phones?

This may help:

http://www.voipphreak.ca/2008/03/29/enable-the-hidden-voip-features-of-windo
ws-mobile-6x-for-free-voip-calls-using-asterisk/

Note, that most sip clients for WINMOB suck and send the voice out the back
speaker instead of the front speaker. I've found one other client (can't
remember the name now) that works through the back speaker, but it's
payware, and nowhere near as lightweight as this method. For now I deal with
the back speaker. 

Thanks,
Matt G

: http://www.voipphreak.ca
: http://www.ratemydialplan.com
: http://www.asterisk-jobs.com

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gordon
Henderson
Sent: Friday, October 03, 2008 6:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] sip clients for smart phones?

On Fri, 3 Oct 2008, Babcock, Michael Alex wrote:

> hi;
> curious does anyone know of a smart phone client that could connect to
> asterisk?

All the Nokia ones which come with Wi-Fi and SIP can as far as I know. I 
use an E90 over Wi-Fi.

Phone gets hot and battery life is about an 1.5 hours of talk time, but...

Gordon

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Re: [asterisk-users] sip clients for smart phones?

[Babcock, Michael Alex]

that's ok i found it
On Oct 3, 2008, at 12:15 PM, Tariq .. wrote:


it is FRING i'm sorry for the mistype...
www.fring.com







AHD Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP
Syria: +963 944 618286
USA: +1 347 562 2308


From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Date: Fri, 3 Oct 2008 12:00:16 -0800
Subject: Re: [asterisk-users] sip clients for smart phones?

is it frig or fring?

On Oct 3, 2008, at 11:49 AM, Tariq .. wrote:

try using Frig.. it's a great client with an SIP client.. i tried it  
on IPhone and on my N82 Nokia phone.. it works great on GPRS and Wi- 
Fi... and i DO use it with my two Asterisk Servers..

regards




AHD Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP
Syria: +963 944 618286
USA: +1 347 562 2308


> From: [EMAIL PROTECTED]
> To: asterisk-users@lists.digium.com
> Date: Fri, 3 Oct 2008 15:33:52 -0400
> Subject: Re: [asterisk-users] sip clients for smart phones?
>
> I use the TytnII with Win Mob 6.1, customized ROM and it's working  
for me -

> through the back speaker though.
>
>
> Thanks,
> Matt G
>
> : http://www.voipphreak.ca
> : http://www.ratemydialplan.com
> : http://www.asterisk-jobs.com
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Mark  
Hamilton

> Sent: Friday, October 03, 2008 2:57 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: Re: [asterisk-users] sip clients for smart phones?
>
> This doesn't work in WinMo6.1 for some reason. Especially on  
touchscreen

> phones. Touch Diamond for instance.
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Matt  
Gibson

> Sent: October 3, 2008 11:52 AM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: Re: [asterisk-users] sip clients for smart phones?
>
> This may help:
>
> http://www.voipphreak.ca/2008/03/29/enable-the-hidden-voip-features-of-windo
> ws-mobile-6x-for-free-voip-calls-using-asterisk/
>
> Note, that most sip clients for WINMOB suck and send the voice out  
the back
> speaker instead of the front speaker. I've found one other client  
(can't

> remember the name now) that works through the back speaker, but it's
> payware, and nowhere near as lightweight as this method. For now I  
deal with

> the back speaker.
>
> Thanks,
> Matt G
>
> : http://www.voipphreak.ca
> : http://www.ratemydialplan.com
> : http://www.asterisk-jobs.com
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Gordon
> Henderson
> Sent: Friday, October 03, 2008 6:03 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] sip clients for smart phones?
>
> On Fri, 3 Oct 2008, Babcock, Michael Alex wrote:
>
> > hi;
> > curious does anyone know of a smart phone client that could  
connect to

> > asterisk?
>
> All the Nokia ones which come with Wi-Fi and SIP can as far as I  
know. I

> use an E90 over Wi-Fi.
>
> Phone gets hot and battery life is about an 1.5 hours of talk  
time, but...

>
> Gordon
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com  
--

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>
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thanks for reading
Systems administrator and own

Re: [asterisk-users] OT: Re: sip clients for smart phones?

[Tariq ..]

i'm using Hotmail webmail.. so what is wrong with it?  


AHD Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP
Syria: +963 944 618286
USA: +1 347 562 2308

> Date: Fri, 3 Oct 2008 22:17:02 +0200> From: [EMAIL PROTECTED]> To: 
> asterisk-users@lists.digium.com> Subject: [asterisk-users] OT: Re: sip 
> clients for smart phones?> > Tariq, fix your email client (it eats line 
> breaks in text/plain).> And please leave out the parts which are not relevant 
> (list> policy). Thanks.> > > Philipp Kempgen> > -- > 
> http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com> Amooma 
> GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de> Geschäftsführer: 
> Stefan Wintermeyer, Handelsregister: Neuwied B14998> -- > > 
> ___> -- Bandwidth and Colocation 
> Provided by http://www.api-digital.com --> > AstriCon 2008 - September 22 - 
> 25 Phoenix, Arizona> Register Now: http://www.astricon.net> > 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] Improving the voice Quality,

[Jai Rangi]

Oh yes, how could I forgot about that?
Thank you,

-Jai


On Fri, Oct 3, 2008 at 1:08 PM, Alex Balashov <[EMAIL PROTECTED]>wrote:

> sipp can simulate RTP traffic.
>
> Jai Rangi wrote:
>
> > Al and Alex,
> > Thank you for your input,
> > Sorry TDM is not the option at this time :( .
> > Everything has been great until last 2-3 days. Machine loads is not the
> > issue, we have multiple asterisk server to share the load. Not much
> > change in traffic.
> >
> > Now it been narrowed down to networking and we are trying to find out
> > where the issue is?  In our Firewall or SP's router. Does any one know
> > of any tool to simulate RTP traffic. Its pain to find out the bad calls
> > out of hundreds of calls.
> > BTW, What should be right value for tos in sip.conf.
> > We have
> > tos=0x68
> > Dont remember how did I come up with this value.
> >
> > I found this
> > http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+tos
> >
> > tos=0x10  low delay
> > tos=0x08  high throughput
> > tos=0x04  high reliability
> > tos=0x02  ECT bit set
> > tos=0x01  CE bit set
> >
> >
> > -Jai
> >
> >
> > On Fri, Oct 3, 2008 at 4:58 AM, Al Baker <[EMAIL PROTECTED]
> > > wrote:
> >
> > USE TDM Circuits - Voice Quality Good
> >
> > Alex Balashov wrote:
> >  > Jai Rangi wrote:
> >  >
> >  >
> >  >> All,
> >  >>
> >  >> I am having audio quality problem in some calls (1-2%) on
> > asterisk. I
> >  >> captured RTP traffic using ethereal and this is what I found
> > with the
> >  >> problematic calls. (Worst cases)
> >  >> Drop by Jitter buff: 25-75%
> >  >> Out of Seq: 50-100% (100 % means very very poor call quality).
> >  >>
> >  >> Has anyone had similar problem? If yes, can you please share your
> >  >> experience on how did you fix this?
> >  >>
> >  >
> >  > Such poor performance is not fixable.  The network, connectivity
> > issues,
> >  > machine load, etc. needs to be addressed - the underlying cause,
> in
> >  > other words.
> >  >
> >  > BTW, 100% out-of-sequence RTP packets leads to a lot more than
> just
> >  > "very very poor call quality."  I don't see how the conversation
> > could
> >  > even be coherent in that situation.
> >  >
> >  > What is more likely is that Wireshark's RTP stats are giving you
> some
> >  > distorted information.  I've found its stream analysis to be
> somewhat
> >  > buggy in that regard.
> >  >
> >  >
> >  >> I was wondering if I can decrease the MTU size to 250-500 on the
> > network
> >  >> card and use that card only for VoIP traffic. Will this have any
> bad
> >  >> effect on sip traffic/packets?
> >  >>
> >  >
> >  > No.  RTP packets are very small - much smaller than that MTU, or
> any
> >  > reasonable MTU you could set.
> >  >
> >  >
> >
> > ___
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> >
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> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> >
> > 
> >
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>
>
> --
> Alex Balashov
> Evariste Systems
> Web: http://www.evaristesys.com/
> Tel: (+1) (678) 954-0670
> Direct : (+1) (678) 954-0671
> Mobile : (+1) (706) 338-8599
>
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>
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[asterisk-users] OT: Re: sip clients for smart phones?

[Philipp Kempgen]

Tariq, fix your email client (it eats line breaks in text/plain).
And please leave out the parts which are not relevant (list
policy). Thanks.


   Philipp Kempgen

-- 
http://www.das-asterisk-buch.de  -  http://www.the-asterisk-book.com
Amooma GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
-- 

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Re: [asterisk-users] sip clients for smart phones?

[Tariq ..]

it is FRING i'm sorry for the mistype... 
www.fring.com  




AHD Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP
Syria: +963 944 618286
USA: +1 347 562 2308

From: [EMAIL PROTECTED]: [EMAIL PROTECTED]: Fri, 3 Oct 2008 12:00:16 
-0800Subject: Re: [asterisk-users] sip clients for smart phones?is it frig or 
fring? 


On Oct 3, 2008, at 11:49 AM, Tariq .. wrote:

try using Frig.. it's a great client with an SIP client.. i tried it on IPhone 
and on my N82 Nokia phone.. it works great on GPRS and Wi-Fi... and i DO use it 
with my two Asterisk Servers.. regards 


AHD Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP
Syria: +963 944 618286
USA: +1 347 562 2308

> From: [EMAIL PROTECTED]> To: asterisk-users@lists.digium.com> Date: Fri, 3 
> Oct 2008 15:33:52 -0400> Subject: Re: [asterisk-users] sip clients for smart 
> phones?> > I use the TytnII with Win Mob 6.1, customized ROM and it's working 
> for me -> through the back speaker though. > > > Thanks,> Matt G> > : 
> http://www.voipphreak.ca> : http://www.ratemydialplan.com> : 
> http://www.asterisk-jobs.com> > > -Original Message-> From: [EMAIL 
> PROTECTED]> [mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton> Sent: 
> Friday, October 03, 2008 2:57 PM> To: 'Asterisk Users Mailing List - 
> Non-Commercial Discussion'> Subject: Re: [asterisk-users] sip clients for 
> smart phones?> > This doesn't work in WinMo6.1 for some reason. Especially on 
> touchscreen> phones. Touch Diamond for instance.> > -Original 
> Message-> From: [EMAIL PROTECTED]> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Matt Gibson> Sent: October 3, 2008 11:52 AM> To: 'Asterisk Users Mailing List 
> - Non-Commercial Discussion'> Subject: Re: [asterisk-users] sip clients for 
> smart phones?> > This may help:> > 
> http://www.voipphreak.ca/2008/03/29/enable-the-hidden-voip-features-of-windo> 
> ws-mobile-6x-for-free-voip-calls-using-asterisk/> > Note, that most sip 
> clients for WINMOB suck and send the voice out the back> speaker instead of 
> the front speaker. I've found one other client (can't> remember the name now) 
> that works through the back speaker, but it's> payware, and nowhere near as 
> lightweight as this method. For now I deal with> the back speaker. > > 
> Thanks,> Matt G> > : http://www.voipphreak.ca> : 
> http://www.ratemydialplan.com> : http://www.asterisk-jobs.com> > 
> -Original Message-> From: [EMAIL PROTECTED]> [mailto:[EMAIL 
> PROTECTED] On Behalf Of Gordon> Henderson> Sent: Friday, October 03, 2008 
> 6:03 AM> To: Asterisk Users Mailing List - Non-Commercial Discussion> 
> Subject: Re: [asterisk-users] sip clients for smart phones?> > On Fri, 3 Oct 
> 2008, Babcock, Michael Alex wrote:> > > hi;> > curious does anyone know of a 
> smart phone client that could connect to> > asterisk?> > All the Nokia ones 
> which come with Wi-Fi and SIP can as far as I know. I > use an E90 over 
> Wi-Fi.> > Phone gets hot and battery life is about an 1.5 hours of talk time, 
> but...> > Gordon> > ___> -- 
> Bandwidth and Colocation Provided by http://www.api-digital.com --> > 
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona> Register Now: 
> http://www.astricon.net> > asterisk-users mailing list> To UNSUBSCRIBE or 
> update options visit:> 
> http://lists.digium.com/mailman/listinfo/asterisk-users> > > 
> ___> -- Bandwidth and Colocation 
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> 25 Phoenix, Arizona> Register Now: http://www.astricon.net> > asterisk-users 
> mailing list> To UNSUBSCRIBE or update options visit:> 
> http://lists.digium.com/mailman/listinfo/asterisk-users> > > 
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> 25 Phoenix, Arizona> Register Now: http://www.astricon.net> > asterisk-users 
> mailing list> To UNSUBSCRIBE or update options visit:> 
> http://lists.digium.com/mailman/listinfo/asterisk-users> > > 
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> mailing list> To UNSUBSCRIBE or update options visit:> 
> http://lists.digium.com/mailman/listinfo/asterisk-users

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Re: [asterisk-users] network monitoring - triggering a phone call in asterisk

[Alex Balashov]

Use callfiles, monitor Manager for the appropriate connect or failure 
events, output those to a state file or report them via a domain socket 
service that a Nagios plugin can connect to and inquire.

Eric Fort wrote:

> I'd like to integrate asterisk into the notification portion of my 
> network monitoring where a call is generated and a message is played 
> when a host or service failed to be reachable (something where nagios 
> could trigger an event in asterisk would rock!)  How could this best be 
> accomplished?  Anyone know an existing solution?
> 
> Eric
> 
> 
> 
> 
> 
> ___
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-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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Re: [asterisk-users] sip clients for smart phones?

[Babcock, Michael Alex]

i'm checking out "fring" right now
On Oct 3, 2008, at 12:02 PM, Mark Hamilton wrote:

> I don't even see it anywhere on TouchFlo3D. I don't see where to  
> even use
> it.
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Matt  
> Gibson
> Sent: October 3, 2008 3:34 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: Re: [asterisk-users] sip clients for smart phones?
>
> I use the TytnII with Win Mob 6.1, customized ROM and it's working  
> for me -
> through the back speaker though.
>
>
> Thanks,
> Matt G
>
> : http://www.voipphreak.ca
> : http://www.ratemydialplan.com
> : http://www.asterisk-jobs.com
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Mark  
> Hamilton
> Sent: Friday, October 03, 2008 2:57 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: Re: [asterisk-users] sip clients for smart phones?
>
> This doesn't work in WinMo6.1 for some reason. Especially on  
> touchscreen
> phones. Touch Diamond for instance.
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Matt  
> Gibson
> Sent: October 3, 2008 11:52 AM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: Re: [asterisk-users] sip clients for smart phones?
>
> This may help:
>
> http://www.voipphreak.ca/2008/03/29/enable-the-hidden-voip-features-of-windo
> ws-mobile-6x-for-free-voip-calls-using-asterisk/
>
> Note, that most sip clients for WINMOB suck and send the voice out  
> the back
> speaker instead of the front speaker. I've found one other client  
> (can't
> remember the name now) that works through the back speaker, but it's
> payware, and nowhere near as lightweight as this method. For now I  
> deal with
> the back speaker.
>
> Thanks,
> Matt G
>
> : http://www.voipphreak.ca
> : http://www.ratemydialplan.com
> : http://www.asterisk-jobs.com
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Gordon
> Henderson
> Sent: Friday, October 03, 2008 6:03 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] sip clients for smart phones?
>
> On Fri, 3 Oct 2008, Babcock, Michael Alex wrote:
>
>> hi;
>> curious does anyone know of a smart phone client that could connect  
>> to
>> asterisk?
>
> All the Nokia ones which come with Wi-Fi and SIP can as far as I  
> know. I
> use an E90 over Wi-Fi.
>
> Phone gets hot and battery life is about an 1.5 hours of talk time,  
> but...
>
> Gordon
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register Now: http://www.astricon.net
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>
>
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>
>
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Re: [asterisk-users] Improving the voice Quality,

[Alex Balashov]

sipp can simulate RTP traffic.

Jai Rangi wrote:

> Al and Alex,
> Thank you for your input,
> Sorry TDM is not the option at this time :( .
> Everything has been great until last 2-3 days. Machine loads is not the 
> issue, we have multiple asterisk server to share the load. Not much 
> change in traffic.
> 
> Now it been narrowed down to networking and we are trying to find out 
> where the issue is?  In our Firewall or SP's router. Does any one know 
> of any tool to simulate RTP traffic. Its pain to find out the bad calls 
> out of hundreds of calls.
> BTW, What should be right value for tos in sip.conf.
> We have
> tos=0x68
> Dont remember how did I come up with this value.
> 
> I found this
> http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+tos
> 
> tos=0x10  low delay
> tos=0x08  high throughput
> tos=0x04  high reliability
> tos=0x02  ECT bit set
> tos=0x01  CE bit set
> 
> 
> -Jai
> 
> 
> On Fri, Oct 3, 2008 at 4:58 AM, Al Baker <[EMAIL PROTECTED] 
> > wrote:
> 
> USE TDM Circuits - Voice Quality Good
> 
> Alex Balashov wrote:
>  > Jai Rangi wrote:
>  >
>  >
>  >> All,
>  >>
>  >> I am having audio quality problem in some calls (1-2%) on
> asterisk. I
>  >> captured RTP traffic using ethereal and this is what I found
> with the
>  >> problematic calls. (Worst cases)
>  >> Drop by Jitter buff: 25-75%
>  >> Out of Seq: 50-100% (100 % means very very poor call quality).
>  >>
>  >> Has anyone had similar problem? If yes, can you please share your
>  >> experience on how did you fix this?
>  >>
>  >
>  > Such poor performance is not fixable.  The network, connectivity
> issues,
>  > machine load, etc. needs to be addressed - the underlying cause, in
>  > other words.
>  >
>  > BTW, 100% out-of-sequence RTP packets leads to a lot more than just
>  > "very very poor call quality."  I don't see how the conversation
> could
>  > even be coherent in that situation.
>  >
>  > What is more likely is that Wireshark's RTP stats are giving you some
>  > distorted information.  I've found its stream analysis to be somewhat
>  > buggy in that regard.
>  >
>  >
>  >> I was wondering if I can decrease the MTU size to 250-500 on the
> network
>  >> card and use that card only for VoIP traffic. Will this have any bad
>  >> effect on sip traffic/packets?
>  >>
>  >
>  > No.  RTP packets are very small - much smaller than that MTU, or any
>  > reasonable MTU you could set.
>  >
>  >
> 
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> 
> 
> 
> 
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-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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[asterisk-users] network monitoring - triggering a phone call in asterisk

[Eric Fort]

I'd like to integrate asterisk into the notification portion of my network
monitoring where a call is generated and a message is played when a host or
service failed to be reachable (something where nagios could trigger an
event in asterisk would rock!)  How could this best be accomplished?  Anyone
know an existing solution?

Eric
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Re: [asterisk-users] sip clients for smart phones?

[Mark Hamilton]

I don't even see it anywhere on TouchFlo3D. I don't see where to even use
it.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt Gibson
Sent: October 3, 2008 3:34 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] sip clients for smart phones?

I use the TytnII with Win Mob 6.1, customized ROM and it's working for me -
through the back speaker though. 


Thanks,
Matt G

: http://www.voipphreak.ca
: http://www.ratemydialplan.com
: http://www.asterisk-jobs.com


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton
Sent: Friday, October 03, 2008 2:57 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] sip clients for smart phones?

This doesn't work in WinMo6.1 for some reason. Especially on touchscreen
phones. Touch Diamond for instance.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt Gibson
Sent: October 3, 2008 11:52 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] sip clients for smart phones?

This may help:

http://www.voipphreak.ca/2008/03/29/enable-the-hidden-voip-features-of-windo
ws-mobile-6x-for-free-voip-calls-using-asterisk/

Note, that most sip clients for WINMOB suck and send the voice out the back
speaker instead of the front speaker. I've found one other client (can't
remember the name now) that works through the back speaker, but it's
payware, and nowhere near as lightweight as this method. For now I deal with
the back speaker. 

Thanks,
Matt G

: http://www.voipphreak.ca
: http://www.ratemydialplan.com
: http://www.asterisk-jobs.com

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gordon
Henderson
Sent: Friday, October 03, 2008 6:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] sip clients for smart phones?

On Fri, 3 Oct 2008, Babcock, Michael Alex wrote:

> hi;
> curious does anyone know of a smart phone client that could connect to
> asterisk?

All the Nokia ones which come with Wi-Fi and SIP can as far as I know. I 
use an E90 over Wi-Fi.

Phone gets hot and battery life is about an 1.5 hours of talk time, but...

Gordon

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Re: [asterisk-users] sip clients for smart phones?

[Babcock, Michael Alex]

is it frig or fring?

On Oct 3, 2008, at 11:49 AM, Tariq .. wrote:

try using Frig.. it's a great client with an SIP client.. i tried it  
on IPhone and on my N82 Nokia phone.. it works great on GPRS and Wi- 
Fi... and i DO use it with my two Asterisk Servers..

regards




AHD Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP
Syria: +963 944 618286
USA: +1 347 562 2308


> From: [EMAIL PROTECTED]
> To: asterisk-users@lists.digium.com
> Date: Fri, 3 Oct 2008 15:33:52 -0400
> Subject: Re: [asterisk-users] sip clients for smart phones?
>
> I use the TytnII with Win Mob 6.1, customized ROM and it's working  
for me -

> through the back speaker though.
>
>
> Thanks,
> Matt G
>
> : http://www.voipphreak.ca
> : http://www.ratemydialplan.com
> : http://www.asterisk-jobs.com
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Mark  
Hamilton

> Sent: Friday, October 03, 2008 2:57 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: Re: [asterisk-users] sip clients for smart phones?
>
> This doesn't work in WinMo6.1 for some reason. Especially on  
touchscreen

> phones. Touch Diamond for instance.
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Matt  
Gibson

> Sent: October 3, 2008 11:52 AM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: Re: [asterisk-users] sip clients for smart phones?
>
> This may help:
>
> http://www.voipphreak.ca/2008/03/29/enable-the-hidden-voip-features-of-windo
> ws-mobile-6x-for-free-voip-calls-using-asterisk/
>
> Note, that most sip clients for WINMOB suck and send the voice out  
the back
> speaker instead of the front speaker. I've found one other client  
(can't

> remember the name now) that works through the back speaker, but it's
> payware, and nowhere near as lightweight as this method. For now I  
deal with

> the back speaker.
>
> Thanks,
> Matt G
>
> : http://www.voipphreak.ca
> : http://www.ratemydialplan.com
> : http://www.asterisk-jobs.com
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Gordon
> Henderson
> Sent: Friday, October 03, 2008 6:03 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] sip clients for smart phones?
>
> On Fri, 3 Oct 2008, Babcock, Michael Alex wrote:
>
> > hi;
> > curious does anyone know of a smart phone client that could  
connect to

> > asterisk?
>
> All the Nokia ones which come with Wi-Fi and SIP can as far as I  
know. I

> use an E90 over Wi-Fi.
>
> Phone gets hot and battery life is about an 1.5 hours of talk  
time, but...

>
> Gordon
>
> ___
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thanks for reading
Systems administrator and owner of http://gwhosting.net
msn: [EMAIL PROTECTED]
twitter: http://twitter.com/creepyblindy

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Re: [asterisk-users] sip clients for smart phones?

[Tariq ..]

try using Frig.. it's a great client with an SIP client.. i tried it on IPhone 
and on my N82 Nokia phone.. it works great on GPRS and Wi-Fi... and i DO use it 
with my two Asterisk Servers.. 
regards 




AHD Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP
Syria: +963 944 618286
USA: +1 347 562 2308

> From: [EMAIL PROTECTED]> To: asterisk-users@lists.digium.com> Date: Fri, 3 
> Oct 2008 15:33:52 -0400> Subject: Re: [asterisk-users] sip clients for smart 
> phones?> > I use the TytnII with Win Mob 6.1, customized ROM and it's working 
> for me -> through the back speaker though. > > > Thanks,> Matt G> > : 
> http://www.voipphreak.ca> : http://www.ratemydialplan.com> : 
> http://www.asterisk-jobs.com> > > -Original Message-> From: [EMAIL 
> PROTECTED]> [mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton> Sent: 
> Friday, October 03, 2008 2:57 PM> To: 'Asterisk Users Mailing List - 
> Non-Commercial Discussion'> Subject: Re: [asterisk-users] sip clients for 
> smart phones?> > This doesn't work in WinMo6.1 for some reason. Especially on 
> touchscreen> phones. Touch Diamond for instance.> > -Original 
> Message-> From: [EMAIL PROTECTED]> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Matt Gibson> Sent: October 3, 2008 11:52 AM> To: 'Asterisk Users Mailing List 
> - Non-Commercial Discussion'> Subject: Re: [asterisk-users] sip clients for 
> smart phones?> > This may help:> > 
> http://www.voipphreak.ca/2008/03/29/enable-the-hidden-voip-features-of-windo> 
> ws-mobile-6x-for-free-voip-calls-using-asterisk/> > Note, that most sip 
> clients for WINMOB suck and send the voice out the back> speaker instead of 
> the front speaker. I've found one other client (can't> remember the name now) 
> that works through the back speaker, but it's> payware, and nowhere near as 
> lightweight as this method. For now I deal with> the back speaker. > > 
> Thanks,> Matt G> > : http://www.voipphreak.ca> : 
> http://www.ratemydialplan.com> : http://www.asterisk-jobs.com> > 
> -Original Message-> From: [EMAIL PROTECTED]> [mailto:[EMAIL 
> PROTECTED] On Behalf Of Gordon> Henderson> Sent: Friday, October 03, 2008 
> 6:03 AM> To: Asterisk Users Mailing List - Non-Commercial Discussion> 
> Subject: Re: [asterisk-users] sip clients for smart phones?> > On Fri, 3 Oct 
> 2008, Babcock, Michael Alex wrote:> > > hi;> > curious does anyone know of a 
> smart phone client that could connect to> > asterisk?> > All the Nokia ones 
> which come with Wi-Fi and SIP can as far as I know. I > use an E90 over 
> Wi-Fi.> > Phone gets hot and battery life is about an 1.5 hours of talk time, 
> but...> > Gordon> > ___> -- 
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Re: [asterisk-users] sip clients for smart phones?

[Matt Gibson]

I use the TytnII with Win Mob 6.1, customized ROM and it's working for me -
through the back speaker though. 


Thanks,
Matt G

: http://www.voipphreak.ca
: http://www.ratemydialplan.com
: http://www.asterisk-jobs.com


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton
Sent: Friday, October 03, 2008 2:57 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] sip clients for smart phones?

This doesn't work in WinMo6.1 for some reason. Especially on touchscreen
phones. Touch Diamond for instance.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt Gibson
Sent: October 3, 2008 11:52 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] sip clients for smart phones?

This may help:

http://www.voipphreak.ca/2008/03/29/enable-the-hidden-voip-features-of-windo
ws-mobile-6x-for-free-voip-calls-using-asterisk/

Note, that most sip clients for WINMOB suck and send the voice out the back
speaker instead of the front speaker. I've found one other client (can't
remember the name now) that works through the back speaker, but it's
payware, and nowhere near as lightweight as this method. For now I deal with
the back speaker. 

Thanks,
Matt G

: http://www.voipphreak.ca
: http://www.ratemydialplan.com
: http://www.asterisk-jobs.com

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gordon
Henderson
Sent: Friday, October 03, 2008 6:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] sip clients for smart phones?

On Fri, 3 Oct 2008, Babcock, Michael Alex wrote:

> hi;
> curious does anyone know of a smart phone client that could connect to
> asterisk?

All the Nokia ones which come with Wi-Fi and SIP can as far as I know. I 
use an E90 over Wi-Fi.

Phone gets hot and battery life is about an 1.5 hours of talk time, but...

Gordon

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Re: [asterisk-users] sip clients for smart phones?

[Babcock, Michael Alex]

thanks i'll give this a try
mike

On Oct 3, 2008, at 10:56 AM, Mark Hamilton wrote:

> This doesn't work in WinMo6.1 for some reason. Especially on  
> touchscreen
> phones. Touch Diamond for instance.
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Matt  
> Gibson
> Sent: October 3, 2008 11:52 AM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: Re: [asterisk-users] sip clients for smart phones?
>
> This may help:
>
> http://www.voipphreak.ca/2008/03/29/enable-the-hidden-voip-features-of-windo
> ws-mobile-6x-for-free-voip-calls-using-asterisk/
>
> Note, that most sip clients for WINMOB suck and send the voice out  
> the back
> speaker instead of the front speaker. I've found one other client  
> (can't
> remember the name now) that works through the back speaker, but it's
> payware, and nowhere near as lightweight as this method. For now I  
> deal with
> the back speaker.
>
> Thanks,
> Matt G
>
> : http://www.voipphreak.ca
> : http://www.ratemydialplan.com
> : http://www.asterisk-jobs.com
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Gordon
> Henderson
> Sent: Friday, October 03, 2008 6:03 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] sip clients for smart phones?
>
> On Fri, 3 Oct 2008, Babcock, Michael Alex wrote:
>
>> hi;
>> curious does anyone know of a smart phone client that could connect  
>> to
>> asterisk?
>
> All the Nokia ones which come with Wi-Fi and SIP can as far as I  
> know. I
> use an E90 over Wi-Fi.
>
> Phone gets hot and battery life is about an 1.5 hours of talk time,  
> but...
>
> Gordon
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>
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thanks for reading
Systems administrator and owner of http://gwhosting.net
msn: [EMAIL PROTECTED]
twitter: http://twitter.com/creepyblindy


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Re: [asterisk-users] Improving the voice Quality,

[Gordon Henderson]

On Fri, 3 Oct 2008, Jai Rangi wrote:

> Al and Alex,
> Thank you for your input,
> Sorry TDM is not the option at this time :( .
> Everything has been great until last 2-3 days. Machine loads is not the
> issue, we have multiple asterisk server to share the load. Not much change
> in traffic.
>
> Now it been narrowed down to networking and we are trying to find out where
> the issue is?  In our Firewall or SP's router. Does any one know of any tool
> to simulate RTP traffic. Its pain to find out the bad calls out of hundreds
> of calls.

Not quite RTP traffic, but iperf might help - if you have access to both 
ends of the link. You can set a bandwidth and packet size which might help 
you.

Gordon

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Re: [asterisk-users] sip clients for smart phones?

[Mark Hamilton]

This doesn't work in WinMo6.1 for some reason. Especially on touchscreen
phones. Touch Diamond for instance.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt Gibson
Sent: October 3, 2008 11:52 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] sip clients for smart phones?

This may help:

http://www.voipphreak.ca/2008/03/29/enable-the-hidden-voip-features-of-windo
ws-mobile-6x-for-free-voip-calls-using-asterisk/

Note, that most sip clients for WINMOB suck and send the voice out the back
speaker instead of the front speaker. I've found one other client (can't
remember the name now) that works through the back speaker, but it's
payware, and nowhere near as lightweight as this method. For now I deal with
the back speaker. 

Thanks,
Matt G

: http://www.voipphreak.ca
: http://www.ratemydialplan.com
: http://www.asterisk-jobs.com

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gordon
Henderson
Sent: Friday, October 03, 2008 6:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] sip clients for smart phones?

On Fri, 3 Oct 2008, Babcock, Michael Alex wrote:

> hi;
> curious does anyone know of a smart phone client that could connect to
> asterisk?

All the Nokia ones which come with Wi-Fi and SIP can as far as I know. I 
use an E90 over Wi-Fi.

Phone gets hot and battery life is about an 1.5 hours of talk time, but...

Gordon

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Re: [asterisk-users] Improving the voice Quality,

[Jai Rangi]

Al and Alex,
Thank you for your input,
Sorry TDM is not the option at this time :( .
Everything has been great until last 2-3 days. Machine loads is not the
issue, we have multiple asterisk server to share the load. Not much change
in traffic.

Now it been narrowed down to networking and we are trying to find out where
the issue is?  In our Firewall or SP's router. Does any one know of any tool
to simulate RTP traffic. Its pain to find out the bad calls out of hundreds
of calls.
BTW, What should be right value for tos in sip.conf.
We have
tos=0x68
Dont remember how did I come up with this value.

I found this
http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+tos

tos=0x10 low delaytos=0x08 high throughput tos=0x04 high
reliabilitytos=0x02 ECT
bit set tos=0x01 CE bit set
-Jai


On Fri, Oct 3, 2008 at 4:58 AM, Al Baker <[EMAIL PROTECTED]> wrote:

> USE TDM Circuits - Voice Quality Good
>
> Alex Balashov wrote:
> > Jai Rangi wrote:
> >
> >
> >> All,
> >>
> >> I am having audio quality problem in some calls (1-2%) on asterisk. I
> >> captured RTP traffic using ethereal and this is what I found with the
> >> problematic calls. (Worst cases)
> >> Drop by Jitter buff: 25-75%
> >> Out of Seq: 50-100% (100 % means very very poor call quality).
> >>
> >> Has anyone had similar problem? If yes, can you please share your
> >> experience on how did you fix this?
> >>
> >
> > Such poor performance is not fixable.  The network, connectivity issues,
> > machine load, etc. needs to be addressed - the underlying cause, in
> > other words.
> >
> > BTW, 100% out-of-sequence RTP packets leads to a lot more than just
> > "very very poor call quality."  I don't see how the conversation could
> > even be coherent in that situation.
> >
> > What is more likely is that Wireshark's RTP stats are giving you some
> > distorted information.  I've found its stream analysis to be somewhat
> > buggy in that regard.
> >
> >
> >> I was wondering if I can decrease the MTU size to 250-500 on the network
> >> card and use that card only for VoIP traffic. Will this have any bad
> >> effect on sip traffic/packets?
> >>
> >
> > No.  RTP packets are very small - much smaller than that MTU, or any
> > reasonable MTU you could set.
> >
> >
>
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Re: [asterisk-users] How to add Callee's name into Dial command ?

[Mark Hamilton]

Why you didn't read the whole thread before saying that is beyond me.

 

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of satish patel
Sent: October 3, 2008 12:02 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] How to add Callee's name into Dial command ?

 

[Mark Hamilton] 


 why you people need this thing in dial command which can possible with
sip.conf callerid options 

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Re: [asterisk-users] sip clients for smart phones?

[Babcock, Michael Alex]

i should have been more clear, thanks a windows mobile smart phone sip  
client. sorry about that
On Oct 3, 2008, at 1:54 AM, randulo wrote:

> On Fri, Oct 3, 2008 at 10:08 AM, Babcock, Michael Alex
> <[EMAIL PROTECTED]> wrote:
>> curious does anyone know of a smart phone client that could connect  
>> to
>
> I'm the Nokia N95 can talk to asterisk.
>
> r
>
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thanks for reading
Systems administrator and owner of http://gwhosting.net
msn: [EMAIL PROTECTED]
twitter: http://twitter.com/creepyblindy


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Re: [asterisk-users] How to add Callee's name into Dial command ?

[satish patel]

 



2008/10/3 Joe Pukepail <[EMAIL PROTECTED]>


I think this is what you want: http://bugs.digium.com/view.php?id=8824


Thanks : this one very interesting.

Bottom line is it doesn't work at the moment right ?


  


On Fri, Oct 3, 2008 at 4:21 AM, Olivier <[EMAIL PROTECTED]> wrote:


Hi,

When dialing a number, I use :
exten => _123X, 1, Dial (SIP/${EXTEN})

Then, I get TRYING and RINGING SIP messages which both include this kind of
line :
To: 

Is it possible, configuring Asterisk 1.4, to get something like this instead
?
To: "John Doe" 

This way, I'm hoping to display callee's name beside (or instead of)
callee's number which would offer a double check for caller which might be
confusing extensions, for instance.


I tried this :
exten => _123X, 1, SIPAddHeader(To: Doe \)

but I still got :
To: 

Regards


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 why you people need this thing in dial command which can possible with
sip.conf callerid options 
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Re: [asterisk-users] Ok message

[Alex Balashov]

200 OK is a SIP response indicating the successful establishment of an 
INVITE transaction.

I can think of no reason why you would not be sending a 200 OK to your 
provider unless you are failing to Answer() the call in your dial plan 
and are instead sending them early media (183 Session in Progress).

A packet capture would be most helpful.

michel freiha wrote:

> Dear All,
> 
> I have a DTMF problem with VOxBone, the company that provide us the DID 
> numbers...Sometimes they sent us DTMF packets and sometimes not...
> VoxBone said asterisk is not sending back OK message to their Gateway 
> that's why they are not sending us the DTMF packets...How to force 
> Asterisk server to reply back by sending OK message?
> 
> Regards
> 
> 
> 
> 
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-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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Re: [asterisk-users] sip clients for smart phones?

[Matt Gibson]

This may help:

http://www.voipphreak.ca/2008/03/29/enable-the-hidden-voip-features-of-windo
ws-mobile-6x-for-free-voip-calls-using-asterisk/

Note, that most sip clients for WINMOB suck and send the voice out the back
speaker instead of the front speaker. I've found one other client (can't
remember the name now) that works through the back speaker, but it's
payware, and nowhere near as lightweight as this method. For now I deal with
the back speaker. 

Thanks,
Matt G

: http://www.voipphreak.ca
: http://www.ratemydialplan.com
: http://www.asterisk-jobs.com

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gordon
Henderson
Sent: Friday, October 03, 2008 6:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] sip clients for smart phones?

On Fri, 3 Oct 2008, Babcock, Michael Alex wrote:

> hi;
> curious does anyone know of a smart phone client that could connect to
> asterisk?

All the Nokia ones which come with Wi-Fi and SIP can as far as I know. I 
use an E90 over Wi-Fi.

Phone gets hot and battery life is about an 1.5 hours of talk time, but...

Gordon

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Re: [asterisk-users] MWI with Siemens Gigaset S450IP

[Kevin P. Fleming]

Olivier wrote:

> 2. R Hook-flash key is now available to transfer calls.
> In s450IP web management server, its defaults settings are :
> Application-type: dtmf-relay
> Application-signal: 16
> 
> Is there anything to configure in features.conf, extensionsconf or
> elsewhere to trigger transfers when R key is pressed ?

I don't believe there is any support for hook-flash style transfers over
SIP in Asterisk; that key should be programmed to use standard SIP
transfer methods, not DTMF emulation methods.

-- 
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)

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Re: [asterisk-users] Ok message

[Philipp Kempgen]

michel freiha schrieb:

> I have a DTMF problem with VOxBone, the company that provide us the DID
> numbers...Sometimes they sent us DTMF packets and sometimes not...
> VoxBone said asterisk is not sending back OK message to their Gateway that's
> why they are not sending us the DTMF packets...How to force Asterisk server
> to reply back by sending OK message?

Why should Asterisk send OK if they didn't send the DTMF packet
in the first place?

SIP debug trace?


   Philipp Kempgen

-- 
http://www.das-asterisk-buch.de  -  http://www.the-asterisk-book.com
Amooma GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
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Re: [asterisk-users] How to add Callee's name into Dial command ?

[Olivier]

2008/10/3 Joe Pukepail <[EMAIL PROTECTED]>

> I think this is what you want: http://bugs.digium.com/view.php?id=8824
>

Thanks : this one very interesting.

Bottom line is it doesn't work at the moment right ?

> 
>
> On Fri, Oct 3, 2008 at 4:21 AM, Olivier <[EMAIL PROTECTED]> wrote:
>
>> Hi,
>>
>> When dialing a number, I use :
>> exten => _123X, 1, Dial (SIP/${EXTEN})
>>
>> Then, I get TRYING and RINGING SIP messages which both include this kind
>> of line :
>> To: 
>>
>> Is it possible, configuring Asterisk 1.4, to get something like this
>> instead ?
>> To: "John Doe" 
>>
>> This way, I'm hoping to display callee's name beside (or instead of)
>> callee's number which would offer a double check for caller which might be
>> confusing extensions, for instance.
>>
>>
>> I tried this :
>> exten => _123X, 1, SIPAddHeader(To: Doe \> \;user=phone\>)
>>
>> but I still got :
>> To: 
>>
>> Regards
>>
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Re: [asterisk-users] How to add Callee's name into Dial command ?

[Olivier]

2008/10/3 Mr Shunz <[EMAIL PROTECTED]>

> Hi all,
>
> after a little bit of googleing, seems that correct syntax is:
>
> exten => _123X, 1,
> SIPAddHeader(P-Asserted-Identity:"${CALLERID(name)}"
> )
> (notice the sip:${CALLERID(num)})


This could make P-Asserted-Identity field appear in INVITE header.
Thanks for that !!

Unfortunately, this doesn't help much to force Asterisk to query caller's
callerid field in sip.conf .



>
> but, IIUC, this sets the header for *outbound* call to 123X number, so
> don't know if CALLER
> can see the header ... and actually, ${CALLERID(num|name)} it's the
> CALLER name, so maybe
> should be ${EXTEN} (i.e. the CALLED number) like
>
> exten => _123X, 1, SIPAddHeader(P-Asserted-Identity:"doe" )
>
> but sorry, can't test by now...
>
> cheers
>
> --
> 
> Daniele Santi   .o.
> [EMAIL PROTECTED] ..o () ascii ribbon campaign
> Linux User #415108  ooo /\  www.asciiribbon.org
> 
>
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Re: [asterisk-users] t1 cards

[Lyle Giese]

T1 is NOT DSL.  Most T1 links you purchase now are brought into your 
building with a type of DSL conversion to extend the distance between 
repeaters/amplifiers.  T1 is purely a digital signal.  DSL converts the 
ones and zeros to audio(multiple tones to provide multi channels of 
data).  A simple analogy is comparing a T1 to DSL as a serial port to a 
modem.


Back in the old days before fiber, copper T1's between CO's had their 
repeaters placed aproximately 1 mile apart.  Best case going T1 port to 
T1 port, I would not expect this to work reliably at distances greater 
than one mile or 1.6 km but that does depend on the quality of the cable 
also.


But in my mind, I would be seriously concerned about lightening 
protection.  I have been around telco's and privately owned facilities 
for a long time and see lightening to be a very serious issue in this 
scenerio. I have seen short distance copper replaced by fiber because of 
issues over time with lightening damage despite having proper telco 
grade protection.


Lyle

Jeff LaCoursiere wrote:

I would say miles.  DSL limits for equiv bandwidth is around 3 miles if I
recall correctly.

j

On Fri, 3 Oct 2008, Eric Fort wrote:

  

without any other hardware than 2 bare ass pci based t1/e1 cards wired back
to back how far can one go between them?  additional hardware defeats the
purpose.

Eric

On Fri, Oct 3, 2008 at 3:01 AM, Gordon Henderson
<[EMAIL PROTECTED]<[EMAIL PROTECTED]>


wrote:
  
On Fri, 3 Oct 2008, Eric Fort wrote:


 yes, more than 300 meters (longer than copper based ethernet allows).  Yes
  

to E1, as I understand it, it's just a config change on many cards anyway.
I'm specificly looking at pci based t1/e1 cards because I'm finding single
port cards on ebay going for 100-200 usd.  in some cases I may want to
drive
a channel bank at the far end, thus t1/e1.  anyone have experience on how
far these pci based cards will drive when wired back to back?



Looks like this is the thing then:

 http://www.blackbox.com/Catalog/Detail.aspx?cid=381,1452,1468&mid=5362

Just over $1000 a pair...

couple that with an OpenVox PRI card at one end, channel bank at the other,
and off you go...

Gordon



  

Eric

On Thu, Oct 2, 2008 at 11:34 PM, Gordon Henderson <
[EMAIL PROTECTED] <[EMAIL PROTECTED]> <
[EMAIL PROTECTED] <[EMAIL PROTECTED]>>> wrote:

 On Thu, 2 Oct 2008, Eric Fort wrote:


 I presently need to connect a few channels of voice and data between

  

multiple locations where I own the copper between them.  Each location
exceeds 300M from any other location.  I'm thinking of generating T1's
and
running those between locations.  If I use PC based cards wired back to
back
(I can do that, right?) what kind of distance can I expect to be able to
span without needing repeaters?  What inexpensive cards can you
recommend
for use with asterisk?  I'm considering either digium or sangoma.  Would
I
get any better performance if I used a sync-serial card connected to a
separate csu/dsu?




300 metres, right? (not 300 miles?)

Why stop at T1? Go for E1 :) with the right kit at each end you ought to
be
able to get 2Mb/sec or more. (distance depending)

Personally, I'd go for a technology that gave me Ethernet at each end -
then it makes it much easier to mix voice and data - But using something
like a sync. modem and line driver then you need a media converter of
some
sorts at each end which might bump up the cost - at the savings of the E1
card in the PC though. Last time I had bare copper to play with (a BT
EPS8
circuit) I had a 2Mb modem at each end going into a Cisco 2600 which was
running CHDLC over the link and acting as nothing more than a dumb media
converter to give me Ethernet at each end. This was 6 years ago though.

Ah, Looks like the technology has improved somewhat:

 http://www.blackbox.com/Catalog/Detail.aspx?cid=381,1452,1468&mid=5261

From the UK site:

Or even:

 http://www.blackbox.com/Catalog/Detail.aspx?cid=425,1423,1424&mid=4946

(same thing from the UK site:)



http://www.blackbox.co.uk/solutions/display.asp?cs=dvh&id=1&doc=lb300a-r2&tx=LAN&sx=Network%20Appliances

You need a pair, obviously...

Hm. US site is $305, UK ?253. Rip-off Britain again by the looks of
it

As for inexpensive cards - OpenVox. Their E1 cards seem to work OK, but
if
using a LAN extender, then they're not neeed at all...

Gordon

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Re: [asterisk-users] How to add Callee's name into Dial command ?

[Olivier]

2008/10/3 Mr Shunz <[EMAIL PROTECTED]>

> > (Even with the missing '}' at the end that I fixed
>
> oops, always forget about those :'(
>
> btw, in our installations we provide a centralized xml phonebook
> (and phones that support it, mostly grandstream and thomson), and
> they (the phones) autmatically set the called name on the display,
> both for internal and external numbers ;)


Are you of that ?
I'm not 100% certain, but I think Thomson phones wouldn't query centralized
directory for outbound calls.
I think centralized directory is only queried when using Directory key.

>
>
> cheers
>
> --
> 
> Daniele Santi   .o.
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> 
>
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Re: [asterisk-users] How to add Callee's name into Dial command ?

[Joe Pukepail]

I think this is what you want: http://bugs.digium.com/view.php?id=8824

On Fri, Oct 3, 2008 at 4:21 AM, Olivier <[EMAIL PROTECTED]> wrote:

> Hi,
>
> When dialing a number, I use :
> exten => _123X, 1, Dial (SIP/${EXTEN})
>
> Then, I get TRYING and RINGING SIP messages which both include this kind of
> line :
> To: 
>
> Is it possible, configuring Asterisk 1.4, to get something like this
> instead ?
> To: "John Doe" 
>
> This way, I'm hoping to display callee's name beside (or instead of)
> callee's number which would offer a double check for caller which might be
> confusing extensions, for instance.
>
>
> I tried this :
> exten => _123X, 1, SIPAddHeader(To: Doe \ \;user=phone\>)
>
> but I still got :
> To: 
>
> Regards
>
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[asterisk-users] Ok message

[michel freiha]

Dear All,

I have a DTMF problem with VOxBone, the company that provide us the DID
numbers...Sometimes they sent us DTMF packets and sometimes not...
VoxBone said asterisk is not sending back OK message to their Gateway that's
why they are not sending us the DTMF packets...How to force Asterisk server
to reply back by sending OK message?

Regards
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[asterisk-users] MWI with Siemens Gigaset S450IP

[Olivier]

Hi,

1. Here http://www.voip-info.org/wiki/view/Siemens+Gigaset+S450IP  it is
mentioned MWI is now working.

In my testings with lastest 02123 firmware, MWI is blinking when missed
calls but not when a message in present in voicemail.
With SIP debug I can see "481 Call Leg/Transaction Does Not Exist" replies
to NOTIFY announcing new messages.
With previous firmware, I had "415 Unsupported Media" if my memory is
correct.

Has anyone been any further ?

2. R Hook-flash key is now available to transfer calls.
In s450IP web management server, its defaults settings are :
Application-type: dtmf-relay
Application-signal: 16

Is there anything to configure in features.conf, extensionsconf or elsewhere
to trigger transfers when R key is pressed ?

Regards
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Re: [asterisk-users] How to add Callee's name into Dial command ?

[Mr Shunz]

Hi all,

after a little bit of googleing, seems that correct syntax is:

exten => _123X, 1,
SIPAddHeader(P-Asserted-Identity:"${CALLERID(name)}"
)
(notice the sip:${CALLERID(num)})

but, IIUC, this sets the header for *outbound* call to 123X number, so
don't know if CALLER
can see the header ... and actually, ${CALLERID(num|name)} it's the
CALLER name, so maybe
should be ${EXTEN} (i.e. the CALLED number) like

exten => _123X, 1, SIPAddHeader(P-Asserted-Identity:"doe" )

but sorry, can't test by now...

cheers

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Re: [asterisk-users] How to add Callee's name into Dial command ?

[Mr Shunz]

> (Even with the missing '}' at the end that I fixed

oops, always forget about those :'(

btw, in our installations we provide a centralized xml phonebook
(and phones that support it, mostly grandstream and thomson), and
they (the phones) autmatically set the called name on the display,
both for internal and external numbers ;)

cheers

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Re: [asterisk-users] How to add Callee's name into Dial command ?

[Josiah Bryan]

Mr Shunz wrote:
> [snip]
> 
>> To make myself clear, here is what I'm trying to do : when Alice is calling
>> Bob (Alice ---> Asterisk--->Bob), I would like Bob's phone to
>> display Alice's name  (no problem, for that) but I would also like Alice's
>> phone screen to display Bob's name (instead of Bob's number)
> 
> mmm ... this wasn't clear on your OP ...
> so you need to show the CALLED name on the CALLER phone ...
> 
>> My SIP hardphone is capable of displaying P-Asserted-Identity in outbound 
>> calls (not just inbound) but I
>> couldn't find any way to teach Asterisk to fill this P-Asserted-Identity 
>> header :
> 
> you can try with: (*** untested ***)
> 
> exten => _123X, 1, SIPAddHeader(P-Asserted-Identity:
> '${CALLERID(name)' <${CALLERID(num)>)

Interesting idea - I can see that being very useful. I know on my old 
SPPA-841s they would do that - but it was based on looking up the dialed 
number in the internal directory (which I programmed using a perl script 
in the asterisk server.) So, when I dialed 213, the name appeared that I 
dialed, confirming I had the right person.

Unfortunately, I never have been able to get my Polycom SoundPoint IP 
500 phones to do that.

I just tested the SIPAddHeader command given above - doesn't work with 
the Polycom Soundpoint IP 500 that I tested with. (Even with the missing 
'}' at the end that I fixed - still doesn't work.)

Good idea thought - anybody have any magic that might make that work?

-josiah

-- 
Josiah Bryan
IT Manager
Productive Concepts, Inc.
[EMAIL PROTECTED]
(765) 964-6009, ext. 224


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Re: [asterisk-users] How to add Callee's name into Dial command ?

[Mr Shunz]

[snip]

> To make myself clear, here is what I'm trying to do : when Alice is calling
> Bob (Alice ---> Asterisk--->Bob), I would like Bob's phone to
> display Alice's name  (no problem, for that) but I would also like Alice's
> phone screen to display Bob's name (instead of Bob's number)

mmm ... this wasn't clear on your OP ...
so you need to show the CALLED name on the CALLER phone ...

> My SIP hardphone is capable of displaying P-Asserted-Identity in outbound 
> calls (not just inbound) but I
> couldn't find any way to teach Asterisk to fill this P-Asserted-Identity 
> header :

you can try with: (*** untested ***)

exten => _123X, 1, SIPAddHeader(P-Asserted-Identity:
'${CALLERID(name)' <${CALLERID(num)>)

cheers

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Re: [asterisk-users] How to add Callee's name into Dial command ?

[Olivier]

Unfortunaltely, both callerid='Jhon Doe' <1234> and
Set(CALLERID(name)="Doe") don't work as I would like them to.

For both, caller's phone screen still displays callee's number, not callee's
name.
When looking at SIP messages, you can can see that, unlike caller's name
which is included in INVITE Header  From: field, INVITE Header To: remains
filled with callee's number, not its name.

To make myself clear, here is what I'm trying to do : when Alice is calling
Bob (Alice ---> Asterisk--->Bob), I would like Bob's phone to
display Alice's name  (no problem, for that) but I would also like Alice's
phone screen to display Bob's name (instead of Bob's number)

My SIP hardphone is capable of displaying P-Asserted-Identity in outbound
calls (not just inbound) but I couldn't find any way to teach Asterisk to
fill this P-Asserted-Identity header :

Alice
Asterisk Bob
 - INVITE bob >

--INVITE Bob ->
  < TRYING with
Callername set 
< TRYING with Callername set 
 < RINGING with
Callername set 
< RINGING with Callername set 


Regards



2008/10/3 Mr Shunz <[EMAIL PROTECTED]>

> Hi,
>
> [snip[
>
> > This way, I'm hoping to display callee's name beside (or instead of)
> > callee's number which would offer a double check for caller which might
> be
> > confusing extensions, for instance.
>
> you can set callerid per-peer in sip.conf
> like:
>
> callerid='Jhon Doe' <1234>
>
> this should work autmagically ;)
>
> cheers
>
> --
> 
> Daniele Santi   .o.
> [EMAIL PROTECTED] ..o () ascii ribbon campaign
> Linux User #415108  ooo /\  www.asciiribbon.org
> 
>
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Re: [asterisk-users] Improving the voice Quality,

[Al Baker]

USE TDM Circuits - Voice Quality Good

Alex Balashov wrote:
> Jai Rangi wrote:
>
>   
>> All,
>>
>> I am having audio quality problem in some calls (1-2%) on asterisk. I 
>> captured RTP traffic using ethereal and this is what I found with the 
>> problematic calls. (Worst cases)
>> Drop by Jitter buff: 25-75%
>> Out of Seq: 50-100% (100 % means very very poor call quality).
>>
>> Has anyone had similar problem? If yes, can you please share your 
>> experience on how did you fix this? 
>> 
>
> Such poor performance is not fixable.  The network, connectivity issues, 
> machine load, etc. needs to be addressed - the underlying cause, in 
> other words.
>
> BTW, 100% out-of-sequence RTP packets leads to a lot more than just 
> "very very poor call quality."  I don't see how the conversation could 
> even be coherent in that situation.
>
> What is more likely is that Wireshark's RTP stats are giving you some 
> distorted information.  I've found its stream analysis to be somewhat 
> buggy in that regard.
>
>   
>> I was wondering if I can decrease the MTU size to 250-500 on the network 
>> card and use that card only for VoIP traffic. Will this have any bad 
>> effect on sip traffic/packets?
>> 
>
> No.  RTP packets are very small - much smaller than that MTU, or any 
> reasonable MTU you could set.
>
>   

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Re: [asterisk-users] How to add Callee's name into Dial command ?

[Mr Shunz]

Hi,

[snip[

> This way, I'm hoping to display callee's name beside (or instead of)
> callee's number which would offer a double check for caller which might be
> confusing extensions, for instance.

you can set callerid per-peer in sip.conf
like:

callerid='Jhon Doe' <1234>

this should work autmagically ;)

cheers

-- 

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[EMAIL PROTECTED] ..o () ascii ribbon campaign
Linux User #415108  ooo /\  www.asciiribbon.org


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Re: [asterisk-users] t1 cards

[Jeff LaCoursiere]

I would say miles.  DSL limits for equiv bandwidth is around 3 miles if I
recall correctly.

j

On Fri, 3 Oct 2008, Eric Fort wrote:

> without any other hardware than 2 bare ass pci based t1/e1 cards wired back
> to back how far can one go between them?  additional hardware defeats the
> purpose.
>
> Eric
>
> On Fri, Oct 3, 2008 at 3:01 AM, Gordon Henderson
> <[EMAIL PROTECTED]<[EMAIL PROTECTED]>
> > wrote:
>
> > On Fri, 3 Oct 2008, Eric Fort wrote:
> >
> >  yes, more than 300 meters (longer than copper based ethernet allows).  Yes
> >> to E1, as I understand it, it's just a config change on many cards anyway.
> >> I'm specificly looking at pci based t1/e1 cards because I'm finding single
> >> port cards on ebay going for 100-200 usd.  in some cases I may want to
> >> drive
> >> a channel bank at the far end, thus t1/e1.  anyone have experience on how
> >> far these pci based cards will drive when wired back to back?
> >>
> >
> > Looks like this is the thing then:
> >
> >  http://www.blackbox.com/Catalog/Detail.aspx?cid=381,1452,1468&mid=5362
> >
> > Just over $1000 a pair...
> >
> > couple that with an OpenVox PRI card at one end, channel bank at the other,
> > and off you go...
> >
> > Gordon
> >
> >
> >
> >> Eric
> >>
> >> On Thu, Oct 2, 2008 at 11:34 PM, Gordon Henderson <
> >> [EMAIL PROTECTED] <[EMAIL PROTECTED]> <
> >> [EMAIL PROTECTED] <[EMAIL PROTECTED]>>> wrote:
> >>
> >>  On Thu, 2 Oct 2008, Eric Fort wrote:
> >>>
> >>>  I presently need to connect a few channels of voice and data between
> >>>
>  multiple locations where I own the copper between them.  Each location
>  exceeds 300M from any other location.  I'm thinking of generating T1's
>  and
>  running those between locations.  If I use PC based cards wired back to
>  back
>  (I can do that, right?) what kind of distance can I expect to be able to
>  span without needing repeaters?  What inexpensive cards can you
>  recommend
>  for use with asterisk?  I'm considering either digium or sangoma.  Would
>  I
>  get any better performance if I used a sync-serial card connected to a
>  separate csu/dsu?
> 
> 
> >>> 300 metres, right? (not 300 miles?)
> >>>
> >>> Why stop at T1? Go for E1 :) with the right kit at each end you ought to
> >>> be
> >>> able to get 2Mb/sec or more. (distance depending)
> >>>
> >>> Personally, I'd go for a technology that gave me Ethernet at each end -
> >>> then it makes it much easier to mix voice and data - But using something
> >>> like a sync. modem and line driver then you need a media converter of
> >>> some
> >>> sorts at each end which might bump up the cost - at the savings of the E1
> >>> card in the PC though. Last time I had bare copper to play with (a BT
> >>> EPS8
> >>> circuit) I had a 2Mb modem at each end going into a Cisco 2600 which was
> >>> running CHDLC over the link and acting as nothing more than a dumb media
> >>> converter to give me Ethernet at each end. This was 6 years ago though.
> >>>
> >>> Ah, Looks like the technology has improved somewhat:
> >>>
> >>>  http://www.blackbox.com/Catalog/Detail.aspx?cid=381,1452,1468&mid=5261
> >>>
> >>> From the UK site:
> >>>
> >>> Or even:
> >>>
> >>>  http://www.blackbox.com/Catalog/Detail.aspx?cid=425,1423,1424&mid=4946
> >>>
> >>> (same thing from the UK site:)
> >>>
> >>>
> >>>
> >>> http://www.blackbox.co.uk/solutions/display.asp?cs=dvh&id=1&doc=lb300a-r2&tx=LAN&sx=Network%20Appliances
> >>>
> >>> You need a pair, obviously...
> >>>
> >>> Hm. US site is $305, UK ?253. Rip-off Britain again by the looks of
> >>> it
> >>>
> >>> As for inexpensive cards - OpenVox. Their E1 cards seem to work OK, but
> >>> if
> >>> using a LAN extender, then they're not neeed at all...
> >>>
> >>> Gordon
> >>>
> >>> ___
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> >>>
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>

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Re: [asterisk-users] DTMF

[Tzafrir Cohen]

On Fri, Oct 03, 2008 at 01:09:41PM +0300, michel freiha wrote:
> On asterisk CLI type rtp debug...YOu should see if DTMF tones are sent in
> the RTP flow and the protocol used

That is: you'll see there rfc2833 digits (sent as rtp signalling). This
naturally won't show any DTMF digits sent in-band (as part of the audio
stream)

DTMF != digit

-- 
   Tzafrir Cohen
icq#16849755  jabber:[EMAIL PROTECTED]
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Re: [asterisk-users] t1 cards

[Eric Fort]

without any other hardware than 2 bare ass pci based t1/e1 cards wired back
to back how far can one go between them?  additional hardware defeats the
purpose.

Eric

On Fri, Oct 3, 2008 at 3:01 AM, Gordon Henderson
<[EMAIL PROTECTED]<[EMAIL PROTECTED]>
> wrote:

> On Fri, 3 Oct 2008, Eric Fort wrote:
>
>  yes, more than 300 meters (longer than copper based ethernet allows).  Yes
>> to E1, as I understand it, it's just a config change on many cards anyway.
>> I'm specificly looking at pci based t1/e1 cards because I'm finding single
>> port cards on ebay going for 100-200 usd.  in some cases I may want to
>> drive
>> a channel bank at the far end, thus t1/e1.  anyone have experience on how
>> far these pci based cards will drive when wired back to back?
>>
>
> Looks like this is the thing then:
>
>  http://www.blackbox.com/Catalog/Detail.aspx?cid=381,1452,1468&mid=5362
>
> Just over $1000 a pair...
>
> couple that with an OpenVox PRI card at one end, channel bank at the other,
> and off you go...
>
> Gordon
>
>
>
>> Eric
>>
>> On Thu, Oct 2, 2008 at 11:34 PM, Gordon Henderson <
>> [EMAIL PROTECTED] <[EMAIL PROTECTED]> <
>> [EMAIL PROTECTED] <[EMAIL PROTECTED]>>> wrote:
>>
>>  On Thu, 2 Oct 2008, Eric Fort wrote:
>>>
>>>  I presently need to connect a few channels of voice and data between
>>>
 multiple locations where I own the copper between them.  Each location
 exceeds 300M from any other location.  I'm thinking of generating T1's
 and
 running those between locations.  If I use PC based cards wired back to
 back
 (I can do that, right?) what kind of distance can I expect to be able to
 span without needing repeaters?  What inexpensive cards can you
 recommend
 for use with asterisk?  I'm considering either digium or sangoma.  Would
 I
 get any better performance if I used a sync-serial card connected to a
 separate csu/dsu?


>>> 300 metres, right? (not 300 miles?)
>>>
>>> Why stop at T1? Go for E1 :) with the right kit at each end you ought to
>>> be
>>> able to get 2Mb/sec or more. (distance depending)
>>>
>>> Personally, I'd go for a technology that gave me Ethernet at each end -
>>> then it makes it much easier to mix voice and data - But using something
>>> like a sync. modem and line driver then you need a media converter of
>>> some
>>> sorts at each end which might bump up the cost - at the savings of the E1
>>> card in the PC though. Last time I had bare copper to play with (a BT
>>> EPS8
>>> circuit) I had a 2Mb modem at each end going into a Cisco 2600 which was
>>> running CHDLC over the link and acting as nothing more than a dumb media
>>> converter to give me Ethernet at each end. This was 6 years ago though.
>>>
>>> Ah, Looks like the technology has improved somewhat:
>>>
>>>  http://www.blackbox.com/Catalog/Detail.aspx?cid=381,1452,1468&mid=5261
>>>
>>> From the UK site:
>>>
>>> Or even:
>>>
>>>  http://www.blackbox.com/Catalog/Detail.aspx?cid=425,1423,1424&mid=4946
>>>
>>> (same thing from the UK site:)
>>>
>>>
>>>
>>> http://www.blackbox.co.uk/solutions/display.asp?cs=dvh&id=1&doc=lb300a-r2&tx=LAN&sx=Network%20Appliances
>>>
>>> You need a pair, obviously...
>>>
>>> Hm. US site is $305, UK £253. Rip-off Britain again by the looks of
>>> it
>>>
>>> As for inexpensive cards - OpenVox. Their E1 cards seem to work OK, but
>>> if
>>> using a LAN extender, then they're not neeed at all...
>>>
>>> Gordon
>>>
>>> ___
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>>>
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Re: [asterisk-users] DTMF

[michel freiha]

On asterisk CLI type rtp debug...YOu should see if DTMF tones are sent in
the RTP flow and the protocol used

Regards

On Thu, Oct 2, 2008 at 11:42 PM, Barton Fisher <[EMAIL PROTECTED]> wrote:

>  How can I know for sure if SIP Trunk Provider is sending DTMF 'inband' or
> 'rfc2833'?
> And more importantly if they could be sending both?
> If I specify 'inband' should they honor that?
>
> Thanks, Bart
>
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Re: [asterisk-users] sip clients for smart phones?

[Rob Hillis]

Babcock, Michael Alex wrote:
> windows smart phone v 6.0 example
> htc shadow
> is what i have. It has wifi abilitys.
>   

Googling for "windows mobile sip" yeilds a multitude of results.  I'm 
sure one of them will point you in the right direction.

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Re: [asterisk-users] sip clients for smart phones?

[Gordon Henderson]

On Fri, 3 Oct 2008, Babcock, Michael Alex wrote:

> hi;
> curious does anyone know of a smart phone client that could connect to
> asterisk?

All the Nokia ones which come with Wi-Fi and SIP can as far as I know. I 
use an E90 over Wi-Fi.

Phone gets hot and battery life is about an 1.5 hours of talk time, but...

Gordon

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Re: [asterisk-users] How to add Callee's name into Dial command ?

[Rob Hillis]

Olivier wrote:
>
> 2008/10/3 Olivier <[EMAIL PROTECTED] >
>
> Hi,
>
> When dialing a number, I use :
> exten => _123X, 1, Dial (SIP/${EXTEN})
>
> Then, I get TRYING and RINGING SIP messages which both include
> this kind of line :
> To: mailto:[EMAIL PROTECTED]>;user=phone>
>
> Is it possible, configuring Asterisk 1.4, to get something like
> this instead ?
> To: "John Doe"  ;user=phone>
>
> This way, I'm hoping to display callee's name beside (or instead
> of) callee's number which would offer a double check for caller
> which might be confusing extensions, for instance.
>
>
> I tried this :
> exten => _123X, 1, SIPAddHeader(To: Doe \ \;user=phone\>)
>
> but I still got :
> To: mailto:[EMAIL PROTECTED]>;user=phone>
>
> I must add I also tried without success :
> exten => _123X, 1, Set(SIALPEERNAME=Doe)

Have you tried Set(CALLERID(name)="Doe")?  This the normal method for 
setting caller ID names in Asterisk.

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Re: [asterisk-users] t1 cards

[Gordon Henderson]

On Fri, 3 Oct 2008, Eric Fort wrote:


yes, more than 300 meters (longer than copper based ethernet allows).  Yes
to E1, as I understand it, it's just a config change on many cards anyway.
I'm specificly looking at pci based t1/e1 cards because I'm finding single
port cards on ebay going for 100-200 usd.  in some cases I may want to drive
a channel bank at the far end, thus t1/e1.  anyone have experience on how
far these pci based cards will drive when wired back to back?


Looks like this is the thing then:

  http://www.blackbox.com/Catalog/Detail.aspx?cid=381,1452,1468&mid=5362

Just over $1000 a pair...

couple that with an OpenVox PRI card at one end, channel bank at the 
other, and off you go...


Gordon




Eric

On Thu, Oct 2, 2008 at 11:34 PM, Gordon Henderson <
[EMAIL PROTECTED] <[EMAIL PROTECTED]>> wrote:


On Thu, 2 Oct 2008, Eric Fort wrote:

 I presently need to connect a few channels of voice and data between

multiple locations where I own the copper between them.  Each location
exceeds 300M from any other location.  I'm thinking of generating T1's and
running those between locations.  If I use PC based cards wired back to
back
(I can do that, right?) what kind of distance can I expect to be able to
span without needing repeaters?  What inexpensive cards can you recommend
for use with asterisk?  I'm considering either digium or sangoma.  Would I
get any better performance if I used a sync-serial card connected to a
separate csu/dsu?



300 metres, right? (not 300 miles?)

Why stop at T1? Go for E1 :) with the right kit at each end you ought to be
able to get 2Mb/sec or more. (distance depending)

Personally, I'd go for a technology that gave me Ethernet at each end -
then it makes it much easier to mix voice and data - But using something
like a sync. modem and line driver then you need a media converter of some
sorts at each end which might bump up the cost - at the savings of the E1
card in the PC though. Last time I had bare copper to play with (a BT EPS8
circuit) I had a 2Mb modem at each end going into a Cisco 2600 which was
running CHDLC over the link and acting as nothing more than a dumb media
converter to give me Ethernet at each end. This was 6 years ago though.

Ah, Looks like the technology has improved somewhat:

 http://www.blackbox.com/Catalog/Detail.aspx?cid=381,1452,1468&mid=5261

From the UK site:

Or even:

 http://www.blackbox.com/Catalog/Detail.aspx?cid=425,1423,1424&mid=4946

(same thing from the UK site:)


http://www.blackbox.co.uk/solutions/display.asp?cs=dvh&id=1&doc=lb300a-r2&tx=LAN&sx=Network%20Appliances

You need a pair, obviously...

Hm. US site is $305, UK £253. Rip-off Britain again by the looks of it

As for inexpensive cards - OpenVox. Their E1 cards seem to work OK, but if
using a LAN extender, then they're not neeed at all...

Gordon

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Re: [asterisk-users] sip clients for smart phones?

[randulo]

On Fri, Oct 3, 2008 at 10:08 AM, Babcock, Michael Alex
<[EMAIL PROTECTED]> wrote:
> curious does anyone know of a smart phone client that could connect to

I'm the Nokia N95 can talk to asterisk.

r

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Re: [asterisk-users] How to add Callee's name into Dial command ?

[Olivier]

2008/10/3 Olivier <[EMAIL PROTECTED]>

> Hi,
>
> When dialing a number, I use :
> exten => _123X, 1, Dial (SIP/${EXTEN})
>
> Then, I get TRYING and RINGING SIP messages which both include this kind of
> line :
> To: 
>
> Is it possible, configuring Asterisk 1.4, to get something like this
> instead ?
> To: "John Doe" 
>
> This way, I'm hoping to display callee's name beside (or instead of)
> callee's number which would offer a double check for caller which might be
> confusing extensions, for instance.
>
>
> I tried this :
> exten => _123X, 1, SIPAddHeader(To: Doe \ \;user=phone\>)
>
> but I still got :
> To: 
>
I must add I also tried without success :
exten => _123X, 1, Set(SIALPEERNAME=Doe)



>
>
> Regards
>
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Re: [asterisk-users] Asterisk 1.4.22 and 1.6.0 Released

[Stefan Gofferje]

Asterisk Development Team schrieb:

[Release info]

Did anyone notice bug #0013531 (http://bugs.digium.com/view.php?id=13531)?
It seems that the hold logic / MOH logic in chan_sip is somehow broken
in 1.6.0...

Terve,
Stefan

-- 
Last words of a stormchaser:
"Where is that rotation on the radar?!"


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[asterisk-users] How to add Callee's name into Dial command ?

[Olivier]

Hi,

When dialing a number, I use :
exten => _123X, 1, Dial (SIP/${EXTEN})

Then, I get TRYING and RINGING SIP messages which both include this kind of
line :
To: 

Is it possible, configuring Asterisk 1.4, to get something like this instead
?
To: "John Doe" 

This way, I'm hoping to display callee's name beside (or instead of)
callee's number which would offer a double check for caller which might be
confusing extensions, for instance.


I tried this :
exten => _123X, 1, SIPAddHeader(To: Doe \)

but I still got :
To: 

Regards
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Re: [asterisk-users] sip clients for smart phones?

[Babcock, Michael Alex]

windows smart phone v 6.0 example
htc shadow
is what i have. It has wifi abilitys.
mike

On Oct 3, 2008, at 12:11 AM, Alex Balashov wrote:

> What's a smart phone?
>
> Babcock, Michael Alex wrote:
>
>> hi;
>> curious does anyone know of a smart phone client that could connect  
>> to
>> asterisk?
>> thanks
>> Mike
>> thanks for reading
>> Systems administrator and owner of http://gwhosting.net
>> msn: [EMAIL PROTECTED]
>> twitter: http://twitter.com/creepyblindy
>>
>>
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>
>
> -- 
> Alex Balashov
> Evariste Systems
> Web: http://www.evaristesys.com/
> Tel: (+1) (678) 954-0670
> Direct : (+1) (678) 954-0671
> Mobile : (+1) (706) 338-8599
>
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thanks for reading
Systems administrator and owner of http://gwhosting.net
msn: [EMAIL PROTECTED]
twitter: http://twitter.com/creepyblindy


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Re: [asterisk-users] Improving the voice Quality,

[Alex Balashov]

Jai Rangi wrote:

> All,
> 
> I am having audio quality problem in some calls (1-2%) on asterisk. I 
> captured RTP traffic using ethereal and this is what I found with the 
> problematic calls. (Worst cases)
> Drop by Jitter buff: 25-75%
> Out of Seq: 50-100% (100 % means very very poor call quality).
> 
> Has anyone had similar problem? If yes, can you please share your 
> experience on how did you fix this? 

Such poor performance is not fixable.  The network, connectivity issues, 
machine load, etc. needs to be addressed - the underlying cause, in 
other words.

BTW, 100% out-of-sequence RTP packets leads to a lot more than just 
"very very poor call quality."  I don't see how the conversation could 
even be coherent in that situation.

What is more likely is that Wireshark's RTP stats are giving you some 
distorted information.  I've found its stream analysis to be somewhat 
buggy in that regard.

> I was wondering if I can decrease the MTU size to 250-500 on the network 
> card and use that card only for VoIP traffic. Will this have any bad 
> effect on sip traffic/packets?

No.  RTP packets are very small - much smaller than that MTU, or any 
reasonable MTU you could set.

-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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Re: [asterisk-users] sip clients for smart phones?

[Alex Balashov]

What's a smart phone?

Babcock, Michael Alex wrote:

> hi;
> curious does anyone know of a smart phone client that could connect to  
> asterisk?
> thanks
> Mike
> thanks for reading
> Systems administrator and owner of http://gwhosting.net
> msn: [EMAIL PROTECTED]
> twitter: http://twitter.com/creepyblindy
> 
> 
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-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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[asterisk-users] sip clients for smart phones?

[Babcock, Michael Alex]

hi;
curious does anyone know of a smart phone client that could connect to  
asterisk?
thanks
Mike
thanks for reading
Systems administrator and owner of http://gwhosting.net
msn: [EMAIL PROTECTED]
twitter: http://twitter.com/creepyblindy


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[asterisk-users] Improving the voice Quality,

[Jai Rangi]

All,

I am having audio quality problem in some calls (1-2%) on asterisk. I
captured RTP traffic using ethereal and this is what I found with the
problematic calls. (Worst cases)
Drop by Jitter buff: 25-75%
Out of Seq: 50-100% (100 % means very very poor call quality).

Has anyone had similar problem? If yes, can you please share your experience
on how did you fix this?

I was wondering if I can decrease the MTU size to 250-500 on the network
card and use that card only for VoIP traffic. Will this have any bad effect
on sip traffic/packets?

Any thoughts?


-Thank you,
-Jai
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[asterisk-users] 2 stage dialing and 484 address incomplete

[Olivier]

Hi,

If my memory serves me right, there was thread (in dev mailing list ?)
explaining how we could implement 2 stages dialing with SIP endpoints:
user dials 1234
then asterisk replies 484 Address Incomplete,
then user dials 5678
then asterisk begins to treat extension 12345678 as if it had been dialed as
a whole.

With compliant hardphones, you could get you phone to display a short text
invite between the series of digits.
This improves user experience, when consulting Voicemail, or asking features
that need a parameter to be set.

Is my memory correct ?
Has anyone a clue ?

Regards
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Re: [asterisk-users] t1 cards

[Eric Fort]

yes, more than 300 meters (longer than copper based ethernet allows).  Yes
to E1, as I understand it, it's just a config change on many cards anyway.
I'm specificly looking at pci based t1/e1 cards because I'm finding single
port cards on ebay going for 100-200 usd.  in some cases I may want to drive
a channel bank at the far end, thus t1/e1.  anyone have experience on how
far these pci based cards will drive when wired back to back?

Eric

On Thu, Oct 2, 2008 at 11:34 PM, Gordon Henderson <
[EMAIL PROTECTED] <[EMAIL PROTECTED]>> wrote:

> On Thu, 2 Oct 2008, Eric Fort wrote:
>
>  I presently need to connect a few channels of voice and data between
>> multiple locations where I own the copper between them.  Each location
>> exceeds 300M from any other location.  I'm thinking of generating T1's and
>> running those between locations.  If I use PC based cards wired back to
>> back
>> (I can do that, right?) what kind of distance can I expect to be able to
>> span without needing repeaters?  What inexpensive cards can you recommend
>> for use with asterisk?  I'm considering either digium or sangoma.  Would I
>> get any better performance if I used a sync-serial card connected to a
>> separate csu/dsu?
>>
>
> 300 metres, right? (not 300 miles?)
>
> Why stop at T1? Go for E1 :) with the right kit at each end you ought to be
> able to get 2Mb/sec or more. (distance depending)
>
> Personally, I'd go for a technology that gave me Ethernet at each end -
> then it makes it much easier to mix voice and data - But using something
> like a sync. modem and line driver then you need a media converter of some
> sorts at each end which might bump up the cost - at the savings of the E1
> card in the PC though. Last time I had bare copper to play with (a BT EPS8
> circuit) I had a 2Mb modem at each end going into a Cisco 2600 which was
> running CHDLC over the link and acting as nothing more than a dumb media
> converter to give me Ethernet at each end. This was 6 years ago though.
>
> Ah, Looks like the technology has improved somewhat:
>
>  http://www.blackbox.com/Catalog/Detail.aspx?cid=381,1452,1468&mid=5261
>
> From the UK site:
>
> Or even:
>
>  http://www.blackbox.com/Catalog/Detail.aspx?cid=425,1423,1424&mid=4946
>
> (same thing from the UK site:)
>
>
> http://www.blackbox.co.uk/solutions/display.asp?cs=dvh&id=1&doc=lb300a-r2&tx=LAN&sx=Network%20Appliances
>
> You need a pair, obviously...
>
> Hm. US site is $305, UK £253. Rip-off Britain again by the looks of it
>
> As for inexpensive cards - OpenVox. Their E1 cards seem to work OK, but if
> using a LAN extender, then they're not neeed at all...
>
> Gordon
>
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