Re: [Asterisk-Users] rxfax(spandsp-0.0.2pre18) and HT488

[Zen Kato]

I chaged to use from 'rxfax' to 'txfax' and I succeeded to receive the
file from * to the FAX under HT488(firmware 1.0.1.2).
My OS is 2.6.11-1.27_FC3smp, CVS-v1-0-04/20/05 with ztdummy.
spandsp is fun!

I made a file:
Channel: SIP/4881
MaxRetries: 0
WaitTime: 20
Application: txfax
Data: /usr/home/zenkato/voip/asterisk/fax/tif/receive.tif|caller
 
Then I copied this file to /var/spool/asterisk/outgoing.
The log is as follows;
*CLI>
-- Attempting call on SIP/4881 for application 
txfax(/home/zenkato/voip/asterisk/fax/tif/receive.tif|caller) (Retry 1)
   > Channel SIP/4881-baf5 was answered.
   > Lauching txfax(/home/zenkato/voip/asterisk/fax/tif/receive.tif|caller) 
on SIP/4881-baf5
The remote was made by 'Japan Electric'
The remote was made by 'Japan Electric'
DIS with final frame tag
In state 10
Start tx document
CFR with final frame tag
In state 4
Start tx page 0
Start tx page 1
MCF with final frame tag
In state 14
May 28 12:35:04 NOTICE[13118]: pbx_spool.c:239 attempt_thread: Call completed 
to SIP/4881
--
I can not find out why 'rxfax' does not work. It might stop after the
end of transmitting file from *. My FAX's LCD shows 'transmission error'.

I attached the log of 'rxfax'. Is 'ECM(Error Correction Mode)'supported
on spandsp-0.0.2pre18? This is Line 81 of the log. 

Regards,

Zen







txfax-test
Description: Binary data
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[Asterisk-Users] Re: Asterisk-Users Digest, Vol 10, Issue 221

[Nguyen Trung Tin]

Hello all.
 
How to compile chan_unicall.c
i have problem when compile chan_unicall.c, error message
please help
 
gcc -c -pipe  -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g  -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE  -O6 -march=i686  -DZAPTEL_OPTIMIZATIONS  -DASTERISK_VERSION=\"CVS-HEAD-05/28/05-06:39:38\" -DASTERISK_VERSION_NUM=99 -DINSTALL_PREFIX=\"\" -DASTETCDIR=\"/etc/asterisk\" -DASTLIBDIR=\"/usr/lib/asterisk\" -DASTVARLIBDIR=\"/var/lib/asterisk\" -DASTVARRUNDIR=\"/var/run/asterisk\" -DASTSPOOLDIR=\"/var/spool/asterisk\" -DASTLOGDIR=\"/var/log/asterisk\" -DASTCONFPATH=\"/etc/asterisk/asterisk.conf\" -DASTMODDIR=\"/usr/lib/asterisk/modules\" -DASTAGIDIR=\"/var/lib/asterisk/agi-bin\" -DBUSYDETECT_MARTIN -fomit-frame-pointer  -Wno-missing-prototypes -Wno-missing-declarations   -DZAPATA_PRI  -DIAX_TRUNKING   -DCRYPTO -fPIC  -o chan_unicall.o chan_unicall.cchan_unicall.c:24:34: asterisk/channel_pvt.h: No such file or
 directorychan_unicall.c:59:25: ../asterisk.h: No such file or directorychan_unicall.c: In function `unicall_digit':chan_unicall.c:623: dereferencing pointer to incomplete typechan_unicall.c: In function `unicall_call':chan_unicall.c:1127: dereferencing pointer to incomplete typechan_unicall.c: In function `unicall_hangup':chan_unicall.c:1319: dereferencing pointer to incomplete typechan_unicall.c:1524: dereferencing pointer to incomplete typechan_unicall.c: In function `unicall_answer':chan_unicall.c:1560: dereferencing pointer to incomplete typechan_unicall.c: In function `unicall_setoption':chan_unicall.c:1586: dereferencing pointer to incomplete typechan_unicall.c: In function `unicall_bridge':chan_unicall.c:1785: dereferencing pointer to incomplete typechan_unicall.c:1786: dereferencing pointer to incomplete typechan_unicall.c:1794: dereferencing pointer to incomplete typechan_unicall.c:1796: dereferenci
 ng
 pointer to incomplete typechan_unicall.c:1991: dereferencing pointer to incomplete typechan_unicall.c:1992: dereferencing pointer to incomplete typechan_unicall.c:2044: dereferencing pointer to incomplete typechan_unicall.c:2046: dereferencing pointer to incomplete typechan_unicall.c:2050: dereferencing pointer to incomplete typechan_unicall.c:2052: dereferencing pointer to incomplete typechan_unicall.c: In function `unicall_fixup':chan_unicall.c:2098: dereferencing pointer to incomplete typechan_unicall.c: In function `unicall_exception':chan_unicall.c:2132: dereferencing pointer to incomplete typechan_unicall.c: In function `unicall_read':chan_unicall.c:2206: dereferencing pointer to incomplete typechan_unicall.c:2251: dereferencing pointer to incomplete typechan_unicall.c:2394: dereferencing pointer to incomplete typechan_unicall.c:2400: dereferencing pointer to incomplete typechan_unicall.c: In function
 `unicall_write':chan_unicall.c:2542: dereferencing pointer to incomplete typechan_unicall.c: In function `unicall_indicate':chan_unicall.c:2614: dereferencing pointer to incomplete typechan_unicall.c: In function `unicall_new':chan_unicall.c:2727: dereferencing pointer to incomplete typechan_unicall.c:2729: dereferencing pointer to incomplete typechan_unicall.c:2754: dereferencing pointer to incomplete typechan_unicall.c:2755: dereferencing pointer to incomplete typechan_unicall.c:2756: dereferencing pointer to incomplete typechan_unicall.c:2757: dereferencing pointer to incomplete typechan_unicall.c:2758: dereferencing pointer to incomplete typechan_unicall.c:2759: dereferencing pointer to incomplete typechan_unicall.c:2760: dereferencing pointer to incomplete typechan_unicall.c:2761: dereferencing pointer to incomplete typechan_unicall.c:2762: dereferencing pointer to incomplete typechan_unicall.c:2763: derefe
 rencing
 pointer to incomplete typechan_unicall.c:2764: dereferencing pointer to incomplete typechan_unicall.c:2765: dereferencing pointer to incomplete typechan_unicall.c:2766: dereferencing pointer to incomplete typechan_unicall.c: In function `setup_unicall':chan_unicall.c:4556: warning: passing arg 1 of `ast_channel_register' from incompatible pointer typechan_unicall.c:4556: too many arguments to function `ast_channel_register'chan_unicall.c:4564: warning: passing arg 1 of `ast_channel_register' from incompatible pointer typechan_unicall.c:4564: too many arguments to function `ast_channel_register'chan_unicall.c: In function `unload_module':chan_unicall.c:4603: warning: passing arg 1 of `ast_channel_unregister' from incompatible pointer typechan_unicall.c:4604: warning: passing arg 1 of `ast_channel_unregister' from incompatible pointer typechan_unicall.c: In function `unicall_sendtext':chan_unicall.c:4666: dereferencing pointe
 r to
 incomplete typemake[1]: *** [chan_unicall.o] Error 1make[1]: Leaving directory `/usr/src/asterisk/channels'make: *** [subdirs] Error 1[EMAIL PROTECTED] asterisk]# 
 
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[Asterisk-Users] Remote Server IAX Configuration

[chawki hammoud]

Hi:

Please help me understand how a remote server with IP
address define a Asterisk client behind a nat.

Thanks. 



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[Asterisk-Users] How to Connect Netphone IP phone with ASterisk

[SYED ADEEL ALI]

I've configured SIP softphone to work with asterisk n it's working fine. but i m unable to connect my netphone IP phone I've connected my phone to LAN and assigned an IP address to it but how can i make call... plz tell me step wise.FREE pop-up blocking with the new MSN Toolbar MSN Toolbar Get it now!

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Re: [Asterisk-Users] VoiPSupply Dot Com: Epilogue

[Andrew Furey]

> Only if you have your clothes on and they don't... ;-)

I should _hope_ they don't have your clothes on :)

Andrew

-- 
Linux supports the notion of a command line or a shell for the same
reason that only children read books with only pictures in them.
Language, be it English or something else, is the only tool flexible
enough to accomplish a sufficiently broad range of tasks.
  -- Bill Garrett
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[Asterisk-Users] Asterisk vs pingtel?

[InternetMarketingMan2001]

Anyone know the differences between the two?






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Re: [Asterisk-Users] G729 vs. gsm

[Michael D Schelin]

Clue or clueless? Your call.

Steve Underwood wrote:


Michael D Schelin wrote:

Steve, you should really test the Codec and have G729 running as a 
pure IP to IP call you can not hear the difference on good networks! 



Well, it does to anyone without hearing damage. It sounds very obviously 
different.


Please do not get me wrong that G711u sounds better through the PSTN.  
Thats a given! You can't convert G729 up and down to G711 and expect 
the sound quality to be there. I'm a carrier and have tested G711 and 
G729 and have found that they both sound great through dedicated 
hardware.



This is meaningless drivel.


Asterisk's colors the G729 a little. Also my



The only time when Asterisk colours G.729 is when there is packet loss. 
Asterisk isn't handling that well.


hearing is fine. Please do not put down the comments of others in this 
forum. I'm stating my comments from my real world trials and this is 
not bad information.



Since it doesn't correlate with the impression of even the developers of 
G.729, it *is* bad information. Realistic people know G.729 will be 
worse. What they need is meaningful guidance as to just how much.


The man must compare codecs on his own and see what works for him. For 
me we've stuck with G711u because it's best through the PSTN. If I was 
running a pure IP to IP system I would use G729, Iblc, or GSM.



In a sane world pure IP to IP systems would't use G.711, G,729, iLBC, or 
GSM. They would be usign a wideband codec, as Skype does. Look at the 
favourable impression people have of that.


Next time, its probably better to argue with someone who hasn't spent 
time in speech codec development. We tend to have a clue what we are 
talking about. :-)


Regards,
Steve

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Re: [Asterisk-Users] Analog Telephone Adapter

[Guillermo Salas M.]

Joseph wrote:

I'll be trying AG-468  4 x FXS about 88.00USD from ATComm and let you
know when I get one (though it might be a while) 



Where yo purchase it?

--
Guillermo Salas M.
Telconet S.A. Manta
Calle 15 y Av. 24 Esq.
Phone : 593 5 262 8071
Mobile: 593 9 985 5138
e-mail: [EMAIL PROTECTED]
www   : http://www.telconet.net
http://www.telcocarrier.net

Linux User: 255902
Soporte en Linea en http://www.manta.telconet.net

Please avoid sending me Word or PowerPoint attachments.
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[Asterisk-Users] Changes on CVS HEAD

[Anton Krall]

I just installed the latest cvs head and seems a lot of commands haven been
depricated.

Where can I see the changes on all cvs head versions in order to keep up
with the changes needed on my side.

I checked the wiki and it still shows all the old commands and no mentions
about the changes.


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[Asterisk-Users] Polycom phones, UNREACHABLE

[Michael George]

I'm having some trouble with Polycom Soundpoint phones.  I have had good luck
deploying them on a local network, but now I've tried putting some in place
which access their * server across the network.

The * server is on a public IP and the polycoms are behind a NAT on a cable
modem broadband connection.

Every so often I get:
May 27 16:12:08 NOTICE[29728]: Peer 'Polycom1' is now UNREACHABLE!
May 27 16:31:54 NOTICE[29728]: Peer 'Polycom1' is now REACHABLE!

(Sometimes the first message says "TOO LAGGED"...)

And as you can see these messages are quite a ways apart, not just a few
seconds.

I have read the archives and found some clues that decreased the frequency of
the problem, but have not eliminated it.  My configuration for the phones in
sip.conf is:

defaultexpirey=3600 ; this is required by our VoIP provider rather than 120

[Polycom_1]
username=Polycom1
secret=
type=friend
canreinvite=no  ; specifically recommended in archives
nat=yes ; phone is behind a NAT
qualify=yes ; I suspected this might help...
host=dynamic
dtmfmode=rfc2833
context=internal
disallow=all
allow=ulaw

In the sip.cfg file for the phone on it's FTP server, I have set:
-server.1.address to the public address of the server
-voIpProt.SIP.outboundProxy.address to the public address of the server
-nat.ip is not set, as the description doesn't make it look like I want to
mess with it...
-there are other possible settings in that file that might be helpful, but
the descriptions are a bit thin in the manual...

I want to deploy more of these phones, but if they are ducking off the server
every so often, that makes them unreliable.

Does anyone have any ideas what the problem might be?

I think if I remove "qualify=yes" from sip.conf it will eliminate the warnings
in the log, but I think the phone will still be unreachable for that time
period and the problem is just less evident...

Thanks!

-- 
-M

There are 10 kinds of people in this world:
Those who can count in binary and those who cannot.
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Re: [Asterisk-Users] Re: Asterisk-Users Digest, Vol 10, Issue 114

[Steve Underwood]

Nguyen Trung Tin wrote:


Hello All
 
I need to use Asterisk with an E1 sangoma card with CAS R2 signalling 
for Vietnam
 
what is difference between libr2 of CVS and libmfc2 of soft-switch.org ?


libr2 is a piece of useless junk, which I have asked Digium to remove. 
The software at soft-switch.org works.


 
how to compile chan_unicall.c on asterisk. asterisk update CVS-head- 
May 27 2005.


Don't use it with CVS-head. It doesn't work with that right now. It 
works with the stable versions up to 1.0.7


Regards,
Steve

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[Asterisk-Users] Re: Asterisk-Users Digest, Vol 10, Issue 114

[Nguyen Trung Tin]

Hello All
 
I need to use Asterisk with an E1 sangoma card with CAS R2 signalling for Vietnam
 
what is difference between libr2 of CVS and libmfc2 of soft-switch.org ?
 
how to compile chan_unicall.c on asterisk. asterisk update CVS-head- May 27 2005.
 
 
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RE: [Asterisk-Users] Asterisk stopping to respond and CPU at the top

[Anton Krall]

I don't use mpg123, I use madplay 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Rich Adamson
|Sent: Viernes, 27 de Mayo de 2005 07:41 p.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] Asterisk stopping to respond and 
|CPU at the top
|
|
|> Guys.
|> 
|> Anybody having problems with asterisk taking all the cou and 
|then not 
|> responding anymore without a reboot?
|> 
|> How can I diagnose what asterisk is doing and why is it 
|taking all the cpu?
|> 
|> Hope you can help...
|
|9 times out of 10 its not asterisk, but mpg123.
|
|Try stopping asterisk, do a ps -ax and see if mpg123 is still running.
|If it is, kill all occurances and restart asterisk.
|
|
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[Asterisk-Users] you can bid on this very small project

[john mills]

http://www.rentacoder.com/RentACoder/misc/BidRequests/ShowBidRequest.asp?lngBidRequestId=288160
 
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Re: Fwd: [Asterisk-Users] Newbie here. Tips on setting up 100 phones wanted.

[Adam Goryachev]

The first problem I would consider is the reliability of the system.
Currently, I assume you are using standard analog lines stretched
between your various buildings, which means you 'only' rely on power at
the central location (which you say is generator provided) and the
physical cable survives. However, once you move to IP based phones, you
have switches, microwave, DSL, T1, etc, etc, etc... Plus power at each
location to power the phone. In such an environment, I'd be most
concerned about what might happen during some severe weather or other
extreme situation.

If the sites were closer together, then perhaps standard ethernet with
PoE would be a good solution...

See more..

On Fri, 2005-05-27 at 11:42 -0700, brandt Milczewski wrote:
> The current system is quite interesting.
> We have an office in a town that is about 50 miles from
> the ski area. The ski area is powered 100% of of generators and the
> telephone access and internet access goes from the Office in town out
> a private T1 to a town about 80 miles down another highway and then we

private T1 means dedicated point-to-point? ie, you 'rent' it from some
telco? Or, you ran your own cable, and installed your own equipment in
the other town, and just happen to use it as a T1??

> shoot a 5Mb microwave signal about 6 miles accross no mans land up to
> the ski area, (that oddly enough is on it's own highway with no
> utilities). 

So, the entire connection to your resort relies on this single 5MB
microwave link??

> From the top of the mountain I ran the signal over T1 to
> the Upper Ski lodge. 

Again, I assume you have installed your own equipment at the top of the
mountain and in the 'upper' ski lodge and run your own copper?

> From there I use an HDSL bridge about 3 miles
> down the hill to the lower lodge. 

So, if any ONE of the above fails, all comms to the lower lodge and
surrounding buildings are completely gone?

> Both lodges have a small size LAN
> about 25 computers. I also run some ethernet extender over copper to a
> couple other buildings and some WAP's these are all inside the
> employee lodge and one of the main day lodges.

I'm wondering, how do the power cables run, or does each 'area' have
it's own local generator, as opposed to a single large generator?

Perhaps you might look at adding some kind of redundancy, such as a
satellite link from the lower lodge. Meaning if you lose any one of the
above connections, you still have full external communications...

> As you can see it is a very unique and quite complex system as is. But
> the LAN is quite extensive and functions very well. As you can guess from
> the description the weak link is the Microwave from the top of the ski area
> down to the town with internet access. But that is very reliable and the
> snowfall we get, which is immense, hasn't been a problem for it yet.

That's great... hopefully it won't become a problem either...

> As for my technical background I build and admin servers and desktops
> in FreeBSD, Windows, and OSX. I learned routing and networking as
> needed for the job and look at this project as just more learning. And
> I'm VERY excited that I found an active community to query.

Well, as for my non-technical background, I've read plenty of novel's
where the basic theme is a group of people who lose all communication
with the rest of the world, and need to survive the elements/etc (plus
one or two people/animals/beasts intent on some evil plan) to make it
out alive...

Perhaps in the real world, that isn't such a likely story.

> Thanks again I hope that helps tell the story and background a bit more.

Of course, I've not had any experience in your environment, nor in
anything similar, so if it works for you, and you are happy with the
reliability, then go for it. (So long as you don't get someone
killed/injured and then get sued

Regards,
Adam
-- 
 -- 
Adam Goryachev
Website Managers
Ph:  +61 2 9345 4395[EMAIL PROTECTED]
Fax: +61 2 9345 4396www.websitemanagers.com.au

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RE: [Asterisk-Users] Survey: E1 prices

[Adam Vocks]

Not sure about E1's but we are paying $1200 per month for a PRI.  Get
this, for a PRI without Caller ID services its $950...

Free incoming calls and free local calling.

I think that is really high, but I don't think I have a whole lot of
choices here in central Illinois with Consolidated Communications...

Adam

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Leo Ann
Boon
Sent: Friday, May 27, 2005 8:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Survey: E1 prices

In .SG, the basic monthly recurring cost is S$360 (roughly US$200) 
excluding outgoing per minute charges (at S$0.014 peak - that's one 
point four cents). The one time installation charge is S$2000.

FYI.

X - Filter wrote:

> Hello List,
>
> I'd like to ask how much you guys pay for an E1 (30 voice lines) and 
> what company. You can email me personally and not the list.
>
> Best regards,
> Eddie
>
> _
> FREE pop-up blocking with the new MSN Toolbar - get it now! 
> http://toolbar.msn.click-url.com/go/onm00200415ave/direct/01/
>
> ___
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Re: [Asterisk-Users] CRM integration (was RE: CallerID)

[Adam Goryachev]

On Fri, 2005-05-27 at 20:44 +0200, Michiel van Baak wrote:
> This can be used on client pc's that are configured to allow
> external apps to be installed/executed.
> What if you are on a terminal in a hotel or fair or
> whatever? The solution I made is intended to be used in a
> webbased CRM/Groupware app with the filosofy that you will
> be totally independent of client hardware.

Could you use javascript, or java from within the browser, which is both
portable, and likely to work on ANY browser that way there is no
installation as such, just visit the page, and leave a browser window
open (minimised) which is 'listening' for connections ??

> I have to agree your way takes up less resources, but if you
> modify my agi script to write XML file instead of putting
> the data back in a DB, the load will be close to 0 (never
> seen a current webserver that cannot do less then 1000 xml
> file serves per second).

Sure, the webserver can serve up probably more than 1000 xml static
files per second, however there are at least two problems with this
method as I mentioned previously:

1) You have a delay between receiving the call and receiving the popup
on-screen (whether it is one second, or 5 seconds, there is a delay)
2) You are increasing network load Consider a multi-site
configuration, where you now have web access over the same network as
your voice
3) You have increased load on both client and server, since the client
must re-load the page each second.

At the end of the day, I'm sure we all agree that a push method is best.

> I just don't want to be dependent on the client's ability to
> run commands not installed.

Yes, I also agree (though not everyone will) that if a push method
requires additional software, then it isn't quite as useful in all
environments...

Personally, I don't know enough about all these scripting languages etc,
but if it is possible, then that would be wonderful :)

Regards,
Adam

-- 
 -- 
Adam Goryachev
Website Managers
Ph:  +61 2 9345 4395[EMAIL PROTECTED]
Fax: +61 2 9345 4396www.websitemanagers.com.au

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[Asterisk-Users] How to timeout using AGI.

[pbx]

How does one process / capture a timeout that has happened in using an AGI
script.. Preferably PHP.

I know you can set the wait timeout for a certain time, but how does the
script continue?

Thanks

Ben

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Re: [Asterisk-Users] G729 vs. gsm

[Steve Underwood]

Michael D Schelin wrote:

Steve, you should really test the Codec and have G729 running as a 
pure IP to IP call you can not hear the difference on good networks! 


Well, it does to anyone without hearing damage. It sounds very obviously 
different.


Please do not get me wrong that G711u sounds better through the PSTN.  
Thats a given! You can't convert G729 up and down to G711 and expect 
the sound quality to be there. I'm a carrier and have tested G711 and 
G729 and have found that they both sound great through dedicated hardware.


This is meaningless drivel.


Asterisk's colors the G729 a little. Also my


The only time when Asterisk colours G.729 is when there is packet loss. 
Asterisk isn't handling that well.


hearing is fine. Please do not put down the comments of others in this 
forum. I'm stating my comments from my real world trials and this is 
not bad information.


Since it doesn't correlate with the impression of even the developers of 
G.729, it *is* bad information. Realistic people know G.729 will be 
worse. What they need is meaningful guidance as to just how much.


The man must compare codecs on his own and see what works for him. For 
me we've stuck with G711u because it's best through the PSTN. If I was 
running a pure IP to IP system I would use G729, Iblc, or GSM.


In a sane world pure IP to IP systems would't use G.711, G,729, iLBC, or 
GSM. They would be usign a wideband codec, as Skype does. Look at the 
favourable impression people have of that.


Next time, its probably better to argue with someone who hasn't spent 
time in speech codec development. We tend to have a clue what we are 
talking about. :-)


Regards,
Steve

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Re: [Asterisk-Users] Survey: E1 prices

[Leo Ann Boon]

In .SG, the basic monthly recurring cost is S$360 (roughly US$200) 
excluding outgoing per minute charges (at S$0.014 peak - that's one 
point four cents). The one time installation charge is S$2000.


FYI.

X - Filter wrote:


Hello List,

I'd like to ask how much you guys pay for an E1 (30 voice lines) and 
what company. You can email me personally and not the list.


Best regards,
Eddie

_
FREE pop-up blocking with the new MSN Toolbar - get it now! 
http://toolbar.msn.click-url.com/go/onm00200415ave/direct/01/


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Re: [Asterisk-Users] pressing a key to get in of voicemail?

[Ing CIP Alejandro Celi Mariátegui]

Hi Moises, I made some changes, but the idea is the same. This is my
extensions.conf file, my extension is the 403 and I'm trying to
configure it:

[context]
...
[ask_voice_mail_no_answer]
exten => s,1,Background(presione_1_para_voicemail)
exten => 1,1,Voicemail(u${voicemail})
exten => 1,2,Hangup()


[default]

exten => s,1,Wait,1
exten => s,2,Answer
exten => s,3,DigitTimeout,5
exten => s,4,ResponseTimeout,10
exten => s,5,BackGround(vm-inicio)
exten => t,1,Dial(SIP/401,30,Tt) ; Default phone

; When I hear the voice, y press 403 keys and...

exten => 403,1,SetVar(voicemail=${ARG1})
exten => 403,2,Dial(SIP/403,10,Tt)
exten => 403,3,Wait,1
exten => 403,66,Goto(ask_voice_mail_no_answer,s,1)
exten => 403,103,Hangup


But don't work, logs say:

-- Starting simple switch on 'Zap/3-1'
-- Executing Wait("Zap/3-1", "1") in new stack
-- Executing Answer("Zap/3-1", "") in new stack
-- Executing DigitTimeout("Zap/3-1", "5") in new stack
-- Set Digit Timeout to 5
-- Executing ResponseTimeout("Zap/3-1", "10") in new stack
-- Set Response Timeout to 10
-- Executing BackGround("Zap/3-1", "vm-inicio") in new stack
-- Playing 'vm-inicio' (language 'en')
  == CDR updated on Zap/3-1
-- Executing SetVar("Zap/3-1", "voicemail=") in new stack
-- Executing Dial("Zap/3-1", "SIP/403|10|Tt") in new stack
-- Called 403
-- SIP/403-293c is ringing
-- Nobody picked up in 1 ms
  == Everyone is busy/congested at this time
-- Timeout on Zap/3-1
  == CDR updated on Zap/3-1
-- Executing Dial("Zap/3-1", "SIP/401|30|Tt") in new stack
-- Called 401
-- SIP/401-3ab3 is ringing
-- Hungup 'Zap/1-1'

Regards,

Alex


-- 
Ing CIP Alejandro Celi Mariátegui 
<[EMAIL PROTECTED]>



El vie, 27-05-2005 a las 10:19, Moises Silva escribió:
> may be something like this?
> 
> [macro-sipextens]
> exten => s,1,SetVar(voicemail=${ARG1})
> exten => s,2,Dial(SIP/${ARG1},40,r)
> exten => s,3,GotoIf($[${DIALSTATUS} = NOANSWER] ? 66  : 3)
> exten => s,4,GotoIf($[${DIALSTATUS} = BUSY] ? 68  : 4)
> exten => s,5,Playback(iss_invalid_sipexten)
> exten => s,6,Hangup()
> exten => s,66,Goto(ask_voice_mail_no_answer,s,1)
> exten => s,68,Goto(ask_voice_mail_busy,s,1)
> 
> the context ask_voice_mail_no_answer could be
> [ask_voice_mail_no_answer]
> exten => s,1,Background(press_1_for_voicemail)
> exten => 1,1,Voicemail(u${voicemail})
> exten => 1,2,Hangup()
> 
> regards
> 
> On 5/26/05, Ing CIP Alejandro Celi Mariátegui <[EMAIL PROTECTED]> wrote:
> > 
> > I've currently got Asterisk configured to take incoming calls, ask for
> > extension, ring the phone and send them directly to the voicemail.
> > 
> > What I want to be able to do is first a message "press 1 for voicemail
> > or hangup" before voicemail come up.
> > 
> > Any ideas?
> > 
> > regards,
> > 
> > --
> > Ing CIP Alejandro Celi Mariátegui
> > <[EMAIL PROTECTED]>
> > 
> > ___
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> > http://lists.digium.com/mailman/listinfo/asterisk-users
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> > 


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[Asterisk-Users] static linking

[Benjamin West]

Has anyone tried or had success statically linking Asterisk?  I'd like
to do this to deploy to smaller boxes that don't have the toolchain
and libraries to build the thing.

I've tried using statifier (at sf.net) which claims to take a dynamic
executable and transform it into a static executable. It seemed to
work on my box, but when I tried it on the target box it did this:
[EMAIL PROTECTED] /]# asterisk 
Illegal instruction
[EMAIL PROTECTED] /]# 

I've also tried messing with the Makefile.  Specifically I did this:
#ASTLINK=-Wl,-E 
#SOLINK=-shared -Xlinker -x
ASTLINK=-Wl,-E 
SOLINK=-static -Xlinker -x

I also tried leaving the ASTLINK line blank.  Both attempts were
unsuccessful and I got unresolved symbol errors.

Thanks,

Ben
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Re: [Asterisk-Users] Asterisk stopping to respond and CPU at the top

[Rich Adamson]

> Guys.
> 
> Anybody having problems with asterisk taking all the cou and then not
> responding anymore without a reboot?
> 
> How can I diagnose what asterisk is doing and why is it taking all the cpu?
> 
> Hope you can help...

9 times out of 10 its not asterisk, but mpg123.

Try stopping asterisk, do a ps -ax and see if mpg123 is still running.
If it is, kill all occurances and restart asterisk.


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Re: [Asterisk-Users] Soyo G688

[Isamar Maia]

Do you have any link? Isn't it PA-1688 Chip?

Isamar


On Fri, 27 May 2005, Waldo Rubinstein wrote:

> Has anyone had any experience with the Soyo G688 phone? I'd like to
> use it as a agent's phone. Is it reliable? How well does it work with
> *? How's the quality? Features?
>
> Thanks,
> Waldo
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[Asterisk-Users] Asterisk stopping to respond and CPU at the top

[Anton Krall]

Guys.

Anybody having problems with asterisk taking all the cou and then not
responding anymore without a reboot?

How can I diagnose what asterisk is doing and why is it taking all the cpu?

Hope you can help...

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Re: [Asterisk-Users] Asterisk con X-lite : Register Ok but no calls (404 Not found)

[Zoa]

They find the users but not the extension.

Have a look at
http://www.asteriskguru.com/tutorials/xlite_softphone.html for a
complete configuration guide.
http://www.asteriskguru.com/tutorials/softphones.html is the first page.

zoa


Romain Barrallon wrote:


Hi all,

I'm working on an implementation of VoIP en Linux.
I have a Debian Suse (*.*.*.173) with an * and a X-lite client and a
Red Hat 9.0 (*.*.*.172) with another softphone X-lite.
Both of the softphones are registering and appear in the peers (sip
show peers) with the good parameters of address and port.
If I try to make a call, * receive the INVITE request and send a 404
NOT FOUND answer.
I can't understand why asterisk doesn't found the users if they are registred...
It's making a "Scheduling Call Destruction".

My config files are :

sip.conf :
[general]



context=default; Default context for incoming calls
recordhistory=yes; Record SIP history by default
port=5060; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes

[]
;Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
;Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
type=friend
username=
secret=
callerid="Thibaud" <>
host=dynamic
context=from-sip
allow=ulaw
qualify=yes

[]
;Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
;Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
type=friend
username=
secret=
callerid="Florentin" <>
host=dynamic
context=from-sip
allow=ulaw
qualify=yes




extensions.conf :



[bogon-calls
exten => _.,1,Congestion

[from-sip]

exten => ,1,Dial(SIP/,20)
exten => ,2,Voicemail(u)
exten => ,102,Voicemail(b)
exten => ,103,Hangup

exten => ,1,Dial(SIP/,20)
exten => ,2,Voicemail(u)
exten => ,102,Voicemail(b)
exten => ,103,Hagup

exten => ,1,VoicemailMain(${CALLERIDNUM})





The critical SIP exchange is :

SEND TIME: 440651449
SEND >> *.*.*.173:5060
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP
*.*.*.172:5060;rport;branch=z9hG4bK5E6AA1A5168F2672CCC77E03310CA049
From: Asterisk ;tag=93980267
To: 
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 30470 INVITE
Proxy-Authorization: Digest
username="",realm="asterisk",nonce="2c887956",response="79eb7583cec4b45e867189dfa7d515dd",uri="sip:[EMAIL
 PROTECTED]"
Max-Forwards: 70
Content-Type: application/sdp
User-Agent: X-Lite release 1105d
Content-Length: 285

v=0
o= 440651420 440651437 IN IP4 *.*.*.172
s=X-Lite
c=IN IP4 *.*.*.172
t=0 0
m=audio 1 RTP/AVP 0 8 98 97 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

RECEIVE TIME: 440651467
RECEIVE << *.*.*.173:5060
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP *.*.*.172:5060;branch=z9hG4bK5E6AA1A5168F2672CCC77E03310CA049
From: Asterisk ;tag=93980267
To: ;tag=as6c9ced81
Call-ID: [EMAIL PROTECTED]
CSeq: 30470 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: 
Content-Length: 0


--
Romain Barrallon
- Etudiant en Télécommunications, Services et Usages à l'INSA de Lyon (France)
- Estudiante de intercambio en la Universidad Tecnica Federico Santa
Maria de Valparaíso (Chile)
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Description: OpenPGP digital signature
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Re: [Asterisk-Users] Wacko Distinctive Ring Patterns being detected??

[Gonzalo Servat]

On 5/28/05, Jay Milk <[EMAIL PROTECTED]> wrote:
> Could you configure your "normal ring" to be recognized as a distinctive
> ring and go into a different context?  That would essentially allow you
> to distinguish between the calls.

Excellent suggestion Jay! Thanks!! I changed the default context (if
no dring match) to go to the context I configured for this "second"
number and made a distinctive ring match on 0,0,0 to go to the main
number context, and works beatifully.

Thanks for taking the time to reply and for your suggestion!

Regards,
Gonzalo
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Re: [Asterisk-Users] SER Config For Asterisk

[Walter Willis]

+ o - va asi !!!


if (metho == INVITE ){
rewriteuriport(192.168.45.12: 5061)
forward(192.168.45.12,5061)
}
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Re: [Asterisk-Users] OH323 problem

[rafael . gonzalez]

Hi jeromy,

They are codec's problem?

How are you configured the files:
extensions.conf
oh323.conf
sip.conf
??

Rafael


Mensaje citado por Jeromy Grimmett <[EMAIL PROTECTED]>:

>
> May 28 01:58:32 WARNING[6821]: chan_sip.c:1830 sip_write: Asked to transmit
> frame type 256, while native formats is 4 (read/write = 4/4)
> May 28 01:58:33 WARNING[6821]: channel.c:2126 ast_channel_make_compatible:
> No path to translate from SIP/3901506-efd7(4) to OH323/L4592(256)
> May 28 01:58:33 WARNING[6821]: app_dial.c:1024 dial_exec: Had to drop call
> because I couldn't make SIP/3901506-efd7 compatible with OH323/L4592
>
>
> this is SIP>h323 gateway > TDM
>
> is anyone getting this??? any solutions?
>
> jeromy
>
>
>
>
>    Global reach, local touch...
>
>
> Jeromy Grimmett
> CEO   Comuniquémonos, Inc. / SA
> 1212 South Hampton Drive
>  =Alexandria%2C+LA+71301&country=us>
> Alexandria, LA 71301
> [EMAIL PROTECTED]
> IM: MSN: [EMAIL PROTECTED]
> http://www.comuniquemonos.com 
> tel:
> fax:
> mobile:   +593 (4) 287 3854
> (501) 646-0680
> +593 (9) 366 6521
>
>
>   Add me
> to your address book...  Want a 
> signature
> like this?
>
>





This message was sent using IMP, the Internet Messaging Program.

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Re: [Asterisk-Users] TDM400P vs SIP3000 x2

[Rich Adamson]

> I just got through trying to set up a Sipura 3000 and am still looking
> for answers.  There is a low volume problem (caller is underwater)  on
> the FXO port that I wish someone would have told me about and I would
> have gone the other route.  (even after upgrading firmware and
> adjusting gain settings)  More details here.
> 
> http://voxilla.com/index.php?name=PNphpBB2&file=viewtopic&t=3500&highlight=vol+volume
> http://voxilla.com/index.php?name=PNphpBB2&file=viewtopic&t=1249&highlight=vol+volume
> 
> Maybe they work great and this one is defective, but others appear to
> have a simular problem and this was my experience.

The good thing about the spa3k is it provides a substantial amount of
flexibility, and the bad thing is the flexibility (and options) make
it very difficult to configure without some fairly extensive experience
or knowledge.

I have one configured for pstn ring-thru to the fxs, but have both the
fxs port and the fxo port also registered with asterisk. (Eg, asterisk
can be totally down and pstn calls still ring thru, and my spouce didn't
need any remedial training to use the house phones lurking on the fxs
port.) With this config, the dialplan in the spa is set so that any
call starting with an "8" is routed to asterisk while all others are
sent out the pstn/fxo port. Love it.

With that config, under the Regional tab, I have fxs input=3 and
fxs output at 0. Then, on the PSTN tab, I have STA to PSTN at -2.0 and
PSTN to STA at -2.0. The end result is that non-technical fxs users
can't tell they are talking through the spa3k. (The audio levels are
just a little lower then if the spa3k wasn't in use.)

In addition, any incoming voip calls (via asterisk) ring the fxs port
with distinctive ringing, so we know from the ring where the call is
originating from. Audio via this path is no problem and depends solely
on the settings under the Regional tab.

If I want to place a call from Asterisk to the pstn thru the fxo port,
I can. The gain settings under the PSTN Line tab impact that call.
Those settings are 100% "dependent" on how far you are located from
the pstn central office, so copying other's settings will do you no
good whatsoever. We happen to be about 15,000 feet from our central
office and the measured cable loss is about -7db. In theory, I should 
be able to set the PSTN Line gains around 5.0 db (or about -2 db less 
then the cable loss), however if I do that echo is noticable. So, I 
had to back those values down -1.0 db at a time to find a value that
was an acceptable level and no noticable echo. (Since at least some
of the options within the spa3k require a reboot to take effect, I
simply rebooted the spa3k after making every change. At least that way,
I know for an absolute fact what values are actually being used
regardless of what has changed.)

There are also a couple of other settings on the spa3k that might be
impacting your quality. Obviously the port impedence needs to match
whatever your country standards are (600 ohms in the US), and I'd
make absolutely sure that is set before dicking with anything else.
On the SIP tab, look at the RTP Packet Size and ensure it says .020
(or 20 milliseconds). For whatever reason, my spa3k had a value of
.030 (30 milliseconds) and audio was not very good. I'm not sure if
that value was the default or if it somehow changed since the box
has had about five different firmware versions in it over the last
year or so. (Note: there was some discussion on the voxilla list about 
that as others were having audio quality issues as well.)

Last, on the Line 1 tab, be sure Silence Supp Enable = no, Preferred
Codec is g711u, and Symmetric RTP = yes. (Do the same on the PSTN tab.)

Given my 20+ years working for a telephone company in an engineer
role, I even find it difficult to configure the spa due to the number
of options and rather poor documentation on the majority of those
options. But, I'm fairly comfortable the spa _can_ be configured to
provide a very reasonable service in most cases. It does for us.

I have detected the spa3k will sometimes inject a little short-duration
noise and I'm pretty sure that certain voices are causing it, but I've
not spent the time to diagnose that as yet. Doesn't happen often enough
to warrant spending much time on it.

I also have a TDM04b card in the asterisk system with four pstn lines
attached to it. The quality of the audio and the functionality of both
are roughly the same. Setting the proper gains on the TDM card is no
easier or harder then setting the spa; both require an understanding
of the telco standards and playing with levels to find an acceptable
balance between levels and echo.

Rich


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RE: [Asterisk-Users] PRI "Actual-HookState" not showing offhookoninbound

[Ronald Hartmann]

SOLVED!

I removed the overlapdial=yes from the Zapata.conf

Issue is resolved.

When a call comes into the box, the PRI Channel "Actual Hook
State is set to OFF HOOK" and thus the echo cancellation routines work
as directed with the echocancel=XXX

Ron

PS> I will write up my experiences into wiki over the holiday weekend.


-Original Message-
From: Ronald Hartmann [mailto:[EMAIL PROTECTED] 
Sent: Friday, May 27, 2005 3:03 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] PRI "Actual-HookState" not showing
offhookoninbound

Stable Version 1.0.7

-Original Message-
From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED] 
Sent: Friday, May 27, 2005 11:36 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] PRI "Actual-HookState" not showing offhook
oninbound

On May 27, 2005 10:57 am, Ronald Hartmann wrote:
> Actual Hookstate: Onhook

What version of Asterisk is this?  CVS HEAD from about an hour ago does
not 
show hookstate for anything but FXS interfaces, and certainly not for
PRI.

-A.




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[Asterisk-Users] Asterisk con X-lite : Register Ok but no calls (404 Not found)

[Romain Barrallon]

Hi all,

I'm working on an implementation of VoIP en Linux.
I have a Debian Suse (*.*.*.173) with an * and a X-lite client and a
Red Hat 9.0 (*.*.*.172) with another softphone X-lite.
Both of the softphones are registering and appear in the peers (sip
show peers) with the good parameters of address and port.
If I try to make a call, * receive the INVITE request and send a 404
NOT FOUND answer.
I can't understand why asterisk doesn't found the users if they are registred...
It's making a "Scheduling Call Destruction".

My config files are :

sip.conf :
[general]
>>context=default; Default context for incoming calls
>>recordhistory=yes; Record SIP history by default
>>port=5060; UDP Port to bind to (SIP standard port is 5060)
>>bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all)
>>srvlookup=yes
>>
>>[]
>>;Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
>>;Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
>>type=friend
>>username=
>>secret=
>>callerid="Thibaud" <>
>>host=dynamic
>>context=from-sip
>>allow=ulaw
>>qualify=yes
>>
>>[]
>>;Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
>>;Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
>>type=friend
>>username=
>>secret=
>>callerid="Florentin" <>
>>host=dynamic
>>context=from-sip
>>allow=ulaw
>>qualify=yes

extensions.conf :
>>[bogon-calls
>>exten => _.,1,Congestion
>>
>>[from-sip]
>>
>>exten => ,1,Dial(SIP/,20)
>>exten => ,2,Voicemail(u)
>>exten => ,102,Voicemail(b)
>>exten => ,103,Hangup
>>
>>exten => ,1,Dial(SIP/,20)
>>exten => ,2,Voicemail(u)
>>exten => ,102,Voicemail(b)
>>exten => ,103,Hagup
>>
>>exten => ,1,VoicemailMain(${CALLERIDNUM})


The critical SIP exchange is :

SEND TIME: 440651449
SEND >> *.*.*.173:5060
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP
*.*.*.172:5060;rport;branch=z9hG4bK5E6AA1A5168F2672CCC77E03310CA049
From: Asterisk ;tag=93980267
To: 
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 30470 INVITE
Proxy-Authorization: Digest
username="",realm="asterisk",nonce="2c887956",response="79eb7583cec4b45e867189dfa7d515dd",uri="sip:[EMAIL
 PROTECTED]"
Max-Forwards: 70
Content-Type: application/sdp
User-Agent: X-Lite release 1105d
Content-Length: 285

v=0
o= 440651420 440651437 IN IP4 *.*.*.172
s=X-Lite
c=IN IP4 *.*.*.172
t=0 0
m=audio 1 RTP/AVP 0 8 98 97 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

RECEIVE TIME: 440651467
RECEIVE << *.*.*.173:5060
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP *.*.*.172:5060;branch=z9hG4bK5E6AA1A5168F2672CCC77E03310CA049
From: Asterisk ;tag=93980267
To: ;tag=as6c9ced81
Call-ID: [EMAIL PROTECTED]
CSeq: 30470 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: 
Content-Length: 0


--
Romain Barrallon
- Etudiant en Télécommunications, Services et Usages à l'INSA de Lyon (France)
- Estudiante de intercambio en la Universidad Tecnica Federico Santa
Maria de Valparaíso (Chile)
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RE: [Asterisk-Users] Wacko Distinctive Ring Patterns being detected??

[Jay Milk]

Could you configure your "normal ring" to be recognized as a distinctive
ring and go into a different context?  That would essentially allow you
to distinguish between the calls.

> -Original Message-
> From: Gonzalo Servat [mailto:[EMAIL PROTECTED] 
> Sent: Friday, May 27, 2005 3:34 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Wacko Distinctive Ring Patterns 
> being detected??
> 
> 
> Hi All,
> 
> I've recently got a "second" number installed on my PSTN 
> line, trusting the Asterisk distinctive ring detection would 
> work as expected. It appeared to work fine at the start, as 
> the second number generated a different ring pattern to 0,0,0 
> (in the console) only to realise that almost every phone call 
> to this "second" number generated a different ring pattern. 
> Sometimes it might detect the same 2 patterns, but this is a 
> rare case. If Asterisk allowed me to configure up to 10 
> ringing patterns, I could probably cover most of the ringing 
> patterns being detected, but unfortunately there is a limit 
> of 3 which means 50% (or more) of the calls are coming in 
> under a distinctive ring pattern not configured in Asterisk, 
> and hence going to the default context.
> 
> Does anyone have any suggestions/ideas/etc on how to resolve 
> this issue?
> 
> Thanks in advance guys.
> Gonzalo
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[Asterisk-Users] Switch from NBX to Asterisk

[Adam Vocks]

We've been talking about
moving from our NBX100 to an asterisk solution for some time now.

 

Well, to make a long
story short, the call processor on our NBX has went south.  We are now
forced to purchase a new call processor >$500 or move our solution to asterisk. 
(Which is up and running and have been tested with Grandstream Adapters.)

 

My question:  Is
there anyone out there that have moved from NBX to Asterisk, and what are the
differences for your users, hold, transferring calls, etc.  Also, what
features will we loose, if any.  And do you recommend analog phones with
sip adapters (Which we already have.) or making an investment in IP phones?

 

Our setup:

NBX had 4 analog lines
coming into it with 14 phones.

Asterisk is set up
running 1.0.7 with a PRI and 100 DID numbers.  We have 20 or so Handytone
286 Adapters as well.

 

Any advice would be
appreciated!

 

Adam

 

 






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RE: [Asterisk-Users] G729 vs. gsm

[Huddleston, Robert]

We too are a carrier / clec and our lucent iMerge for the PTSN is all
g711-ulaw

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael D
Schelin
Sent: Friday, May 27, 2005 3:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] G729 vs. gsm

Steve, you should really test the Codec and have G729 running as a pure 
IP to IP call you can not hear the difference on good networks! Please 
do not get me wrong that G711u sounds better through the PSTN.  Thats a 
given! You can't convert G729 up and down to G711 and expect the sound 
quality to be there. I'm a carrier and have tested G711 and G729 and 
have found that they both sound great through dedicated hardware. 
Asterisk's colors the G729 a little. Also my hearing is fine. Please do 
not put down the comments of others in this forum. I'm stating my 
comments from my real world trials and this is not bad information. The 
man must compare codecs on his own and see what works for him. For me 
we've stuck with G711u because it's best through the PSTN. If I was 
running a pure IP to IP system I would use G729, Iblc, or GSM.

Mike


Steve Underwood wrote:
> Michael D Schelin wrote:
> 
>> I have used G729 and it sounds almost as good as G711U. The problem is 
>> the way Asterisk uses it. It does not sound robotic and it's not 
>> suppose  to sound that way. Most Carriers want the calls to be in 
>> g711u so thats why I use G711u otherwise I want to save money on 
>> bandwidth. G729 on Asterisk adds latency. this could be one of your 
>> problems. Also you will not get music on hold to play well with G729.
> 
> 
> G.729 doesn't sound that bad. However, if you find it hard to tell G.729 
> from G.711, I think you should have your hearing checked. :-) It really 
> doesn't help people to assess what is right for them, if other people 
> make these exaggerated and unreasonable claims. Even the people 
> promoting G.729 give it a MOS far below G.711. You are certainly right 
> about music on hold, but even voice plus a little background noise can 
> sound bloody awful through G.729. Its performance is *very* dependant on 
> compressing just a single human voice.
> 
> Regards,
> Steve
> 
> ___
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> 

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[Asterisk-Users] Wacko Distinctive Ring Patterns being detected??

[Gonzalo Servat]

Hi All,

I've recently got a "second" number installed on my PSTN line,
trusting the Asterisk distinctive ring detection would work as
expected. It appeared to work fine at the start, as the second number
generated a different ring pattern to 0,0,0 (in the console) only to
realise that almost every phone call to this "second" number generated
a different ring pattern. Sometimes it might detect the same 2
patterns, but this is a rare case.
If Asterisk allowed me to configure up to 10 ringing patterns, I could
probably cover most of the ringing patterns being detected, but
unfortunately there is a limit of 3 which means 50% (or more) of the
calls are coming in under a distinctive ring pattern not configured in
Asterisk, and hence going to the default context.

Does anyone have any suggestions/ideas/etc on how to resolve this issue?

Thanks in advance guys.
Gonzalo
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[Asterisk-Users] DVG-1120S does not show callerid Name and resets time

[Ryan Laginski]

Hi,
I am having problems with callerid name and the time with my
dvg-1120S. Every time I receive a call, it reverts the phone to
January 1st 12:00am. I've looked everywhere in the browser and telnet
configuration to change this.

Also, it never shows the name of the caller. I've even tried forcing
the caller id info using SetCallerID, but it still doesn't show the
name.

Any suggestions?
Thanks,
-Ryan
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RE: [Asterisk-Users] VoiPSupply Dot Com: Epilogue

[Robert Webb]

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Scott
Wolfe
Sent: Friday, May 27, 2005 4:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] VoiPSupply Dot Com: Epilogue

Maybe I should my pictures in with me and supermodels. :-)

Cheers,
   -Scott




Only if you have your clothes on and they don't... ;-)



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Re: [Asterisk-Users] VoiPSupply Dot Com: Epilogue

[Scott Wolfe]

Maybe I should my pictures in with me and supermodels. :-)

Cheers,
   -Scott

- Original Message - 
From: "Wiley Siler" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Sent: Friday, May 27, 2005 12:26 PM
Subject: RE: [Asterisk-Users] VoiPSupply Dot Com: Epilogue


LOL - You mean he actually 'met' Newt Gingrich?  How dare you not extend
him credit!!!
I mean seriously... For such a distinguished individual...

Hey, not only have I met the heads of several multi-billion dollar
corps, I have gotten absolutely blasted drunk with them.
So I should get credit, a 40% discount, and your daughters phone number,
right???  LOL

Seriously, though.  I think it is a sigh of relief that this hopefully
will be all over and off the list.
I for one have seen enough positive comments to know that your company
is a quality player.
The fact that you have followed up with the community and been so
forthright also says a lot.
Mistakes happen.  Sometimes people get inconvenienced.  The quality
companies address the issue and fix it as best they can.
I don't think we can ask for much more than that.  Keep up the good work
and keep that pricing aggressive...  8)

Cheers,
Wiley Siler
Who has been drunk with "important" people





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RE: [Asterisk-Users] Newbie here. Tips on setting up 100 phones w anted.

[Rich Adamson]

> >It will be about 100 phones at about 20 locations all within
> >about 4 miles of each other.
> 
> Perhaps a more pressing question might be how you are going to backhaul
> Ethernet in a 4-mile radius. You can't run a Cat 5 cable more than 100
> metres reliably, and using Ethernet repeaters every hundred metres or so
> isn't practical. You will need a fiber backbone or something like that. What
> is your plan to create an Ethernet network to tie these locations together? 

Or any of lots of different wireless facilities. We work with
multiple isp's that have customers ten miles or more and voip is
no problem for most of them.


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[Asterisk-Users] sip phone behind nat connecting to an asterisk box that has one port on the open internet

[Lance Grover]

Hello all,
  I have an asterisk box on the open internet (one port). I can get
sip clients from behind a nated network to register with this box and
make phone calls, but when the other person picks up they can hear me
just fine but I cannot hear them at all.  Is there anyone that can
help me get this working.  Right now I am open to any type of sip
phones soft or hard to work so I can then narrow it down from there. 
The reason for this is that I have a friend that has a mac who needs
to connect to this asterisk box and the iaxcom application that speaks
IAX is a little bit unstable on the mac.  I guess if someone knows of
a good IAX soft phone for mac then that would be another option.

-- 
Thanks,

Lance Grover
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RE: [Asterisk-Users] Newbie here. Tips on setting up 100 phones wanted.

[Rusty Shackleford]

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> brandt Milczewski
> Sent: Friday, May 27, 2005 10:00 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] Newbie here. Tips on setting up 100 
> phones wanted.
> 
> 
> I'm looking at setting up Asterisk for a completely IP 
> environment. All intercompany calls.
> 
> I work for a ski area. I currently use a 3Com Superstack for 
> in our office. And an old small town phone system for up at 
> the mountain. The phone system is dying and I'm hoping to 
> bring IP to replace the old phones. It will be about 100 
> phones at about 20 locations all within about 4 miles of each other.

With a run of 4 miles, it’s a safe bet that some segments will be (as
others have pointed out) beyond the 100 meter limit of ethernet. In
practice, under optimal conditions, you can fudge this a bit (sometimes
a lot) but I wouldn't count on it.

For the longer hauls, you might want to consider point-to-point DSL,
using something like this
 http://www.paradyne.com/products/SNE2000/
Conceivably, if you own the copper, you could do anything you want with
it.

-- 
Internal Virus Database is out-of-date.
Checked by AVG Anti-Virus.
Version: 7.0.322 / Virus Database: 266.11.14 - Release Date: 05/20/2005
 

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[Asterisk-Users] SIP REFER: Trying again

[Hendrik Magilsen]

Still having trouble, I even tried to remove it from the
allowed methods, but the other end still sent a REFER anyway which asterisk
accepts but kills the call anyway.

 

Asterisk is being used in a back-to-back sip arrangement to
take advantage of its IVR and ACD capabilities.   So, inbound call from
SIP provider in to asterisk,  runs through IVR and ACD perfectly and then
when ready leaves asterisk back to the SIP provider.  It all works well
until the end user transfers or moves to a conference bridge outside of
asterisk.  The Provider sends a REFER back to the asterisk advising that
the endpoint is going to move.  Asterisk looks up the info in it’s
extension list (it isn’t there of course) and kills the call.

 

All of the required information is contained in the REFER,
but does not appear to be used by asterisk.

 

Is there anyway around this?  Can the asterisk be
forced to respond with method not allowed (in which case another method must be
used).  Or is there another way to manage REFER methods that I haven’t
found yet.  Is the REFER dealt with any differently in the CVS load (I’m
using the standard version off the website 1.0.7)?

 

I do have a workaround but it’s ugly and I’d
rather solve this problem at source.

 

This is the second post for this issue and am hoping for
some insight.

 

Thanks

Hendrik






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RE: [Asterisk-Users] Newbie here. Tips on setting up 100 phones w anted.

[Colin Anderson]

Cool. Sounds like you've got your poop in a group already. As to your
original question, I am partial to the Snom 190 phones, they are easy to set
up, look and perform great, users really like them, and they seem quite
tough. I have two of them running in a shop environment where they get
covered in sawdust every day and all the users do is blow the sawdust off
and dial. Working for 5 months so far no problems touch wood. 

hth
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RE: [Asterisk-Users] VoiPSupply Dot Com: Epilogue

[Wiley Siler]

LOL - You mean he actually 'met' Newt Gingrich?  How dare you not extend
him credit!!!
I mean seriously... For such a distinguished individual...

Hey, not only have I met the heads of several multi-billion dollar
corps, I have gotten absolutely blasted drunk with them.  
So I should get credit, a 40% discount, and your daughters phone number,
right???  LOL

Seriously, though.  I think it is a sigh of relief that this hopefully
will be all over and off the list.
I for one have seen enough positive comments to know that your company
is a quality player.
The fact that you have followed up with the community and been so
forthright also says a lot.
Mistakes happen.  Sometimes people get inconvenienced.  The quality
companies address the issue and fix it as best they can.
I don't think we can ask for much more than that.  Keep up the good work
and keep that pricing aggressive...  8)

Cheers,
Wiley Siler
Who has been drunk with "important" people





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[Asterisk-Users] Soyo G688

[Waldo Rubinstein]

Has anyone had any experience with the Soyo G688 phone? I'd like to  
use it as a agent's phone. Is it reliable? How well does it work with  
*? How's the quality? Features?


Thanks,
Waldo
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Re: [Asterisk-Users] G729 vs. gsm

[Michael D Schelin]

Steve, you should really test the Codec and have G729 running as a pure 
IP to IP call you can not hear the difference on good networks! Please 
do not get me wrong that G711u sounds better through the PSTN.  Thats a 
given! You can't convert G729 up and down to G711 and expect the sound 
quality to be there. I'm a carrier and have tested G711 and G729 and 
have found that they both sound great through dedicated hardware. 
Asterisk's colors the G729 a little. Also my hearing is fine. Please do 
not put down the comments of others in this forum. I'm stating my 
comments from my real world trials and this is not bad information. The 
man must compare codecs on his own and see what works for him. For me 
we've stuck with G711u because it's best through the PSTN. If I was 
running a pure IP to IP system I would use G729, Iblc, or GSM.


Mike


Steve Underwood wrote:

Michael D Schelin wrote:

I have used G729 and it sounds almost as good as G711U. The problem is 
the way Asterisk uses it. It does not sound robotic and it's not 
suppose  to sound that way. Most Carriers want the calls to be in 
g711u so thats why I use G711u otherwise I want to save money on 
bandwidth. G729 on Asterisk adds latency. this could be one of your 
problems. Also you will not get music on hold to play well with G729.



G.729 doesn't sound that bad. However, if you find it hard to tell G.729 
from G.711, I think you should have your hearing checked. :-) It really 
doesn't help people to assess what is right for them, if other people 
make these exaggerated and unreasonable claims. Even the people 
promoting G.729 give it a MOS far below G.711. You are certainly right 
about music on hold, but even voice plus a little background noise can 
sound bloody awful through G.729. Its performance is *very* dependant on 
compressing just a single human voice.


Regards,
Steve

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RE: [Asterisk-Users] PRI "Actual-HookState" not showing offhook oninbound

[Ronald Hartmann]

Stable Version 1.0.7

-Original Message-
From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED] 
Sent: Friday, May 27, 2005 11:36 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] PRI "Actual-HookState" not showing offhook
oninbound

On May 27, 2005 10:57 am, Ronald Hartmann wrote:
> Actual Hookstate: Onhook

What version of Asterisk is this?  CVS HEAD from about an hour ago does
not 
show hookstate for anything but FXS interfaces, and certainly not for
PRI.

-A.


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Re: [Asterisk-Users] Unable to create channel of type 'Zap' with zaphfc driver

[Aitor]

Yes I have group=1. Also I've try with 
exten => 203,1,Dial,Zap/1/number 
and always the same message:
May 27 20:56:20 NOTICE[1041]: app_dial.c:759 dial_exec: Unable to 
create 
channel of type 'Zap'
  == Everyone is busy/congested at this time
I think that the problem is not in configuration... I have this message in the 
syslog:
May 27 20:56:25 localhost kernel: zaphfc: bchan rx fifo not enough 
bytes to 
receive! (z1=8191, z2=8191, wanted 8 got 0), probably a buffer overrun.
Hundred of times repeated... It seems as the line wasnt connected, no?
Thanks a lot.
El Viernes 27 Mayo 2005 19:31, Jean-Christophe Heger escribió:
> exten => 203,1,Dial,Zap/1/
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Re: [Asterisk-Users] VoiPSupply Dot Com: Epilogue

[Cory Andrews]

I promise this will be the last chapter in the story, at least from my 
end.  If we could only get George Lucas to say the same thing about the 
godawful new Star Wars movies he keeps cranking out.


I did some research this morning, interviewed the involved parties on 
our end, and have come to the following conclusions.


We did indeed make an error on our end.  When the sales rep received the 
purchase order from Ken-Ton, he failed to notice they had supplied an 
alternate shipping address, and the order was put in with their default 
shipping address on file.  By the time the error was realized, perhaps a 
day's delay was incurred, and then we moved swiftly to correct the 
error.  All through this process, Darren claims to have been in constant 
contact with Ken-Ton Electronics to advise them of the situation.


Even in lieu of our poor performance, the customer did see fit to 
purchase additional equipment from us.  On their next order, Darren (the 
sales rep who committed the clerical error on the mis-shipped earlier 
order) took it upon himself to hand deliver the equipment to the 
customer site, and to offer his personal apology for the less than 
perfect service the first go-round.


In the course of Darren's conversation with the customer, the subject of 
credit terms arose.  Darren relayed to the customer that we generally do 
not offer terms to our customers, which is, for the most part, a true 
statement.  I suspect that the principals at Ken-Ton Electronics may 
have taken personal offense to our broad policy of not offering credit, 
and may in fact have concluded that we did not find them worthy of credit.


Unfortunately, due to the volatile nature of technology business, it is 
exceedingly difficult for us to make sound decision regarding the 
extension of credit to customers.  We are a quite large company, with 
annual sales for 2005 projected to approach $30 million in hardware 
sales, but at the same time we are privately funded and do not rely at 
this point on outside investment or venture capital.  Having written 
down signifigant bad debt at the end of 2004 from customers who had 
stellar D&B scores but still managed to find ways to bankrupt their 
companies, we made a decision only to extend credit to very large 
accounts, and only those we felt were completely risk free.  We knew we 
would perhaps lose or alienate some customers in the transition, but the 
long term pros far outweighed the short term cons.


In summary, we are still a bit amazed that the situation resulted in 
such a lengthy, public diatribe from Mr. Vesterling. 

Karl is and important and very busy fellow, and he has taken the liberty 
of sending me some photos of himself hobnobbing with other very 
important and very busy people, reminding me "*_Look...  You're in my 
hometown, and I'm not your average Buffalo Ambassador:"


_*I'm not sure what the point is there either, but here they are for 
your viewing pleasure.


Karl with former house speaker Newt Gingrich 
http://www.voipsupply.com/images/karl1.jpg


Karl with luminary Jack Quinn
http://www.voipsupply.com/images/karl2.jpg

We realize that time is money, and we will continue to improve our 
process with the ultimate goal of 100% customer satisfaction. 


Cory Andrews
Senior Partner
VOIPSupply.com
+
V: 800.398.VOIP X22
F: 716.630.1548
E: [EMAIL PROTECTED]



Karl J. Vesterling wrote:


At 08:59 AM 5/27/2005, you wrote:


[ snip for brevity ]
I just wanted to clarify ... this isn't a voipsupply.com problem at 
all, but
rather a courier screwup... which happens anywhere and at anytime... 
right?



TWO screw ups in the shipment.
1.) It was shipped to the Bill-To address.  Since there is no one 
there during the day I had to sit and wait for it lest it not be 
delivered.


2.) when an order is placed on a Tuesday AM (or) Monday PM, and it's 
priority overnight, and it's across town, and the tracking number was 
supplied on Wednesday one would expect that it would show up Thursday, 
not Friday.


So, what we have here is one problem compounded by another, none on 
behalf of the courier.



-A.
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Best Regards,
Karl J. Vesterling
*E-Mail:* [EMAIL PROTECTED]


*Telephone:
Washington DC:* (202) 448-3009 Extension 0
*Annapolis MD:* (240) 524-6706 Extension 0
*Bethesda MD:* (301) 576-3014 Extension 0
*Niagara Falls NY:* (716) 286-9175 Extension 0
*Buffalo NY:* (716) 608-1121 Extension 0

*Yahoo Messenger:* karl_vesterling
*ICQ: *1548052
*AOL Instant Messenger:* n2vqm


---

[Asterisk-Users] asterisk and nortel CS1000 using SIP

[Jerry Geis]

Sorry wrong model CS1000 not BS1000. typo

Thanks,

Jerry

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[Asterisk-Users] asterisk and nortel BS1000 using SIP

[Jerry Geis]

Any WORKING uses of asterisk connected to a nortel BS1000 using SIP?

I can setup asterisk fine. I have never seen a BS1000 the Nortel guy is not
being able to set it up? Anyone have setup steps?

Thanks,

Jerry

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RE: [Asterisk-Users] Upgraded firmware on Polycom 500, digit-order problems

[Wiley Siler]

Wow, I am pretty sure you should update both the bootrom 
and sip.ld at the same time.
I would do this with at least one phone and see what 
happens.
 
W
 


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Chris 
CoulthurstSent: Friday, May 27, 2005 11:37 AMTo: 
asterisk-users@lists.digium.comSubject: [Asterisk-Users] Upgraded 
firmware on Polycom 500,digit-order problems


Ever since I upgraded my Polycom 500 
to the newest sip.ld (kept the old bootrom), when I dial things like “*98” for 
voicemail, the screen shows “9*8” and doesn’t dial my voicemail system!  Is 
this user error, or errors in the new firmware?
 
Chris Coulthurst
[EMAIL PROTECTED]
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Re: [Asterisk-Users] CRM integration (was RE: CallerID)

[Michiel van Baak]

On 09:01, Fri 27 May 05, Max W Blackmer Jr wrote:
> hello everyone,
> 
> I just had a thought on this subject why not create a daemon process on
> the Client PC That registers its self and What phone the user is
> connected. An AGI script could monitor the progress and when answered
> could send a push to the registered daemon which would push a link to
> the registered daemon on the telephone operators on the desk top. this
> would not waist resource as much as polling?  perhaps the daemon could
> be written in something portable like java or even as a small applet
> that is launched first and minimized to launch the CRM App in a browser
> and could be written into other CRM Applications.
> 
> What are your thoughts on this?
> 
> Max W. Blackmer Jr.

This can be used on client pc's that are configured to allow
external apps to be installed/executed.
What if you are on a terminal in a hotel or fair or
whatever? The solution I made is intended to be used in a
webbased CRM/Groupware app with the filosofy that you will
be totally independent of client hardware.
I have to agree your way takes up less resources, but if you
modify my agi script to write XML file instead of putting
the data back in a DB, the load will be close to 0 (never
seen a current webserver that cannot do less then 1000 xml
file serves per second).
I just don't want to be dependent on the client's ability to
run commands not installed.
-- 
Michiel van Baak
http://lunteren.vanbaak.info
[EMAIL PROTECTED]
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D

"Two of the most famous products of Berkeley are LSD and BSD. I don't think 
that this is a coincidence."

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Fwd: [Asterisk-Users] Newbie here. Tips on setting up 100 phones wanted.

[brandt Milczewski]

So in order to answer the background and backbone questions here is
the system as it is. I hope this isn't too much for the list but I'll
post it in response to a few inquiries.


The current system is quite interesting.
We have an office in a town that is about 50 miles from
the ski area. The ski area is powered 100% of of generators and the
telephone access and internet access goes from the Office in town out
a private T1 to a town about 80 miles down another highway and then we
shoot a 5Mb microwave signal about 6 miles accross no mans land up to
the ski area, (that oddly enough is on it's own highway with no
utilities). From the top of the mountain I ran the signal over T1 to
the Upper Ski lodge. From there I use an HDSL bridge about 3 miles
down the hill to the lower lodge. Both lodges have a small size LAN
about 25 computers. I also run some ethernet extender over copper to a
couple other buildings and some WAP's these are all inside the
employee lodge and one of the main day lodges.

As you can see it is a very unique and quite complex system as is. But
the LAN is quite extensive and functions very well. As you can guess from
the description the weak link is the Microwave from the top of the ski area
down to the town with internet access. But that is very reliable and the
snowfall we get, which is immense, hasn't been a problem for it yet.

As for my technical background I build and admin servers and desktops
in FreeBSD, Windows, and OSX. I learned routing and networking as
needed for the job and look at this project as just more learning. And
I'm VERY excited that I found an active community to query.

Thanks again I hope that helps tell the story and background a bit more.

-brandt

On 5/27/05, Wiley Siler <[EMAIL PROTECTED]> wrote:
> OK.  Here are some pointers
>
> Be sure to read up here...
> www.voip-info.org
>
> Lots of info and very specific to SIP, Asterisk, etc, etc...
> Also, with SIP, firewalls become an issue.  Check out the SIP stuff at
> above URL.
>
> Next, you will need to make sure you have some redundancy for your
> Asterisk box.
> Good news is that you can plan on using pretty simple dual processor
> boxes (Supermicro highly reccommended for compat)
>
> Next, how do you intend to connect to the PSTN?
>
> Just so you know, this is going to be a HUGE project form the sounds of
> it...
> I have a friend who owns a company in Denver that provides a hosted PBX
> solution if you are interested.
> They provide the Polycom IP500 phones with the accounts.
> www.unitybn.com  Ask for Greg Mennard and reference me...
> Have to throw that bone out there...
>
> What is your technical background?
>
> Thanks,
> Wiley
>
>
>
> -Original Message-
> From: brandt Milczewski [mailto:[EMAIL PROTECTED]
> Sent: Friday, May 27, 2005 10:45 AM
> To: Wiley Siler
> Subject: Re: [Asterisk-Users] Newbie here. Tips on setting up 100 phones
> wanted.
>
> hehe I hear this.. Fiber doesn't exist currently but is part of phase
> one infrastructure buildup. The current old phone system has been
> limping along for the last couple of years and is about ready to give up
> the ghost. It is off for the summer and the guy who runs it gives no
> guarantees that it will start up again. It's my job to come up with
> alternatives.
>
>
> On 5/27/05, Wiley Siler <[EMAIL PROTECTED]> wrote:
> > I thought he meant that as well but I hope that what will occur is
> > that there is DSL somewhere already that can be utilized.
> > That conflicts with the 'old town PBX' scenario as well though.
> >
> > So, assuming there is DSL already, that even makes you wonder why
> > bother if a phone line already exists and local calling is free?
> >
> > I think we need more details on this one.
> >
> > W
> >
> >
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of Colin
> > Anderson
> > Sent: Friday, May 27, 2005 10:18 AM
> > To: 'brandt Milczewski'; 'Asterisk Users Mailing List - Non-Commercial
>
> > Discussion'
> > Subject: RE: [Asterisk-Users] Newbie here. Tips on setting up 100
> > phones wanted.
> >
> > >It will be about 100 phones at about 20 locations all within about 4
> > >miles of each other.
> >
> > Perhaps a more pressing question might be how you are going to
> > backhaul Ethernet in a 4-mile radius. You can't run a Cat 5 cable more
>
> > than 100 metres reliably, and using Ethernet repeaters every hundred
> > metres or so isn't practical. You will need a fiber backbone or
> something like that.
> > What is your plan to create an Ethernet network to tie these locations
>
> > together?
> > ___
> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
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[Asterisk-Users] Upgraded firmware on Polycom 500, digit-order problems

[Chris Coulthurst]

Ever since I upgraded my Polycom 500 to the newest sip.ld
(kept the old bootrom), when I dial things like “*98” for
voicemail, the screen shows “9*8” and doesn’t dial my
voicemail system!  Is this user error, or errors in the new firmware?

 

Chris Coulthurst

[EMAIL PROTECTED]






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[Asterisk-Users] How to Connect Netphone IP phone with ASterisk

[SYED ADEEL ALI]

I've configured SIP softphone to work with asterisk n it's working fine. 
but i m unable to connect my netphone IP phone I've connected my phone 
to LAN and assigned an IP address to it but how can i make call... plz 
tell me step wise.


_
Express yourself instantly with MSN Messenger! Download today it's FREE! 
http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/


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Re: [Asterisk-Users] VoiPSupply Dot Com

[C F]

On 5/27/05, Karl J. Vesterling <[EMAIL PROTECTED]> wrote:
>  At 08:59 AM 5/27/2005, you wrote:
>  
> [ snip for brevity ]
>  I just wanted to clarify ... this isn't a voipsupply.com problem at all,
> but 
>  rather a courier screwup... which happens anywhere and at anytime... right?
>  
>  TWO screw ups in the shipment.
>  1.) It was shipped to the Bill-To address.  Since there is no one there
> during the day I had to sit and wait for it lest it not be delivered.

This screw up has to do with the person that ordered it, because they
didn't have the ship to address on file with their bank.

>  2.) when an order is placed on a Tuesday AM (or) Monday PM, and it's
> priority overnight, and it's across town, and the tracking number was
> supplied on Wednesday one would expect that it would show up Thursday, not
> Friday.

See above, again this is a screw up that happened because of the one
that ordered it, by NOT having the ship to address on file with their
bank.

Anyhow, you were already answered before that it had to do because YOU
didn't have the address on file with your bank. Why are you repeating
this lie that it is voipsupply.coms fault?
Be repeating it you make yourself look more like a politician or media
person, but certainly not someone that is in the electronic
engineering business. No I will not believe it because I read it
twice, so stop it.
> 
>  So, what we have here is one problem compounded by another, none on behalf
> of the courier.

Exactly, but on behalf of the ordered.

If you give a Ship-To address that is NOT on file with your bank, you
will NOT get it to that address, and it WILL delay shipping.
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[Asterisk-Users] Call waiting on TDM-400 FXO

[Kim Culhan]

Is pstn call waiting working on a Digium TDM-400 with FXO ?

Configuration in zapata.conf:

callwaiting=yes
callwaitingcallerid=yes
callprogress=yes
 
If an incoming call happens while the FXO channel has a call in progress,
and the call is routed to a FXS channel (which has callwaiting=yes in
zapata.conf) the call on the FXO is interrupted.

Is anyone else seeing this ?

-kim

--
[EMAIL PROTECTED]
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[Asterisk-Users] Another OH323 Problem

[Jeromy Grimmett]

Title: Message



anyone got any ideas 
on this?
 
TDM > H323 
Gateway > SIP
 
Inbound H.323 call 
'ip$200.93.237.82:12984/2853' detected.Channel OH323/R2853 created and 
attached for inbound H.323 call 'ip$200.93.237.82:12984/2853'.Setting 
channel 'OH323/R2853' (ip$200.93.237.82:12984/2853) native format to 
g723!Channel OH323/R2853 (call 'ip$200.93.237.82:12984/2853') RX byte count 
is 1.Channel OH323/R2853 (call 'ip$200.93.237.82:12984/2853') RX byte count 
is 2.Channel OH323/R2853 (call 'ip$200.93.237.82:12984/2853') RX byte count 
is 3.Channel OH323/R2853 (call 'ip$200.93.237.82:12984/2853') RX byte count 
is 4.Channel OH323/R2853 (call 'ip$200.93.237.82:12984/2853') RX byte count 
is 5.Channel OH323/R2853 (call 'ip$200.93.237.82:12984/2853') RX byte count 
is 6.Channel OH323/R2853 (call 'ip$200.93.237.82:12984/2853') RX byte count 
is 7.Channel OH323/R2853 (call 'ip$200.93.237.82:12984/2853') RX byte count 
is 8.Channel OH323/R2853 (call 'ip$200.93.237.82:12984/2853') RX byte count 
is 9.Channel OH323/R2853 (call 'ip$200.93.237.82:12984/2853') RX byte count 
is 10.Call 'ip$200.93.237.82:12984/2853' cleared.Call 
'ip$200.93.237.82:12984/2853' without owner has already been cleared 
(2).
any and all help 
would be appreciated...
 
jeromy
 



  
  

  


  

  
  

  


  
  Global reach, local 
  touch...

  

  


  Jeromy GrimmettCEO 
  Comuniquémonos, Inc. / SA1212 South Hampton DriveAlexandria, LA 
71301 

  [EMAIL PROTECTED]IM: MSN: 
[EMAIL PROTECTED]http://www.comuniquemonos.com 
  

  
  
tel: fax: 
  mobile: 
+593 (4) 287 3854(501) 
  646-0680+593 (9) 366 6521 
  
  
  

  


  Add me to your address book...
  Want a signature like 
  this?
 
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[Asterisk-Users] OH323 problem

[Jeromy Grimmett]

Title: Message


May 28 01:58:32 WARNING[6821]: chan_sip.c:1830 
sip_write: Asked to transmit frame type 256, while native formats is 4 
(read/write = 4/4)May 28 01:58:33 WARNING[6821]: channel.c:2126 
ast_channel_make_compatible: No path to translate from SIP/3901506-efd7(4) to 
OH323/L4592(256)May 28 01:58:33 WARNING[6821]: app_dial.c:1024 dial_exec: 
Had to drop call because I couldn't make SIP/3901506-efd7 compatible with 
OH323/L4592
 
this is SIP>h323 
gateway > TDM
 
is anyone getting 
this??? any solutions?
 
jeromy
 



  
  

  


  

  
  

  


  
  Global reach, local 
  touch...

  

  


  Jeromy GrimmettCEO 
  Comuniquémonos, Inc. / SA1212 South Hampton DriveAlexandria, LA 
71301 

  [EMAIL PROTECTED]IM: MSN: 
[EMAIL PROTECTED]http://www.comuniquemonos.com 
  

  
  
tel: fax: 
  mobile: 
+593 (4) 287 3854(501) 
  646-0680+593 (9) 366 6521 
  
  
  

  


  Add me to your address book...
  Want a signature like 
  this?
 
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Re: [Asterisk-Users] Temporary unavailable -????

[Ronald Wiplinger]

Peter Bowyer wrote:


On 27/05/05, Ronald Wiplinger <[EMAIL PROTECTED]> wrote:
 


The person on 617 is unavailable ---  Why
   



Maybe he's in the bathroom?

 



No, I am testing phone (not in the bathroom ;-)  )


The condition being reported is coming from the UA on the end of the
SIP call - is there a DND setting or something there?

 



It is an eyebeam phone, I just downloaded and installed. I can make 
phone calls, but it is not reachable.
Could it be a Windows setting? The server and the phones are on the same 
LAN.



bye

Ronald


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Re: [Asterisk-Users] G729 vs. gsm

[Andrew Kohlsmith]

On May 27, 2005 01:27 pm, Michael D Schelin wrote:
> I have used G729 and it sounds almost as good as G711U. The problem is
> the way Asterisk uses it. It does not sound robotic and it's not suppose
>   to sound that way. Most Carriers want the calls to be in g711u so
> thats why I use G711u otherwise I want to save money on bandwidth. G729
> on Asterisk adds latency. this could be one of your problems. Also you
> will not get music on hold to play well with G729.

Latency is increased any time you have to take time to mangle bits.  This 
means going from any codec to any other codec.  Of course, some codecs take 
more time to mangle the bits than others, and this is the case with any 
decent-sounding compressed codec such as gsm or g729 or even ilbc or speex.

If the latency's too high, get a more powerful system.  The codecs are all 
coded fairly well so there's not much to be gained by trying to reimplement 
the codec.

Also g729 sounds nothing like toll quality; the compressed codecs are able to 
compress so well by throwing away large chunks of the already limited trunk 
bandwidth.  As a result, music sounds like crap on most compressed codecs.  
Personally I find gsm plays on hold music quite well (obviously not amazingly 
well and nowhere near as good as ulaw), but yes g729 mangles it pretty 
well.  :-)

It's not an Asterisk problem, and it's not a g729 problem.  Hell, it's not 
even an implmentation issue.  It's the nature of the beast.

-A.
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RE: [Asterisk-Users] VoiPSupply Dot Com

[Rusty Shackleford]

Title: Message




  
  -Original Message-From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Cory 
  AndrewsSent: Thursday, May 26, 2005 6:33 PMTo: Asterisk 
  Users Mailing List - Non-Commercial DiscussionSubject: RE: 
  [Asterisk-Users] VoiPSupply Dot Com
   
  Karl, first off I apologize for any inconvenience on your recent 
  order.  I will take a look at your transaction to see where things may 
  have gone awry.  We do make mistakes, but we strive to not make the same 
  mistake more than once.  Secondly, I apologize to the list moderator for 
  the pseudo-commercial nature of this post.  The grievance was aired on 
  this list, and I felt compelled to respond to this list and I realize much of 
  this may be more appropriate for the BIZ list. 
   
Cory,
 
You sir, are a class 
act. The message quoted above (snipped, for brevity) is an excellent example of 
how customer relations should be handled. While it appears that the the 
issue was, at least in part, due to some less than effective business 
processes on your end, as well as a partially clueless customer, you 
handled the customer with courtesy and respect; the hallmark of a company 
that truly VALUES their customers. Clearly, you understand that it is the 
interest of the business to make those customers happy. 

 
Some of the other 
vendors on these lists would do well to pay attention to the lesson that Cory 
just gave.


--
Internal Virus Database is out-of-date.
Checked by AVG Anti-Virus.
Version: 7.0.322 / Virus Database: 266.11.14 - Release Date: 05/20/2005
 
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Re: [Asterisk-Users] G729 vs. gsm

[Steve Underwood]

Michael D Schelin wrote:

I have used G729 and it sounds almost as good as G711U. The problem is 
the way Asterisk uses it. It does not sound robotic and it's not 
suppose  to sound that way. Most Carriers want the calls to be in 
g711u so thats why I use G711u otherwise I want to save money on 
bandwidth. G729 on Asterisk adds latency. this could be one of your 
problems. Also you will not get music on hold to play well with G729.


G.729 doesn't sound that bad. However, if you find it hard to tell G.729 
from G.711, I think you should have your hearing checked. :-) It really 
doesn't help people to assess what is right for them, if other people 
make these exaggerated and unreasonable claims. Even the people 
promoting G.729 give it a MOS far below G.711. You are certainly right 
about music on hold, but even voice plus a little background noise can 
sound bloody awful through G.729. Its performance is *very* dependant on 
compressing just a single human voice.


Regards,
Steve

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[Asterisk-Users] Polycom IP 500 SIP bootrom and firmware upgrades

[Jeff Ramsey]

I am using version 1.4.1.0040 and 2.6.1.0003. I have read about newer
versions out there, how can I get them? I'm having an intermittent issue
regarding DHCP with one of my phones, and I recall when I loaded 1.4.1 on
this one phone, it failed once, then it succeeded the second time.

I am wondering if a firmware update will help. Or is there any known issues
with Fedora Core 2 dhcpd and IP 500s?

-- 
Jeff Ramsey
MIS Administrator
Tubafor Mill, Inc.



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Re: [Asterisk-Users] Dropping frame of G.729 since we already have a VAD frame at the end

[Jean-Christophe Heger]

I've got the same issue with a Swissvoice IP10S SIP phone. I couldn't 
find much information with this issue, but it seems to appear because 
Asterisk does not support variable length for g.729 (don't ask me what 
it really means). Anyway, it is recommanded to disable the silence 
suppression, which seems related  to this issue. Unfortunately, it 
didn't work for me. But after setting the phone to canreinvite=no in the 
sip.conf,  the connection worked allright. Don't ask me why.


Jean-Christophe


Bartosz Jozwiak a écrit :


I have this showing on my cli while being in a call.
Then connection gets broken.
Can someone tell me what it means ?

Dropping frame of G.729 since we already have a VAD frame at the end

Thank you in advance.
Bartosz
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Re: [Asterisk-Users] Newbie here. Tips on setting up 100 phones wanted.

[Peter Svensson]

On Fri, 27 May 2005, Mike Clark wrote:

> brandt Milczewski wrote:
> 
> >I work for a ski area. I currently use a 3Com Superstack for in our
> >office. And an old small town phone system for up at the mountain. The
> >phone system is dying and I'm hoping to bring IP to replace the old
> >phones. It will be about 100 phones at about 20 locations all within
> >about 4 miles of each other.
> >
> >I'm looking for tips on the types of phones to look at. Cost is
> >secondary only to reliabity. Any tips?
> >
> We have over 100 Polycomm IP 300/500s installed, and they work great. 
> The 300 will save you $50+ per phone over the 500, but you can't beat 
> the 500 for a quality deskset with a good speakerphone.

For that number of phones buying or getting loaner samples is probably the 
way to go. Resonable alternatives include Polycom, Snom or possibly even 
the GXP-2000, though the latter depends somewhat how much trust you place 
on future software upgrades.

In these quantities you should be able to get a good price on any of these 
phones.

Peter

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Re: [Asterisk-Users] Unable to create channel of type 'Zap' with zaphfc driver

[Jean-Christophe Heger]

Have you placed a group=1 in zapata.conf ?

For a trial, you can use: exten => 203,1,Dial,Zap/1/onetelephonnumber


Aitor a écrit :


I new in asterisk world so, please, forgive me if I say something stupid.
At least, and after a lot of tryes, the isdn card seems to be registered:
[chan_zap.so] => (Zapata Telephony w/PRI)
 == Parsing '/etc/asterisk/zapata.conf': Found
   -- Registered channel 1, PRI Signalling signalling
   -- Registered channel 2, PRI Signalling signalling
   -- Automatically generated pseudo channel
 == Starting D-Channel on span 1   
And some info about channels:

*CLI> zap show channels
  Chan Extension  Context Language   MusicOnHold
pseudodefault
 1default
 2default
*CLI> zap show channel 1
Channel: 1
File Descriptor: 21
Span: 1
Extension:
Dialing: no
Context: default
Caller ID string:
Destroy: 0
InAlarm: 1
Signalling Type: PRI Signalling
Owner: 
Real: 
Callwait: 
Threeway: 
Confno: -1
Propagated Conference: -1
Real in conference: 0
DSP: no
Relax DTMF: no
Dialing/CallwaitCAS: 0/0
Default law: alaw
Fax Handled: no
Pulse phone: no
Echo Cancellation: 128 taps, currently OFF
PRI Flags:
PRI Logical Span: Implicit
Actual Hookstate: Onhook

I'm making this test in extensions.conf
exten => 203,1,Dial,Zap/g1/onetelephonnumber
But asterisk always says:
Unable to create channel of type 'Zap'
What I'm doing wrongly?
Thanks

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Re: [Asterisk-Users] G729 vs. gsm

[Michael D Schelin]

I have used G729 and it sounds almost as good as G711U. The problem is 
the way Asterisk uses it. It does not sound robotic and it's not suppose 
 to sound that way. Most Carriers want the calls to be in g711u so 
thats why I use G711u otherwise I want to save money on bandwidth. G729 
on Asterisk adds latency. this could be one of your problems. Also you 
will not get music on hold to play well with G729.



Andrew Kohlsmith wrote:


On May 27, 2005 06:12 am, chawki hammoud wrote:


I installed G729 from Diguim and I was expecting the
sound quality on my i686 machine to be better than
gsm. Compared to gsm,  G729 sounds closer and a little
robotic. Is this what is supposed to be or am I
missing something?



It sounded more or less the same to me, perhaps with GSM being a little more 
human (I can easily listen to music on hold with GSM).




I am interested in G729 because the internet in my
country is very expensive and I want to save every bit
possible. I want to use G729 because it takes less
bandwidth for each additional call between two IAX
servers than other codecs.



Make sure you use IAX2 trunking then.  It can give you very large bandwidth 
savings when you have multiple audio streams between two servers since the 
UDP overhead is not repeated for every call.


-A.
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RE: [Asterisk-Users] Newbie here. Tips on setting up 100 phones wanted.

[Wiley Siler]

I thought he meant that as well but I hope that what will occur is that
there is DSL somewhere already that can be utilized.
That conflicts with the 'old town PBX' scenario as well though. 

So, assuming there is DSL already, that even makes you wonder why bother
if a phone line already exists and local calling is free?

I think we need more details on this one.

W


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Colin
Anderson
Sent: Friday, May 27, 2005 10:18 AM
To: 'brandt Milczewski'; 'Asterisk Users Mailing List - Non-Commercial
Discussion'
Subject: RE: [Asterisk-Users] Newbie here. Tips on setting up 100 phones
wanted.

>It will be about 100 phones at about 20 locations all within about 4 
>miles of each other.

Perhaps a more pressing question might be how you are going to backhaul
Ethernet in a 4-mile radius. You can't run a Cat 5 cable more than 100
metres reliably, and using Ethernet repeaters every hundred metres or so
isn't practical. You will need a fiber backbone or something like that.
What is your plan to create an Ethernet network to tie these locations
together? 
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RE: [Asterisk-Users] Newbie here. Tips on setting up 100 phones w anted.

[Peter Svensson]

On Fri, 27 May 2005, Colin Anderson wrote:

> >It will be about 100 phones at about 20 locations all within
> >about 4 miles of each other.
> 
> Perhaps a more pressing question might be how you are going to backhaul
> Ethernet in a 4-mile radius. You can't run a Cat 5 cable more than 100
> metres reliably, and using Ethernet repeaters every hundred metres or so
> isn't practical. You will need a fiber backbone or something like that. What
> is your plan to create an Ethernet network to tie these locations together? 

I suppose he could use 10Base5 (Thicknet). That gives you a whooping 500m 
per segment. ;-)

Realistically there are lots of options - fibers, free space optics etc.

Peter


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RE: [Asterisk-Users] Newbie here. Tips on setting up 100 phones wanted.

[Wiley Siler]

Well, that will be pretty preferential

As stated before, I love the Polycom IP500.  I think it is just a great
phone for less than $200.
It configs easily once you get used to the config file and Polycoms have
great speaker phones.

Many love the Ciscos... Admittedly a beautiful phone but harder to
configure as I understand it and you have to get a license...
They tend to cost a little more but they are artwork in motion as far as
looks go.

With 100 phones, I would seriously consider ease of setup.  To that
note, Polycom is supposedly now supporting HTTPS download of config
scripts which means you could create a repository for your files on an
FTP server and secutely transfer the configs over the internet.  Pretty
cool.

Basically, you can go to any voipsupply.com or other and look for phones
that are asthetically pleasing and cost over $150.
Even some of the phones that cost around the $100 are nice.  How fancy
do you really want and how will you manage configuration?

What other info can you give on the project?  Are you proficient with
Asterisk and do you understand issues of SIP (firewall traversal, etc)?

Thanks,
Wiley





 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of brandt
Milczewski
Sent: Friday, May 27, 2005 10:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Newbie here. Tips on setting up 100 phones
wanted.

I'm looking at setting up Asterisk for a completely IP environment.
All intercompany calls.

I work for a ski area. I currently use a 3Com Superstack for in our
office. And an old small town phone system for up at the mountain. The
phone system is dying and I'm hoping to bring IP to replace the old
phones. It will be about 100 phones at about 20 locations all within
about 4 miles of each other.

I'm looking for tips on the types of phones to look at. Cost is
secondary only to reliabity. Any tips?

TIA,
-brandt
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RE: [Asterisk-Users] Newbie here. Tips on setting up 100 phones wanted.

[Dean Collins]

Wireless bridges??


> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Colin Anderson
> Sent: Friday, 27 May 2005 1:18 PM
> To: 'brandt Milczewski'; 'Asterisk Users Mailing List - Non-Commercial
> Discussion'
> Subject: RE: [Asterisk-Users] Newbie here. Tips on setting up 100
phones
> wanted.
> 
> >It will be about 100 phones at about 20 locations all within
> >about 4 miles of each other.
> 
> Perhaps a more pressing question might be how you are going to
backhaul
> Ethernet in a 4-mile radius. You can't run a Cat 5 cable more than 100
> metres reliably, and using Ethernet repeaters every hundred metres or
so
> isn't practical. You will need a fiber backbone or something like
that.
> What
> is your plan to create an Ethernet network to tie these locations
> together?
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Re: [Asterisk-Users] Grandstream GSX-2000 - dead :-(

[Peter Svensson]

On Fri, 27 May 2005, Mark Elkins wrote:

> I tried to do an HTTP update from the Grand Stream web site...

You upgraded the firmware over the Internet? You are braver than I am. I 
would have used a local http server.

> Is there a magic re-incarnation routine ?
> (Power on whilst holding down some buttons?, Sprinkling chickens blood?)

Have you tried the Grandstream support?

> I chose an HTTP upgrade over TFTP - as I read that there were potential
> issues with TFTP at this firmware level.

Tftpd upgrades work well for us on that particular phone.

Peter


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Re: [Asterisk-Users] Newbie here. Tips on setting up 100 phones wanted.

[Mike Clark]

brandt Milczewski wrote:


I'm looking at setting up Asterisk for a completely IP environment.
All intercompany calls.

I work for a ski area. I currently use a 3Com Superstack for in our
office. And an old small town phone system for up at the mountain. The
phone system is dying and I'm hoping to bring IP to replace the old
phones. It will be about 100 phones at about 20 locations all within
about 4 miles of each other.

I'm looking for tips on the types of phones to look at. Cost is
secondary only to reliabity. Any tips?

TIA,
-brandt
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We have over 100 Polycomm IP 300/500s installed, and they work great. 
The 300 will save you $50+ per phone over the 500, but you can't beat 
the 500 for a quality deskset with a good speakerphone.


Mike Clark
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RE: [Asterisk-Users] Newbie here. Tips on setting up 100 phones wanted.

[Adam Collard]

If you have the money, I would recommend the Cisco 7900 series, but if
you need cheap phones, go with sipura. I can get you all you need if you
want. The Sipura phones run about $100. 


Adam Collard
General Manager, ER Wireless
(800) 757-5669 x4861
(810) 496-0161 Fax
(517) 242-1800 Cell
Nextel DC 131*256784*19
[EMAIL PROTECTED]

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of brandt
Milczewski
Sent: Friday, May 27, 2005 5:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Newbie here. Tips on setting up 100 phones
wanted.

I'm looking at setting up Asterisk for a completely IP environment.
All intercompany calls.

I work for a ski area. I currently use a 3Com Superstack for in our
office. And an old small town phone system for up at the mountain. The
phone system is dying and I'm hoping to bring IP to replace the old
phones. It will be about 100 phones at about 20 locations all within
about 4 miles of each other.

I'm looking for tips on the types of phones to look at. Cost is
secondary only to reliabity. Any tips?

TIA,
-brandt
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RE: [Asterisk-Users] Newbie here. Tips on setting up 100 phones w anted.

[Colin Anderson]

>It will be about 100 phones at about 20 locations all within
>about 4 miles of each other.

Perhaps a more pressing question might be how you are going to backhaul
Ethernet in a 4-mile radius. You can't run a Cat 5 cable more than 100
metres reliably, and using Ethernet repeaters every hundred metres or so
isn't practical. You will need a fiber backbone or something like that. What
is your plan to create an Ethernet network to tie these locations together? 
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Re: [Asterisk-Users] Temporary unavailable -????

[Peter Bowyer]

On 27/05/05, Ronald Wiplinger <[EMAIL PROTECTED]> wrote:
> The person on 617 is unavailable ---  Why

Maybe he's in the bathroom?

The condition being reported is coming from the UA on the end of the
SIP call - is there a DND setting or something there?

Peter

-- 
Peter Bowyer
Email: [EMAIL PROTECTED]
Tel: +44 1296 768003
VoIP: sip:[EMAIL PROTECTED]
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[Asterisk-Users] Newbie here. Tips on setting up 100 phones wanted.

[brandt Milczewski]

I'm looking at setting up Asterisk for a completely IP environment.
All intercompany calls.

I work for a ski area. I currently use a 3Com Superstack for in our
office. And an old small town phone system for up at the mountain. The
phone system is dying and I'm hoping to bring IP to replace the old
phones. It will be about 100 phones at about 20 locations all within
about 4 miles of each other.

I'm looking for tips on the types of phones to look at. Cost is
secondary only to reliabity. Any tips?

TIA,
-brandt
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Re: [Asterisk-Users] VoiPSupply Dot Com

[Karl J. Vesterling]

At 08:59 AM 5/27/2005, you wrote:
[ snip for brevity ]
I just wanted to clarify ... this isn't a voipsupply.com problem at all,
but 
rather a courier screwup... which happens anywhere and at anytime...
right?

TWO screw ups in the shipment.
1.) It was shipped to the Bill-To address.  Since there is no one
there during the day I had to sit and wait for it lest it not be
delivered.
2.) when an order is placed on a Tuesday AM (or) Monday PM, and it's
priority overnight, and it's across town, and the tracking number was
supplied on Wednesday one would expect that it would show up Thursday,
not Friday.
So, what we have here is one problem compounded by another, none on
behalf of the courier.
-A.
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Best Regards, 
Karl J. Vesterling
E-Mail: [EMAIL PROTECTED]

Telephone:
Washington DC: (202) 448-3009 Extension 0
Annapolis MD: (240) 524-6706 Extension 0
Bethesda MD: (301) 576-3014 Extension 0
Niagara Falls NY: (716) 286-9175 Extension 0
Buffalo NY: (716) 608-1121 Extension 0 

Yahoo Messenger: karl_vesterling
ICQ: 1548052
AOL Instant Messenger: n2vqm



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RE: [Asterisk-Users] Size of extensions.conf

[John Melody]

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> MF Hulber
> Sent: Thursday, May 26, 2005 10:01 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Size of extensions.conf
> 
> Although you may not see it displayed on a reload it may 
> actually be loaded.  Try "show dialplan" and its alternatives 
> to be sure that what you are looking for is not loaded.
> 
> MARK.

Yes, the show dialplan shows all contexts loaded. I will check the 
bug databases - perhaps someone has logged this before. 

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Re: [Asterisk-Users] DID - B8 Message

[Timothy Costello]

On May 27, 2005, at 6:50 AM, Andrew Kohlsmith wrote:


On May 27, 2005 03:20 am, Nathaniel Angelo A. Torres (247talk) wrote:
Any idea how I can generate a B* message on the asterisk box (out of 
order)

message?


Easy.  Do *not* have an exten => line in your PRI incoming context that
matches the number you want SIT to be played for.  At least for Bell 
Canada
PRIs, * will return  something along the lines of "no number here" and 
Bell

will take care of notifying the calling party.

e.g. if your DID block is 555-1000 to 555-1010 and you want 555-1001 
to appear

out of service:

exten => 5551000,1,DoSomething()
exten => 5551002,1,DoSomething()
exten => 5551003,1,DoSomething()
exten => 5551004,1,DoSomething()
exten => 5551005,1,DoSomething()
exten => 5551006,1,DoSomething()
exten => 5551007,1,DoSomething()
exten => 5551008,1,DoSomething()
exten => 5551009,1,DoSomething()
exten => 5551010,1,DoSomething()

(note the total absence of 5551001)



Or you can have additional control (If your using the CVS version) with:

exten => 5551001,1,SetVar(PRI_CAUSE=1)
exten => 5551001,2,Hangup

For more info see:
http://www.voip-info.org/wiki-Asterisk+variable+PRI_CAUSE

Later;
Tim

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RE: [Asterisk-Users] VoiceMail with Polycom 500

[Wiley Siler]

Not sure I understand your meaning.  You have a phone with 105 as the
registered extension but you want to dial 105 and get voicemail?

Lets assume that *98 is your voicemail extension.

If you dial *98 from any phone it should ask for the extension and
password.

So extension *98 looks something like this

*98,1,Answer
*98,2,VoiceMail()
*98.3,Hangup

Now.  In voicemail.conf, you will have to have an entry for 105.  See
the wikif or examples.

Now.  Back to your query.  Are you saying that you want a phone
configured as extension 105 be able to dial 105 (itself) and dump right
into its own voicemail?  If so, this is not required and * does not work
that way as configured out of box.  Just set these setting in your
Polycom config file and you can get into voicemail via the Voicemail
(some models say messages) button.


  mailto:[EMAIL PROTECTED] On Behalf Of Kib Eki
Sent: Friday, May 27, 2005 6:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] VoiceMail with Polycom 500

Hi,

I want to use the sip extension 105 as the voicemailbox number.
When i initiate a call to the number 105 from my polycom 105 i only get
a call on new line to phone.
But i want in this moment is the voicemailmenu which ask me for my
password.

How can this be done with the polycom phone?

Kib

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Re: [Asterisk-Users] CRM integration (was RE: CallerID)

[Peter Bowyer]

On 27/05/05, Max W Blackmer Jr <[EMAIL PROTECTED]> wrote:
> hello everyone,
> 
> I just had a thought on this subject why not create a daemon process on
> the Client PC That registers its self and What phone the user is
> connected. An AGI script could monitor the progress and when answered
> could send a push to the registered daemon which would push a link to
> the registered daemon on the telephone operators on the desk top. this
> would not waist resource as much as polling?  perhaps the daemon could
> be written in something portable like java or even as a small applet
> that is launched first and minimized to launch the CRM App in a browser
> and could be written into other CRM Applications.

That's effectively what my modifed YAC does - the registration part
currently requires a manual visit to a web page, but could easily be
integrated into YAC.

My YAC listener can accept a command to launch a URL on the client.
The URL can be sent from (eg) an AGI which can take the callerid info
and do whatever smarts are necessary.

Peter
-- 
Peter Bowyer
Email: [EMAIL PROTECTED]
Tel: +44 1296 768003
VoIP: sip:[EMAIL PROTECTED]
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[Asterisk-Users] Temporary unavailable -????

[Ronald Wiplinger]

The person on 617 is unavailable ---  Why

*CLI>
   -- SIP Seeding peers from Astdb: '617' at [EMAIL PROTECTED]:6990 
for 3600

   -- Executing Dial("SIP/601-f18a", "SIP/617|60|tr") in new stack
   -- Called 617
   -- Got SIP response 480 "Temporarily Unavailable" back from 
192.168.250.107

   -- SIP/617-602e is circuit-busy



*CLI> sip show peers
Name/username  HostDyn Nat ACL Mask 
Port Status   
617/617192.168.250.107  D   N  255.255.255.255  
6990 OK (1 ms)


*CLI> sip show users
Username   Secret   Accountcode  
Def.Context  ACL  NAT  
617password 
default  No   Always   




bye

Ronald

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RE: [Asterisk-Users] CRM integration (was RE: CallerID)

[Max W Blackmer Jr]

hello everyone,

I just had a thought on this subject why not create a daemon process on
the Client PC That registers its self and What phone the user is
connected. An AGI script could monitor the progress and when answered
could send a push to the registered daemon which would push a link to
the registered daemon on the telephone operators on the desk top. this
would not waist resource as much as polling?  perhaps the daemon could
be written in something portable like java or even as a small applet
that is launched first and minimized to launch the CRM App in a browser
and could be written into other CRM Applications.

What are your thoughts on this?

Max W. Blackmer Jr.


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Re: [Asterisk-Users] Recommended Network Latency

[Mike Benoit]

Define happily? 

Jitter is obviously important, but latency is too. For day-to-day
business calls, 250ms is a little high. Both parties will definitely
notice it. In my experience you will find yourself talking over one
another quite often. Even with 100ms this continues to happen from time
to time.

There is no doubt that it can be done, but if your latency ever exceeds
300ms, people tend to get frustrated and it could really start to cause
problems.

Once your agents realize there is a 250ms latency, and get used to it,
it might not be so bad. But personally, it wouldn't be acceptable to me.



On Fri, 2005-05-27 at 16:25 +0100, Tony Hoyle wrote:
> Waldo Rubinstein wrote:
> > 
> > I'm planning on setting up some remote agents and before doing so, I  
> > did some simple PING tests to measure latency. The average latency I  
> > got was 250ms. Does anyone have experience in terms of quality of  calls 
> > when there is such high latency? Can anyone comment?
> > 
> Latency isn't the issue - you could happily carry on a call over a 
> 2000ms latency.. satellite links can introduce this easily.
> 
> Your problem is jitter (ping stability).  Some software will calculate 
> this for you - eg. mtr gives the standard deviation for its ping history 
> (mine hovers around the 0.4 mark over 12 hops which is really stable AFAIK).
> 
> You can reduce jitter using good QoS, but it's better to have it as low 
> as possible to start with.
> 
> Tony
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-- 
Mike Benoit <[EMAIL PROTECTED]>


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Re: [Asterisk-Users] PRI "Actual-HookState" not showing offhook on inbound

[Andrew Kohlsmith]

On May 27, 2005 10:57 am, Ronald Hartmann wrote:
> Actual Hookstate: Onhook

What version of Asterisk is this?  CVS HEAD from about an hour ago does not 
show hookstate for anything but FXS interfaces, and certainly not for PRI.

-A.
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RE: [Asterisk-Users] Grandstream GSX-2000 - dead :-(

[jltaylor]

I've got three GS 100 Phones with same problem.
Some lights.
Some no lights.
Some garbled display.

I would welcome suggestions for a resurrection.

James Taylor
MetroTel
3505 Summerhill Road
Suite 11
Texarkana, Tx  75503
903-793-1956


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Mark Elkins
Sent: Friday, May 27, 2005 10:01 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Grandstream GSX-2000 - dead :-(


I have a Grandstream GSX-2000 with ..
Software Version:Program-- 1.0.0.3Bootloader-- 1.0.0.3

I tried to do an HTTP update from the Grand Stream web site...

After half an hour, I recycled power and now its dead... LED's come on
and stay on, screen and buttons are dead. Connectivity to
Grandstream.com was always good - whenever I checked (I downloaded the
"User Manual" in a couple of minutes), the site states five minutes to
load, so waiting more than 30 mins should have been OK, and they do have
this "Please Powercycle" in red print too...

Is there a magic re-incarnation routine ?
(Power on whilst holding down some buttons?, Sprinkling chickens blood?)

I chose an HTTP upgrade over TFTP - as I read that there were potential
issues with TFTP at this firmware level.


-- 
  .  . ___. .__  Posix Systems - Sth Africa.  e.164 VOIP ready
 /| /|   / /__   [EMAIL PROTECTED]  -  Mark J Elkins, Cisco CCIE
/ |/ |ARK \_/ /__ LKINS  Tel: +27 12 807 0590  Cell: +27 82 601 0496

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RE: [Asterisk-Users] Dial By Name

[Carlton O'Riley]

Check the documentation for the Directory application on
http://www.voip-info.org .

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> [EMAIL PROTECTED]
> Sent: Friday, May 27, 2005 10:54 AM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Dial By Name
> 
> 
> Is it possible to have a Dial my name menu in Asterisk?   
> 
> AZ
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Re: [Asterisk-Users] Recommended Network Latency

[Tony Hoyle]

Waldo Rubinstein wrote:


I'm planning on setting up some remote agents and before doing so, I  
did some simple PING tests to measure latency. The average latency I  
got was 250ms. Does anyone have experience in terms of quality of  calls 
when there is such high latency? Can anyone comment?


Latency isn't the issue - you could happily carry on a call over a 
2000ms latency.. satellite links can introduce this easily.


Your problem is jitter (ping stability).  Some software will calculate 
this for you - eg. mtr gives the standard deviation for its ping history 
(mine hovers around the 0.4 mark over 12 hops which is really stable AFAIK).


You can reduce jitter using good QoS, but it's better to have it as low 
as possible to start with.


Tony
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Re: [Asterisk-Users] pressing a key to get in of voicemail?

[Moises Silva]

may be something like this?

[macro-sipextens]
exten => s,1,SetVar(voicemail=${ARG1})
exten => s,2,Dial(SIP/${ARG1},40,r)
exten => s,3,GotoIf($[${DIALSTATUS} = NOANSWER] ? 66  : 3)
exten => s,4,GotoIf($[${DIALSTATUS} = BUSY] ? 68  : 4)
exten => s,5,Playback(iss_invalid_sipexten)
exten => s,6,Hangup()
exten => s,66,Goto(ask_voice_mail_no_answer,s,1)
exten => s,68,Goto(ask_voice_mail_busy,s,1)

the context ask_voice_mail_no_answer could be
[ask_voice_mail_no_answer]
exten => s,1,Background(press_1_for_voicemail)
exten => 1,1,Voicemail(u${voicemail})
exten => 1,2,Hangup()

regards

On 5/26/05, Ing CIP Alejandro Celi Mariátegui <[EMAIL PROTECTED]> wrote:
> 
> I've currently got Asterisk configured to take incoming calls, ask for
> extension, ring the phone and send them directly to the voicemail.
> 
> What I want to be able to do is first a message "press 1 for voicemail
> or hangup" before voicemail come up.
> 
> Any ideas?
> 
> regards,
> 
> --
> Ing CIP Alejandro Celi Mariátegui
> <[EMAIL PROTECTED]>
> 
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-- 
"Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org";
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[Asterisk-Users] Recommended Network Latency

[Waldo Rubinstein]

I'm planning on setting up some remote agents and before doing so, I  
did some simple PING tests to measure latency. The average latency I  
got was 250ms. Does anyone have experience in terms of quality of  
calls when there is such high latency? Can anyone comment?


Thanks,
Waldo
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[Asterisk-Users] Grandstream GSX-2000 - dead :-(

[Mark Elkins]

I have a Grandstream GSX-2000 with ..
Software Version:Program-- 1.0.0.3Bootloader-- 1.0.0.3

I tried to do an HTTP update from the Grand Stream web site...

After half an hour, I recycled power and now its dead... LED's come on
and stay on, screen and buttons are dead. Connectivity to
Grandstream.com was always good - whenever I checked (I downloaded the
"User Manual" in a couple of minutes), the site states five minutes to
load, so waiting more than 30 mins should have been OK, and they do have
this "Please Powercycle" in red print too...

Is there a magic re-incarnation routine ?
(Power on whilst holding down some buttons?, Sprinkling chickens blood?)

I chose an HTTP upgrade over TFTP - as I read that there were potential
issues with TFTP at this firmware level.


-- 
  .  . ___. .__  Posix Systems - Sth Africa.  e.164 VOIP ready
 /| /|   / /__   [EMAIL PROTECTED]  -  Mark J Elkins, Cisco CCIE
/ |/ |ARK \_/ /__ LKINS  Tel: +27 12 807 0590  Cell: +27 82 601 0496

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Re: [Asterisk-Users] Interco H323 : IPNx (from WTL) and *

[Zoa]

I tried it a while, its impossible.
(Well you can get it to work, but not in a stable way)

Zoa.


Laurent Tostain wrote:


Hi,

   Someone released a succefull interconnection in H323 with WTL equipement
?
   I'm trying to do that with an IPNx. But get dead air.

   With chan_oh323 it's fine, all works. With chan_h323 => dead air.

   The configuration is GW to GW.

   This is my configuration from h323.conf:

   [general]
   port=1720
   bindaddr=my.ipaddr
   dtmfmode=rfc2833
   gatekeeper=DISABLE
   AllowGKRouted=no
   context=default

   [termination]
   type=peer
   host=some.ipaddr
   noFastStart=no
   noH245Tunneling=no
   noSilenceSuppression=yes
   progress_setup=3
   progress_alert=8
   progress_audio=yes
   disallow=all
   allow=ulaw

   Any Suggestion is welcome.

Thanks!

__
Laurent TOSTAIN


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[Asterisk-Users] PRI "Actual-HookState" not showing offhook on inbound

[Ronald Hartmann]

Good day list,

I have been fighting echo problems on my PRI card.

Everything is working great outbound, however inbound calls have
echo.

I have found the issue but need help in fixing it.

During outbound calling "zap show channel 18" shows that the
call is off-hook and echo cancellation is running.

During inbound calling though, "zap show channel 18" shows that
the call is "ON-HOOK" and echo cancellation is turned off.

I have no ideas as to how or why this is happening.  Any input is
greatly appreciated.

~ron


[CLI DIAGNOSTICS:]
*CLI> zap show channels
   Chan Extension  Context Language   MusicOnHold
 pseudofrom-pstn   en
 18 5134632401 from-pstn   en
 19 5134632401 from-pstn   en
 20from-pstn   en
 21from-pstn   en
 22from-pstn   en
 23from-pstn   en


*CLI> zap show channel 18
Channel: 18CLI>
File Descriptor: 19
Span: 1
Extension: 5134632401
Dialing: no
Context: from-pstn
Caller ID string: 513486
Destroy: 0
InAlarm: 0
Signalling Type: PRI Signalling
Owner: Zap/18-1
Real: Zap/18-1
Callwait: 
Threeway: 
Confno: -1
Propagated Conference: -1
Real in conference: 0
DSP: yes
Relax DTMF: no
Dialing/CallwaitCAS: 0/0
Default law: ulaw
Fax Handled: no
Pulse phone: no
Echo Cancellation: 128 taps, currently OFF
PRI Flags: Call
PRI Logical Span: Implicit
Actual Confinfo: Num/0, Mode/0x
Actual Confmute: No
Actual Hookstate: Onhook



[Zapata.conf]
[channels]
language=en

signalling=pri_cpe ; pri_cpe = PRI slave ; pri_net = PRI master
switchtype=national
pridialplan=unknown
echocancel=yes ; You can set this to 32, 64, or 128, tweak to your
needs.
echocancelwhenbridged=yes
echotraining=400 ; Asterisk trains to the beginning of the call, number
is in milliseconds
usecallerid=yes
callerid=asreceived
overlapdial=yes
immediate=no
group=0,1
context=from-pstn
channel => 18-23 ; Set this to 1-15,17-31 for E1


[ZAPTEL.conf]
span=1,1,0,esf,b8zs
bchan=18-23 # set this to 1-15,17-31 for E1
dchan=24 # set this to 16 for E1

defaultzone=us
loadzone=us






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[Asterisk-Users] Unable to create channel of type 'Zap' with zaphfc driver

[Aitor]

I new in asterisk world so, please, forgive me if I say something stupid.
At least, and after a lot of tryes, the isdn card seems to be registered:
[chan_zap.so] => (Zapata Telephony w/PRI)
  == Parsing '/etc/asterisk/zapata.conf': Found
-- Registered channel 1, PRI Signalling signalling
-- Registered channel 2, PRI Signalling signalling
-- Automatically generated pseudo channel
  == Starting D-Channel on span 1   

And some info about channels:
*CLI> zap show channels
   Chan Extension  Context Language   MusicOnHold
 pseudodefault
  1default
  2default
*CLI> zap show channel 1
Channel: 1
File Descriptor: 21
Span: 1
Extension:
Dialing: no
Context: default
Caller ID string:
Destroy: 0
InAlarm: 1
Signalling Type: PRI Signalling
Owner: 
Real: 
Callwait: 
Threeway: 
Confno: -1
Propagated Conference: -1
Real in conference: 0
DSP: no
Relax DTMF: no
Dialing/CallwaitCAS: 0/0
Default law: alaw
Fax Handled: no
Pulse phone: no
Echo Cancellation: 128 taps, currently OFF
PRI Flags:
PRI Logical Span: Implicit
Actual Hookstate: Onhook

I'm making this test in extensions.conf
exten => 203,1,Dial,Zap/g1/onetelephonnumber
But asterisk always says:
Unable to create channel of type 'Zap'
What I'm doing wrongly?
Thanks

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[Asterisk-Users] Dial By Name

[azasadny]

Is it possible to have a Dial my name menu in Asterisk? 

AZ
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[Asterisk-Users] Interco H323 : IPNx (from WTL) and *

[Laurent Tostain]

Hi,
 
Someone released a succefull interconnection in H323 with WTL equipement
?
I'm trying to do that with an IPNx. But get dead air.

With chan_oh323 it's fine, all works. With chan_h323 => dead air.

The configuration is GW to GW.

This is my configuration from h323.conf:

[general]
port=1720
bindaddr=my.ipaddr
dtmfmode=rfc2833
gatekeeper=DISABLE
AllowGKRouted=no
context=default

[termination]
type=peer
host=some.ipaddr
noFastStart=no
noH245Tunneling=no
noSilenceSuppression=yes
progress_setup=3
progress_alert=8
progress_audio=yes
disallow=all
allow=ulaw

Any Suggestion is welcome.

Thanks!
   
__
Laurent TOSTAIN


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