date:20050907


 I am wanting to front-end a legacy PBX with an asterisk box. I have done 
 plenty 
 of asterisk work over the last 6 months to PRI circuits, but not with a PBX 
 being involved.
 
 I know I can use asterisk and digium cards in this manner, but do I need 
 separate cards for the PRI - Asterisk side to the Asterisk - PBX side, or 
 will 
 a 4-port PRI card do the job? (I already have a spare one of these).

The 4-port card will work just fine.

 In other words, can I use SPAN 1 as a timing source, then provide timing to 
 the 
 PBX connected to SPAN 2 of the same card?

Yes. In fact, the 4-port card will be a slight advantage over two 
single port cards as all ports on the 4-port card will have their
clocks in sync with your external timing source.

Keep in mind that all T1/E1 spans having timing embedded in their
transmit legs; you can't turn that off even if you tried. The clock
timing source is always an engineering decision as to chosing which
receive leg to use for clock sync. (Obviously, the span from the
pstn would be your timing source and not the span to the pbx. If
you already are using the PRI with the PBX, then no changes required
on the PBX side for clock sync.)

The config examples in zapata.conf and the wiki are good. Not much
to configure really.

You will probably want to focus more on options that your pstn 
provider can/will impact such as the number of digits to be sent 
from them to you, which channel is the d channel, the digits they 
expect from you for each call (whether prefixed with 1, 0 or 
whatever), etc.

As sort of a side note, the 4-port card gives you another slight
advantage from an ongoing support perspective. The third (or forth)
port could be connected to a test asterisk box on which you can
stage/test future asterisk code before moving it into the production
box. Think about reserving a couple of DID numbers for the test
box if you'll be using DID.


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Re: [Asterisk-Users] PHP and ASterisk Manager

2005-09-07 Thread Christoph Eicke

I looked into the source code of Asterisk to figure out how the printf() 
statements were spaced. That's the power of open source, you can look under 
the hood for these questions. It's easy to find, even for non-C-Gurus. Just 
do a grep for the string that you want inside of the Asterisk source 
directory and it will give you the file that the string you are looking for 
is in. Then simply open the file, search for the string and look at the 
printf() statement.

Christoph

On Tuesday 06 September 2005 21:16, Anton Krall wrote:
 I was able to do and if and while loops to get the block of lines I want..
 Now.. Another issue.

 I need to parse the line read to insert it into a table but seems Asterisk
 inserts TABS or SPACES inconsistantly.. For example:

 Xxx(TAB)xxx(5 spaces)xxx
 Next line
 Xxx(TAB)xxx(3 spaces)xxx

 Im having a hard time figuring out how Asterisk Manager returns the stuff
 :)

 Well..s o far so good...

 |-Original Message-
 |From: [EMAIL PROTECTED]
 |[mailto:[EMAIL PROTECTED] On Behalf Of
 |Matthew Boehm
 |Sent: Martes, 06 de Septiembre de 2005 01:49 p.m.
 |To: Asterisk Users Mailing List - Non-Commercial Discussion
 |Subject: Re: [Asterisk-Users] PHP and ASterisk Manager
 |
 |Anton Krall wrote:
 | Guys, is anybody using PHP sockets to connect to the Manager
 |
 |and send
 |
 | command like show voicemail users for example or any other?
 |
 | My question is, how to parse the return info in a way that can be
 | shown back to the user via web (discard all the manager
 |
 |responses not needed)?
 |
 |Use preg_match() to match the lines you want the user to see
 |on the website.
 |
 |$socket = fsockopen(localhost,5038, $errno, $errstr, 30);
 |
 |if(!$socket) {
 | print No socket;
 | exit();
 |}
 |
 |fputs($socket, Action: Login\r\n);
 |fputs($socket, Events: Off\r\n);
 |fputs($socket, UserName: bleh\r\n);
 |fputs($socket, Secret: bleh\r\n\r\n);
 |
 |fputs($socket, Action: Command\r\n);
 |fputs($socket, Command: show channels\r\n\r\n);
 |
 |fputs($socket, Action: Logoff\r\n\r\n);
 |
 |while(!feof($socket)) {
 | $buff = fgets($socket,1024);
 | if(preg_match(/SIP\/.*/, $buff)) {
 | print I found a SIP call;
 | }
 |}
 |
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RE: [Asterisk-Users] queues

2005-09-07 Thread altus

Hi
So if I have this
queues.conf
[general]
[default]
[example_queue]
music = default
strategy = rrmemory
context = queue-out ; Here we go when the caller presses a single digit,
while in the queue
timeout = 20
wrapuptime=10
announce-frequency = 30
announce-holdtime = yes
joinempty = yes
member = SIP/101
member = SIP/102
member = SIP/103
member = SIP/104

extensions.conf

exten = 3,1,Playback(some_announce)
exten = 3,2,Queue(example_queue|tT|||300) 
exten = 3,3,Dial(SIP/100)  

It will ring 104 for 20s,then 103 for 20s,then 102 for 20s and then 101
for 20s.

It will keep on doing this for 300s then go the 100

If a second call comes it,it will start at 103 then 102 ens?
Thanks for the help


On Wed, 2005-09-07 at 08:07 +0200, Jens von Bülow wrote:
 Hi Altus,
 
 Try roundrobin with memory...
 
 snip
 Calls are distributed among the members handling a queue with one of several 
 strategies, defined in queues.conf 
 
 ringall: ring all available channels until one answers (default) 
 roundrobin: take turns ringing each available interface 
 leastrecent: ring interface which was least recently called by this queue 
 fewestcalls: ring the one with fewest completed calls from this queue 
 random: ring random interface 
 rrmemory: round robin with memory, remember where we left off last ring pass
 /snip
 
 Also, as a rule of thumb, if you look at a call queues from the clients' 
 perspective, a ringall strategy is what you have to do... (the others just 
 can add huge delays in answering a call).
 
 Hope that Helps
 Jens
 
 
 -Original Message-
 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of altus
 Sent: 07 September 2005 07:57 AM
 To: asterisk
 Subject: [Asterisk-Users] queues
 
 Good day all
 I need some help with queues please.
 I know how to do a rounrobin in the queues.conf but I dont think its
 going to work in this situation 
 Got got a IVR setup and option 3 is sales
 The sales people are 101,102,103,104 and the switchboard is 100
 The trick comes is
 The 1st call for extension 3 goes to 101,but if 101 does not answer in
 20 it goes to the switchboard,100
 Then the second call of the day goes to 102,if not answer in 20s it goes
 to the switchboard,100
 and so on and then just starts over again.
 Do I uses queues for this and then how?If I put it in a queues.conf and
 a roundroben,wont it then just try 101,and if not answer then 102 and if
 no answer 103...and so on?
 This is my queses.conf
 
 [general]
 
 [default]
 [example_queue]
 music = default
 strategy = roundrobin
 context = queue-out ; Here we go when the caller presses a single digit,
 while in the queue
 timeout = 20
 wrapuptime=10
 announce-frequency = 30
 announce-holdtime = yes
 joinempty = yes
 member = SIP/101
 member = SIP/102
 member = SIP/103
 member = SIP/104
 
 and my extensions.conf 
 
 exten = 3,1,Playback(some_announce)
 exten = 3,2,Queue(example_queue|tT|||20) 
 exten = 3,3,Dial(SIP/100)  
 
 
 h
 
 
 aph
 raph
 
 
 h
 
 Æ 
  
-- 

Thanks
Altus Snyman
Stormcorp Network Solutions
+27 11 8071141 exten 301

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[Asterisk-Users] Some info about Cisco's 79xx, and Sipura's phones

2005-09-07 Thread Irakli Natsvlishvili


Hello folks,

I've did some tests with different phones and Asterisk last two days and 
here are some results, which I want to share with audience.


Cisco's 79xx and Sipura's phones/adapters on INVITE always reply with their 
preferred codec.


So, for example, if Cisco's/Sipura's phone has preferred_codec g729a(18) and 
it receives INVITE from UA which has preferred codec ULAW(0), it will always 
reply with g729 and ignore what is preferred codec of calling party.


Also, if two UAs have canreinvite=yes in SIP.CONF, then there is no 
difference in which order codecs are listed. If Cisco/Sipura's UA is called, 
then resulted codec after re-INVITEs will be preferred codec of CALLED 
party.


There are other UA's which reply with the preferred codec of calling party. 
For example, SNOM and Grandstream behave this way.


Hope this helps.

I.N. 


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Re: [Asterisk-Users] Utility to find length of wav49 file

2005-09-07 Thread Malcolm Taylor

Thanks Flynn.  

Unfortunately the files aren't written by the voicemail application.  I was
hoping that there was some little command-line utility which would return
basic sound information when passed the filename.

Malcolm

-Original Message-
From: El Flynn [mailto:[EMAIL PROTECTED] 
Sent: Tuesday, September 06, 2005 9:08 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: [SPAM - header check] - Re: [Asterisk-Users] Utility to find length
of wav49 file - Email found in subject

Malcolm Taylor wrote:
 Can anyone point me in the direction of a utility which will let me 
 determine the length (in seconds) of a wav49 file created by Asterisk?
 
 Many thanks,
 
   Malcolm
 

if you're talking about the duration of a voicemail, you could do:

grep duration msg.txt

from the command-line. each voicemail left has an accompanying text file
that gives details about the message.

unless you're talking about something completely different...

Flynn

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[Asterisk-Users] zaptel init script

2005-09-07 Thread Christian Richter


Hi List,

we've made a litle script which is called /etc/init.d/zaptel. It scans 
the pci bus and creates by request a /etc/zaptel.conf and a 
/etc/asterisk/zapa.conf.


Also it loads the modules automagically.

If there are volunteers who want to try this out (it'll make first setup 
of an asterisk with digium cards easier) just grab it at:


www.beronet.com/downloads/zaptel-init.tar.gz

and type make install after unpacking.

Good Luck ;)


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[Asterisk-Users] Hosted PBX (vPBX) and Call/PickUP Groups

2005-09-07 Thread René Mayorga

Hi,
I'm working with this issue for a while, Now I already solve the
dialplan issues, but I still have a question about the Callgroups,
I read at www.voip-info.org that , there is a 63 limit of callgroups.
And I'm wondering why?? and if the 1.2.0beta version supported more than
63 Groups?? (I did'nt find any Changelog for 1.2) or If not There is any
unoficial patch for that ?

Thanks in advance.

-- 
René Mayorga [EMAIL PROTECTED]
El Salvador Telecom S.A. de C.V.

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Re: [Asterisk-Users] PRI in and out

2005-09-07 Thread altus

I got the same setup,sort of
I connected a single port sangoma to my pbx
My ony problem is,when a call comes in and it gets transfered back out
that it does not detect the hangup?So that channel keeps being open
Any ideas why


On Wed, 2005-09-07 at 01:40 -0600, Rich Adamson wrote:
  I am wanting to front-end a legacy PBX with an asterisk box. I have done 
  plenty 
  of asterisk work over the last 6 months to PRI circuits, but not with a PBX 
  being involved.
  
  I know I can use asterisk and digium cards in this manner, but do I need 
  separate cards for the PRI - Asterisk side to the Asterisk - PBX side, or 
  will 
  a 4-port PRI card do the job? (I already have a spare one of these).
 
 The 4-port card will work just fine.
 
  In other words, can I use SPAN 1 as a timing source, then provide timing to 
  the 
  PBX connected to SPAN 2 of the same card?
 
 Yes. In fact, the 4-port card will be a slight advantage over two 
 single port cards as all ports on the 4-port card will have their
 clocks in sync with your external timing source.
 
 Keep in mind that all T1/E1 spans having timing embedded in their
 transmit legs; you can't turn that off even if you tried. The clock
 timing source is always an engineering decision as to chosing which
 receive leg to use for clock sync. (Obviously, the span from the
 pstn would be your timing source and not the span to the pbx. If
 you already are using the PRI with the PBX, then no changes required
 on the PBX side for clock sync.)
 
 The config examples in zapata.conf and the wiki are good. Not much
 to configure really.
 
 You will probably want to focus more on options that your pstn 
 provider can/will impact such as the number of digits to be sent 
 from them to you, which channel is the d channel, the digits they 
 expect from you for each call (whether prefixed with 1, 0 or 
 whatever), etc.
 
 As sort of a side note, the 4-port card gives you another slight
 advantage from an ongoing support perspective. The third (or forth)
 port could be connected to a test asterisk box on which you can
 stage/test future asterisk code before moving it into the production
 box. Think about reserving a couple of DID numbers for the test
 box if you'll be using DID.
 
 
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-- 

Thanks
Altus Snyman
Stormcorp Network Solutions
+27 11 8071141 exten 301

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[Asterisk-Users] asterisk, SIP, Re-INVITEs and different contexts

2005-09-07 Thread Irakli Natsvlishvili


Hello!

Hmmm... Folks, I beg you pardon, if I'm telling something which was said 
before, but actually I have not found this anywhere, neither on 
Voip-info.org or in several Asterisk's docs.


So, here is the statement:

If SIP extensions are in DIFFERENT CONTEXTS, then RTP traffic between them 
will ALWAYS go via Asterisk.


I.e. Asterisk WILL NOT issue Re-INVITE even if:

1. Both UAs have canreinvite=yes in their SIP.CONF
2. Both UAs have same codecs
3. There are no t, T settings in Dial command.

I'd like to have a confirmation from * developers about this statement.


I.N.
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Re: [Asterisk-Users] asterisk, SIP, Re-INVITEs and different contexts

2005-09-07 Thread Olle E. Johansson

Irakli Natsvlishvili wrote:
 If SIP extensions are in DIFFERENT CONTEXTS, then RTP traffic between
 them will ALWAYS go via Asterisk.
Dial plan contexts has nothing to do with how we set up RTP traffic.

 I.e. Asterisk WILL NOT issue Re-INVITE even if:
 
 1. Both UAs have canreinvite=yes in their SIP.CONF
If canreinvite=yes, we *will* issue a re-invite if possible.
 2. Both UAs have same codecs
 3. There are no t, T settings in Dial command.
Or h,H or nat=yes.

It is easier to turn it around:
Asterisk will issue a re-invite unless there is a reason
to keep the audio going through Asterisk

* NAT traversal issues
* Canreinvite=no
* Anything that needs asterisk to listen for DTMF in call
* Codecs that needs to be transcoded

/Olle

---
Astricon 2005 - where you will learn about Asterisk and re-invites!
http://www.astricon.net/2005/ October 12-14 Anaheim, California
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[Asterisk-Users] Max concurrent faxes with txfax/spandsp?

2005-09-07 Thread Roger Schreiter


Hi,

I tried to use txfax to send several faxes at the
same time.
It seams, that one can't send more than 3 faxes at once,
or one risks to get 50% and more aborted faxes due to
errors.

The CPU usage is below 97%.
I tried with Opteron and IntelP4: same result.

Ok, I know, that faxing via a digital line is complicate,
and I shouldn't complain, but I would like to know, whether
these are typical values or whether one could increase
the max fax number by any means? Maybe force to a slower,
but more error proof modulation?


Regards,
Roger.


P.S.
After faxing approx 100 faxes,
CLI show channels
shows a lot of channels, which seam to be forgotten, not
hangup faxlines.
Is this a known weakness of txfax?

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Re: [Asterisk-Users] zaptel init script

2005-09-07 Thread Tzafrir Cohen

On Wed, Sep 07, 2005 at 09:32:58AM +0200, Christian Richter wrote:
 Hi List,
 
 we've made a litle script which is called /etc/init.d/zaptel. It scans 
 the pci bus and creates by request a /etc/zaptel.conf and a 
 /etc/asterisk/zapata.conf.
 
 Also it loads the modules automagically.
 
 If there are volunteers who want to try this out (it'll make first setup 
 of an asterisk with digium cards easier) just grab it at:
 
 www.beronet.com/downloads/zaptel-init.tar.gz
 
 and type make install after unpacking.
 
 Good Luck ;)

Are you aware of http://tzafrir.org.il/genzaptelconf ?

That script is intended for one-time discovery. The current zaptel init.d
script in Xorcom Rapid is much simpler and has very little discovery: 
if no zaptel card module was loaded, it will load ztdummy.

And who is expected to actually load a card driver at boot time? Well,
the system has all that information and it is the job of the hotplug
script to extract it and load the relevant modules. So far it has done
that very well.

My aproach in the script was different: parse information in 
/proc/zaptel/ . Though I admit that the end result is an over-grown bash
script . The atvantage is that it is easy to debug: /proc/zaptel/n only
exsits if the module was loaded. I don't have to hope which card belongs
to which span because I look at spans directly.

There are some things I was not so happy with in my script. The defaults
for ISDN switch types (both BRI and PRI) are probably not good enough.
And there is no reasonable way to get decent per-channel or per-span
configuration into the auto-generated parts of zapata.conf .

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend
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Re: [Asterisk-Users] zaptel init script

2005-09-07 Thread Christian Richter


Tzafrir Cohen wrote:


On Wed, Sep 07, 2005 at 09:32:58AM +0200, Christian Richter wrote:
 


Hi List,

we've made a litle script which is called /etc/init.d/zaptel. It scans 
the pci bus and creates by request a /etc/zaptel.conf and a 
/etc/asterisk/zapata.conf.


Also it loads the modules automagically.

If there are volunteers who want to try this out (it'll make first setup 
of an asterisk with digium cards easier) just grab it at:


www.beronet.com/downloads/zaptel-init.tar.gz

and type make install after unpacking.

Good Luck ;)
   



Are you aware of http://tzafrir.org.il/genzaptelconf ?

 


No i wasn't.


That script is intended for one-time discovery. The current zaptel init.d
script in Xorcom Rapid is much simpler and has very little discovery: 
if no zaptel card module was loaded, it will load ztdummy.


And who is expected to actually load a card driver at boot time? Well,
the system has all that information and it is the job of the hotplug
script to extract it and load the relevant modules. So far it has done
that very well.

My aproach in the script was different: parse information in 
/proc/zaptel/ . Though I admit that the end result is an over-grown bash

script . The atvantage is that it is easy to debug: /proc/zaptel/n only
exsits if the module was loaded. I don't have to hope which card belongs
to which span because I look at spans directly.

There are some things I was not so happy with in my script. The defaults
for ISDN switch types (both BRI and PRI) are probably not good enough.
And there is no reasonable way to get decent per-channel or per-span
configuration into the auto-generated parts of zapata.conf .

 

I see. I had a short look over your script and found it makes nearly the 
same thing like /etc/init.d/zaptel. But our approach is only to generate 
a default zaptel.conf and zapata.conf without any extensions, trunk and 
phones stuff.


But thanks for this info we hadn't loaded ztdummy if no card was 
available, now we're doing it also ;)


Greets,

crich
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Re: [Asterisk-Users] Which Linux distribution?

2005-09-07 Thread John Daragon


YT Lim wrote:

We have tried Asterisk 1.0.9 on FC4 and have never
been able to get CAPI (with Fritz card, fcpci) to work
properly. Apart from that Asterisk works fine in
switching internal calls. But it's useless if we can't
make outgoing calls on our ISDN line.

We are considering abandoning FC4 for Debian or SuSe.
What is the general concensus on the best Linux to run
Asterisk with CAPI?


SUSE (as far as I know) is the only distro that really *expects* you to 
be using ISDN2e as a matter of course.


jd
--

John Daragon  [EMAIL PROTECTED]
argv[0] limited
Lambs Lawn Cottage,  Staple Fitzpaine,  Taunton,  TA3 5SL,  UK
v +44 (0) 1460 234068   f +44 (0) 1460 234069   m +44 (0) 7836 576127


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[Asterisk-Users] How to connect many analog lines to Asterisk?

2005-09-07 Thread Josip Gracin


Hello!

If I have more than a hundred analog telephones (analog lines) that need 
to be connected to Asterisk PBX, what kind of hardware do I need, and 
where can I buy it?


Thanks in advance!
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Re: [Asterisk-Users] How to connect many analog lines to Asterisk?

2005-09-07 Thread gincantalupo


Hi,
try to search with google for channelbank or something similar.

Giorgio

Josip Gracin wrote:


Hello!

If I have more than a hundred analog telephones (analog lines) that 
need to be connected to Asterisk PBX, what kind of hardware do I need, 
and where can I buy it?


Thanks in advance!
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Re: [Asterisk-Users] Ethernet / TcpIp phones

2005-09-07 Thread gincantalupo


Hi,
can you be a little clearer???
Every VoIP hardphone can be connected to Ethernet except for USB models.

Giorgio

Alex wrote:

Is there any VoIP phones available which can be plugged directly to 
the Ethernet network?


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Re: [Asterisk-Users] Ethernet / TcpIp phones

2005-09-07 Thread Christoph Eicke

try google for VoIP Phone ;-)
or here: http://www.voip-info.org/tiki-index.php?page=Asterisk+phones

On Wednesday 07 September 2005 11:19, Alex wrote:
 Is there any VoIP phones available which can be plugged directly to the
 Ethernet network?

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[Asterisk-Users] Ethernet / TcpIp phones

2005-09-07 Thread Alex

Is there any VoIP phones available which can be plugged directly to the 
Ethernet network?


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[Asterisk-Users] presence settings and Eyebeam

2005-09-07 Thread Vahan Yerkanian


What is the proper way of adding hints to multiple extensions?


In my case extensions are the same as the sip usernames, while as per 
http://www.voip-info.org/tiki-index.php?page=Asterisk%20presence


exten = 1234,hint,SIP/1234 works,

exten = _1,hint,SIP/${EXTEN} doesn't. Not sure if I can even use 
${EXTEN} here...


Any hints?
Vahan
begin:vcard
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n:Yerkanian;Vahan
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Re: [Asterisk-Users] CTI and Asterisk

2005-09-07 Thread Giovanni Miano

Il modo migliore è quello di utilizzare AMI (Asterisk Mang. Interface)

Buon lavoro

2005/9/7, Stefano Blasco [EMAIL PROTECTED]:
  
  
 
 Hi all, 
 
 i have a question: 
 
   
 
 what about a CTI implementation with Asterisk. 
 
 I've been looking for info in www.voip-info.org and in google, but 
 
 There are no precise informations! 
 
   
 
 Thanks a lot 
 
   
 
 stefano 
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Re: [Asterisk-Users] Which Linux distribution?

2005-09-07 Thread Tzafrir Cohen

On Wed, Sep 07, 2005 at 10:10:05AM +0100, John Daragon wrote:
 YT Lim wrote:
 We have tried Asterisk 1.0.9 on FC4 and have never
 been able to get CAPI (with Fritz card, fcpci) to work
 properly. Apart from that Asterisk works fine in
 switching internal calls. But it's useless if we can't
 make outgoing calls on our ISDN line.
 
 We are considering abandoning FC4 for Debian or SuSe.
 What is the general concensus on the best Linux to run
 Asterisk with CAPI?
 
 SUSE (as far as I know) is the only distro that really *expects* you to 
 be using ISDN2e as a matter of course.

Only Linux distro that is generally something that is a bit hasty to
say, given the fact that there are so many of them ;-) .

Mandrake is quite Europe-centric as well. I'm not sure about ISDN
support. 

Debian has generally a large european installed base and a variety of
ISDN-related packages as a result.

Sorry, I won't make your life easier :-p

-- 
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Re: [Asterisk-Users] presence settings and Eyebeam

2005-09-07 Thread Olle E. Johansson

Vahan Yerkanian wrote:
 What is the proper way of adding hints to multiple extensions?
 
 
 In my case extensions are the same as the sip usernames, while as per
 http://www.voip-info.org/tiki-index.php?page=Asterisk%20presence
 
 exten = 1234,hint,SIP/1234 works,
 
 exten = _1,hint,SIP/${EXTEN} doesn't. Not sure if I can even use
 ${EXTEN} here...
 
 Any hints?
File a bug report if it does not work. I think it would be a good idea
if it works, even though I usually don't recommend using the extension
as the peer name. ;-)

/O
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Re: [Asterisk-Users] Working example of ALERT_INFO with Cisco ATAs?

2005-09-07 Thread Olle E. Johansson

Brian Capouch wrote:
 I am wondering if there are any tricks getting the Cisco ATAs to do
 distinctive rings via the ALERT_INFO variable?
 
 I have seen some contradictory information in the Wiki, and I tried the
 example there.  I then sniffed the connection between the server and the
 ATA and didn't see the header sent like it is supposed to be.
 
 If someone out there has a handle on this and would be willing to help,
 I'd sure appreciate it.
 
 I'm doing this right now with ARA; the table entry in question looks
 like this (sorry about linewrap):
 
  exten  | priority |  app   | appdata
 +--++-
  brianc | 1| SetMusicOnHold | native-random
  brianc | 2| SetVar | ALERT_INFO=Bellcore-dr2
  brianc | 3| NoOp   | ${ALERT_INFO}
  brianc | 4| Dial   | SIP/ata1|23|t
 
 At priority 3 I can see that the variable has been set correctly, but
 nothing ever gets sent out.  Everything else (e.g. the MOH) works just
 fine.
 
 I'm much obliged for any help that might be lurking out there.
Try setting _ALERT_INFO

/Olle
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[Asterisk-Users] ISDN PBX integration

2005-09-07 Thread Shahar Livne

Hello list,

I am trying to connect an old ISDN PBX to my asterisk system.
The setup includes an asterisk (1.0.9) running on the Soekris
hardware, with an ISDN card (Billion BIPAC PCI), and I run
zaphfc-bristuff-0.2.0-RC8k kernel module in NT mode (modes=1).

When I connect an ISDN phone to the card (using cross ISDN cable + 40v
power supply), I manage to make calls from and to the asterisk.

When I try to connect the ISDN PBX instead, still using the same cross
ISDN cable and the power supply, I get rings to the right extensions
of the ISDN PBX, but no call setup happens, and after 2 rings comes
silence.

I have tried this with NT and TE modes of the zaphfc kernel module (I
thought it has to do with point-to-point or point-to-multipoint
issue). I have tried it both with cross ISDN cable and normal ISDN
cable (I just used Ethernet cable), all without any success (not even
the mentioned rings).

When this ISDN PBX is connected to normal ISDN line, it just works.

If there is anyone with some knowledge about ISDN equipment and its
integration with asterisk – I will be happy to get some advice – what
to check or how is it actually supposed to work.

Thanks,
Shahar

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[Asterisk-Users] channels VHF/ HF radio in asterisk

2005-09-07 Thread makevuy


Hy,
I have a network with WIFI communication and VHF/ HF channels.
I have integrated asterisk in the network using SIP, ZAP and IAX2 
channels for WIFI communications, but I don't Know How I could integrate 
the VHF/ HF channels.


I have heard speaking about app_rpt project, but I don't Know very much 
about this.


Could I integrate VHF/ HF channels with this application? if the answer 
is yes, How?


Regards.

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[Asterisk-Users] Packet Cable

2005-09-07 Thread Chris Mason (Lists)

The local CATV company is offering internet using packet cable, and they 
have asked about using Asterisk in their office. Is there any working 
packet cable interface?


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[Asterisk-Users] -- PROGRESS with cause code 34 received?

2005-09-07 Thread Roy Sigurd Karlsbakk


hi

i get these messages every now and then

-- PROGRESS with cause code 34 received

wtf is this?

roy
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Re: [Asterisk-Users] Which Linux distribution?

2005-09-07 Thread John Daragon


Tzafrir Cohen wrote:

On Wed, Sep 07, 2005 at 10:10:05AM +0100, John Daragon wrote:


YT Lim wrote:


We have tried Asterisk 1.0.9 on FC4 and have never
been able to get CAPI (with Fritz card, fcpci) to work
properly. Apart from that Asterisk works fine in
switching internal calls. But it's useless if we can't
make outgoing calls on our ISDN line.

We are considering abandoning FC4 for Debian or SuSe.
What is the general concensus on the best Linux to run
Asterisk with CAPI?


SUSE (as far as I know) is the only distro that really *expects* you to 
be using ISDN2e as a matter of course.



Only Linux distro that is generally something that is a bit hasty to
say, given the fact that there are so many of them ;-) .


You're absolutely right.


Mandrake is quite Europe-centric as well. I'm not sure about ISDN
support. 


It's shipped with the packages; I looked at it when I first started 
installing *, but couldn't get fcpci to work at the time.  CAPI appears 
to have been written on (or for) SUSE in the first place, and SUSE was 
the first distro I came across that supported ISDN2e out of the box.




Debian has generally a large european installed base and a variety of
ISDN-related packages as a result.

Sorry, I won't make your life easier :-p



You mean it's *supposed* to be easy ?

jd
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Re: [Asterisk-Users] presence settings and Eyebeam

2005-09-07 Thread Vahan Yerkanian

Done. Not sure if picked categories under SIP Mantis correct but here it 
 is: http://bugs.digium.com/view.php?id=5149


VY

Olle E. Johansson wrote:

File a bug report if it does not work. I think it would be a good idea
if it works, even though I usually don't recommend using the extension
as the peer name. ;-)

/O
begin:vcard
fn:Vahan Yerkanian
n:Yerkanian;Vahan
org:ARMINCO Global Telecommunications;Head, Research  Development dept.
adr:;;28, Isahakian ave., PO BOX 10;Yerevan;;375009;Armenia
email;internet:[EMAIL PROTECTED]
x-mozilla-html:FALSE
url:http://www.arminco.com/
version:2.1
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Re: [Asterisk-Users] Some problems (SendDTMF, Wait, Parked Calls)

1  2. You could use the dial macro.  Check out the screening macro on
http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+Dial

More 1. To send tones, use SendDTMF:
http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+SendDTMF

A little more 1.  I'm not sure the best way to pause for a second. 
You could record a second of silence and play it back or you could
create an agi that calls sleep for a second.

3.  Just make an extension you can dial from your cellphone that goes
to the ParkAndAnnounce app:
http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+ParkAndAnnounce

On 9/6/05, Rubens Sanchez [EMAIL PROTECTED] wrote:
 Hi all! I would like to solve some problems:
 
 I have a sip provider that lets me make pstn calls after listening some
 stuff and entering a pin number:
 
 1) How can I make Asterisk enter the pin number? Then wait 1 second and
 enter the phone number?
 I have in extensions.conf:
 exten = 6*,1,Dial,SIP/[EMAIL PROTECTED],60,tr
 I have tried with w (like with ZAP channels) but it does not work, nor
 having a second priority with SendDTMF.
 
 2) How can I silence the first seconds of this call (so I do not have to
 listen to their stuff)? or play some music on hold?
 
 3) Another diferent problem I would like to solve is how to park and
 incoming call without answering, so I can call Asterisk from my cellphone,
 dial the 700 extension and the other person does not have to pay during that
 time.
 
 Thanks,
 Rubens
 
 
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Re: [Asterisk-Users] Hosted PBX (vPBX) and Call/PickUP Groups

I'm not sure about why, but it's it is hardcoded into asterisk.  Back
when it was a limit of 31, I searched around and increased the value
on my box and recompiled.  It did not seem to adversely affect the
system.

On 9/7/05, René Mayorga [EMAIL PROTECTED] wrote:
 Hi,
 I'm working with this issue for a while, Now I already solve the
 dialplan issues, but I still have a question about the Callgroups,
 I read at www.voip-info.org that , there is a 63 limit of callgroups.
 And I'm wondering why?? and if the 1.2.0beta version supported more than
 63 Groups?? (I did'nt find any Changelog for 1.2) or If not There is any
 unoficial patch for that ?
 
 Thanks in advance.
 
 --
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 El Salvador Telecom S.A. de C.V.
 
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[Asterisk-Users] Desincripcion de la lista de Asterisk

2005-09-07 Thread Will Velez




Buenos días quiero 
que ya no me llegue mas correo electrónico de la lista Asterisk, como puedo 
hacerlo
Gracias
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Re: [Asterisk-Users] Desincripcion de la lista de Asterisk

Unsubscribe directions are at the bottom of each email.  

Translation via google: Las direcciones de unsubscribe están en el
fondo de cada email.


On 9/7/05, Will Velez [EMAIL PROTECTED] wrote:
 Buenos días quiero que ya no me llegue mas correo electrónico de la lista
 Asterisk, como puedo hacerlo
 Gracias
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Re: [Asterisk-Users] channels VHF/ HF radio in asterisk

2005-09-07 Thread Mark Phillips


2 ways.

1) buy into the app_rpt system. They have a bespoke card for your PC 
that can drive radio's etc. It's mainly aimed at repeater owners.


2) connect a phone patch between an ATA and your HF rig. This solution 
is currently being used to provied phone services from a few Red Cross 
shelters to the ARC HQ in Montgomery, AL. It works well.


Mark, KC2ENI

makevuy wrote:

Hy,
I have a network with WIFI communication and VHF/ HF channels.
I have integrated asterisk in the network using SIP, ZAP and IAX2 
channels for WIFI communications, but I don't Know How I could integrate 
the VHF/ HF channels.


I have heard speaking about app_rpt project, but I don't Know very much 
about this.


Could I integrate VHF/ HF channels with this application? if the answer 
is yes, How?


Regards.

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--

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Re: [Asterisk-Users] Packet Cable

2005-09-07 Thread Mark Phillips

Why do you care about an interface? The job of your cable modem/bridge 
should be to convert from your local ethernet to their peculiar data 
network.


/JFDI

Mark

Chris Mason (Lists) wrote:
The local CATV company is offering internet using packet cable, and they 
have asked about using Asterisk in their office. Is there any working 
packet cable interface?




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Re: [Asterisk-Users] Which Linux distribution?

2005-09-07 Thread Mark Phillips

Being a German package this would make sense. ISDN is DT's circuit of 
choice and can be found in the vast majority of businesses across Der 
Fatherland.




John Daragon wrote:

YT Lim wrote:


We have tried Asterisk 1.0.9 on FC4 and have never
been able to get CAPI (with Fritz card, fcpci) to work
properly. Apart from that Asterisk works fine in
switching internal calls. But it's useless if we can't
make outgoing calls on our ISDN line.

We are considering abandoning FC4 for Debian or SuSe.
What is the general concensus on the best Linux to run
Asterisk with CAPI?



SUSE (as far as I know) is the only distro that really *expects* you to 
be using ISDN2e as a matter of course.


jd


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[Asterisk-Users] Eeven Stranger - Asterisk BT100 Password Issue

2005-09-07 Thread Aisling









Following on from my below email, things
have taken another bizarre twist



I have continued getting the error when
2092 tries to listen to messages by dialing .



--Executing VoiceMailMain
(SIP/2092-6918, 2092) in new stack

--Playing vm-password
(language en)

WARNING: app_voicemail.c:4922 vm_authentication: Unable to read password.



Then I decided to plug out my BT100 and
left it plugged out for a few hours. When I plugged it back in and dialed  for
voicemail, bizarrely I could hear the voicemail main menu and was prompted for
a password. When I entered the password, I was able to listen to the messages..This
is what appeared on the Asterisk console



--Executing VoiceMailMain
(SIP/2092-6918, 2092) in new stack

--Playing vm-password
(language en)

--Incorrect password 1234
for user 2092 context = any) //here I entered the
incorrect password 1234

--Playing vm-incorrect
(language en)

--Playing vm-password
(language en)

--Incorrect password 1234
for user 2092 context = any) //again here I entered
the incorrect password 1234

--Playing vm-incorrect
(language en)

--Playing vm-password
(language en)

Unable to create lock file
/var/spool/asterisk/voicemail/from-sip/2092/Old/:
No such file or directory

Unable to create lock file
/var/spool/asterisk/voicemail/from-sip/2092/Old/:
No such file or directory

Unable to create lock file
/var/spool/asterisk/voicemail/from-sip/2092/INBOX/:
No such file or directory

Unable to create lock file
/var/spool/asterisk/voicemail/from-sip/2092/INBOX/:
No such file or directory

-- Playing vm-youhave
(language en) .//here I entered the correct
password and heard that I had no messages

-- Playing vm-no
(language en)

-- Playing vm-messages
(language en)

--Playing vm-opts
(language en)





But then to add another twist, I hung up
the phone and dialed  again. This time it didnt work and I got the
same old error as before. I tried plugging out the phone again but it did not
make a difference.



Does anyone know what those extra
messages on the console mean or how I can solve this? I am obviously missing
something important  How do I get it?



Many Thanks.





-Original Message-
From: Aisling
[mailto:[EMAIL PROTECTED]] 
Sent: 06 September 2005 18:09
To:
'asterisk-users@lists.digium.com'
Subject: Asterisk BT100 Password Issue



Hi,



I am getting the following error when I attempt to listen to
voice messages by dialing  (I can hear nothing):



--Executing VoiceMailMain (SIP/2092-6918,
2092) in new stack

--Playing vm-password (language
en)

WARNING: app_voicemail.c:4922 vm_authentication: Unable to
read password.



I read in previous posts that this may be to do with the
dtmf settings and have set both (asterisk and BT100) to info. This has not
helped. My phones register with SER (port 5060) and use Asterisk for voicemail
(port 5064).

My configs are below along with my BT100 settings:



;Grandstream BT100



SIP Server:  x.x.x.x:5060

SIP User ID: 2092

Authenticate ID: 2092

Name 2092



SER then forwards to port 5064 of Asterisk.



;sip.conf



[general]

bindport=5064

bindaddr=0.0.0.0

disallow=all

allow=ulaw

allow=alaw

allow=gsm

srvlookup=yes

canreinvite=no

autocreeper=yes

nat=yes



[2092]

type=friend

username=2092

canreinvite=no

context=from-sip

mailbox=2092

host=dynamic

nat=no

dtmfmode=INFO

disallow=all

allow=alaw

allow=ulaw



;extensions.conf

[general]

static=yes

writeprotect=yes



[from-sip]

exten = 2092, 1, Dial (SIP/2092, 20)

exten = 2092, 2 , Voicemail (u2092)

exten = 2092, 102, Voicemail (b2092)

exten = 2092, 103, Hangup



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



;voicemail.conf

[general]

format=wav



[from-sip]

2092 = 2092, 2092, emailaddress



Has anyone any inkling as to what the cause could be?



Many thanks,

Aisling.




---Legal  Disclaimer---

The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.




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Re: [Asterisk-Users] Eeven Stranger - Asterisk BT100 Password Issue

I always get an unable to read password error if I hang up without
entering a password when prompted.  Maybe is this what you are doing? 
Even if you hear nothing, it is probably still expecting a password to
be entered.

On 9/7/05, Aisling [EMAIL PROTECTED] wrote:
 
 
 Following on from my below email, things have taken another bizarre twist……
 
  
 
 I have continued getting the error when 2092 tries to listen to messages by
 dialing .
 
  
 
 --Executing VoiceMailMain (SIP/2092-6918, 2092) in new stack
 
 --Playing 'vm-password' (language 'en')
 
 WARNING: app_voicemail.c:4922 vm_authentication: Unable to read password.
 
  
 
 Then I decided to plug out my BT100 and left it plugged out for a few hours.
 When I plugged it back in and dialed   for voicemail, bizarrely I could
 hear the voicemail main menu and was prompted for a password. When I entered
 the password, I was able to listen to the messages…..This is what appeared
 on the Asterisk console
 
  
 
 --Executing VoiceMailMain (SIP/2092-6918, 2092) in new stack
 
 --Playing 'vm-password' (language 'en')
 
 --Incorrect password '1234' for user '2092' context = any) //here I
 entered the incorrect password 1234
 
 --Playing 'vm-incorrect' (language 'en')
 
 --Playing 'vm-password' (language 'en')
 
 --Incorrect password '1234' for user '2092' context = any) //again
 here I entered the incorrect password 1234
 
 --Playing 'vm-incorrect' (language 'en')
 
 --Playing 'vm-password' (language 'en')
 
  Unable to create lock file 
 '/var/spool/asterisk/voicemail/from-sip/2092/Old/': No such
 file or directory
 
 Unable to create lock file 
 '/var/spool/asterisk/voicemail/from-sip/2092/Old/': No such
 file or directory
 
 Unable to create lock file 
 '/var/spool/asterisk/voicemail/from-sip/2092/INBOX/': No
 such file or directory
 
 Unable to create lock file 
 '/var/spool/asterisk/voicemail/from-sip/2092/INBOX/': No
 such file or directory
 
 -- Playing 'vm-youhave' (language 'en') …….//here I entered the correct
 password and heard that I had no messages
 
 -- Playing 'vm-no' (language 'en')
 
 -- Playing 'vm-messages' (language 'en')
 
 --Playing 'vm-opts' (language 'en')
 
  
 
  
 
 But then to add another twist, I hung up the phone and dialed  again.
 This time it didn't work and I got the same old error as before. I tried
 plugging out the phone again but it did not make a difference.
 
  
 
 Does anyone know what those extra messages on the console mean or how I can
 solve this? I am obviously missing something important – How do I get it?
 
  
 
 Many Thanks.
 
  
 
  
 
 -Original Message-
 From: Aisling [mailto:[EMAIL PROTECTED] 
 Sent: 06 September 2005 18:09
 To: 'asterisk-users@lists.digium.com'
 Subject: Asterisk BT100 Password Issue
 
  
 
 Hi,
 
  
 
 I am getting the following error when I attempt to listen to voice messages
 by dialing  (I can hear nothing):
 
  
 
 --Executing VoiceMailMain (SIP/2092-6918, 2092) in new stack
 
 --Playing 'vm-password' (language 'en')
 
 WARNING: app_voicemail.c:4922 vm_authentication: Unable to read password.
 
  
 
 I read in previous posts that this may be to do with the dtmf settings and
 have set both (asterisk and BT100) to info. This has not helped. My phones
 register with SER (port 5060) and use Asterisk for voicemail (port 5064).
 
 My configs are below along with my BT100 settings:
 
  
 
 ;Grandstream BT100
 
  
 
 SIP Server:x.x.x.x:5060
 
 SIP User ID:  2092
 
 Authenticate ID: 2092
 
 Name2092
 
  
 
 SER then forwards to port 5064 of Asterisk.
 
  
 
 ;sip.conf
 
  
 
 [general]
 
 bindport=5064
 
 bindaddr=0.0.0.0
 
 disallow=all
 
 allow=ulaw
 
 allow=alaw
 
 allow=gsm
 
 srvlookup=yes
 
 canreinvite=no
 
 autocreeper=yes
 
 nat=yes
 
  
 
 [2092]
 
 type=friend
 
 username=2092
 
 canreinvite=no
 
 context=from-sip
 
 mailbox=2092
 
 host=dynamic
 
 nat=no
 
 dtmfmode=INFO
 
 disallow=all
 
 allow=alaw
 
 allow=ulaw
 
  
 
 ;extensions.conf
 
 [general]
 
 static=yes
 
 writeprotect=yes
 
  
 
 [from-sip]
 
 exten = 2092, 1, Dial (SIP/2092, 20)
 
 exten = 2092, 2 , Voicemail (u2092)
 
 exten = 2092, 102, Voicemail (b2092)
 
 exten = 2092, 103, Hangup
 
  
 
 exten = , 1, VoicemailMain(${CALLERIDNUM})
 
  
 
 ;voicemail.conf
 
 [general]
 
 format=wav
 
  
 
 [from-sip]
 
 2092 = 2092, 2092, emailaddress
 
  
 
 Has anyone any inkling as to what the cause could be?
 
  
 
 Many thanks,
 
 Aisling.---Legal
 Disclaimer--- The above
 electronic mail transmission is confidential and intended only for the
 person to whom it is addressed. Its contents may be protected by legal
 and/or professional privilege. Should it be received by you in error please
 contact the sender at the above quoted email address. Any unauthorised form
 of reproduction of this message is strictly prohibited. The Institute does
 not guarantee the security

[Asterisk-Users] 2 X100P and SIP outbound routing

2005-09-07 Thread Paul Goodyear

Current setup

2 x X100P cards connected to 2 analogue lines
 Using prefix 7 and 8 before number
SIP gateway to SipGate to make VoIP calls
 Using prefix 9 before number.

Is it possible so that if I dial a number:

0800 8000 8000

that it will try to route the call over the first analogue line, if
that fails/busy, goto analogue line 2, if that fails/budy then goto the
SIP account, if for some reason this fails, or the net is down, then
the No lines available is played.

Thanks.

Paul.
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RE: [Asterisk-Users] Eeven Stranger - Asterisk BT100 Password Issue

2005-09-07 Thread Aisling

I hear absolutely nothing. The problem is I don't even get a chance to
enter the password. I dial  and press send on my phone. Immediately
the following error appears on the asterisk console:

--Executing VoiceMailMain (SIP/2092-6918, 2092) in new stack
--Playing 'vm-password' (language 'en')
WARNING: app_voicemail.c:4922 vm_authentication: Unable to read
password.

So if I enter the password it makes absolutely no difference (I've tried
nothing happens). That one time that it did work (when I plugged my
phone out for a few hours - strange!), I heard the menu. I was prompted
for the password and when I entered it I heard that I had no messages. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Flobi
Sent: 07 September 2005 14:02
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Eeven Stranger - Asterisk BT100 Password
Issue

I always get an unable to read password error if I hang up without
entering a password when prompted.  Maybe is this what you are doing? 
Even if you hear nothing, it is probably still expecting a password to
be entered.

On 9/7/05, Aisling [EMAIL PROTECTED] wrote:
 
 
 Following on from my below email, things have taken another bizarre
twist..
 
  
 
 I have continued getting the error when 2092 tries to listen to
messages by
 dialing .
 
  
 
 --Executing VoiceMailMain (SIP/2092-6918, 2092) in new stack
 
 --Playing 'vm-password' (language 'en')
 
 WARNING: app_voicemail.c:4922 vm_authentication: Unable to read
password.
 
  
 
 Then I decided to plug out my BT100 and left it plugged out for a few
hours.
 When I plugged it back in and dialed   for voicemail, bizarrely I
could
 hear the voicemail main menu and was prompted for a password. When I
entered
 the password, I was able to listen to the messages...This is what
appeared
 on the Asterisk console
 
  
 
 --Executing VoiceMailMain (SIP/2092-6918, 2092) in new stack
 
 --Playing 'vm-password' (language 'en')
 
 --Incorrect password '1234' for user '2092' context = any)
//here I
 entered the incorrect password 1234
 
 --Playing 'vm-incorrect' (language 'en')
 
 --Playing 'vm-password' (language 'en')
 
 --Incorrect password '1234' for user '2092' context = any)
//again
 here I entered the incorrect password 1234
 
 --Playing 'vm-incorrect' (language 'en')
 
 --Playing 'vm-password' (language 'en')
 
  Unable to create lock file 
 '/var/spool/asterisk/voicemail/from-sip/2092/Old/': No such
 file or directory
 
 Unable to create lock file 
 '/var/spool/asterisk/voicemail/from-sip/2092/Old/': No such
 file or directory
 
 Unable to create lock file 
 '/var/spool/asterisk/voicemail/from-sip/2092/INBOX/': No
 such file or directory
 
 Unable to create lock file 
 '/var/spool/asterisk/voicemail/from-sip/2092/INBOX/': No
 such file or directory
 
 -- Playing 'vm-youhave' (language 'en') ...//here I entered the
correct
 password and heard that I had no messages
 
 -- Playing 'vm-no' (language 'en')
 
 -- Playing 'vm-messages' (language 'en')
 
 --Playing 'vm-opts' (language 'en')
 
  
 
  
 
 But then to add another twist, I hung up the phone and dialed 
again.
 This time it didn't work and I got the same old error as before. I
tried
 plugging out the phone again but it did not make a difference.
 
  
 
 Does anyone know what those extra messages on the console mean or how
I can
 solve this? I am obviously missing something important - How do I get
it?
 
  
 
 Many Thanks.
 
  
 
  
 
 -Original Message-
 From: Aisling [mailto:[EMAIL PROTECTED] 
 Sent: 06 September 2005 18:09
 To: 'asterisk-users@lists.digium.com'
 Subject: Asterisk BT100 Password Issue
 
  
 
 Hi,
 
  
 
 I am getting the following error when I attempt to listen to voice
messages
 by dialing  (I can hear nothing):
 
  
 
 --Executing VoiceMailMain (SIP/2092-6918, 2092) in new stack
 
 --Playing 'vm-password' (language 'en')
 
 WARNING: app_voicemail.c:4922 vm_authentication: Unable to read
password.
 
  
 
 I read in previous posts that this may be to do with the dtmf settings
and
 have set both (asterisk and BT100) to info. This has not helped. My
phones
 register with SER (port 5060) and use Asterisk for voicemail (port
5064).
 
 My configs are below along with my BT100 settings:
 
  
 
 ;Grandstream BT100
 
  
 
 SIP Server:x.x.x.x:5060
 
 SIP User ID:  2092
 
 Authenticate ID: 2092
 
 Name2092
 
  
 
 SER then forwards to port 5064 of Asterisk.
 
  
 
 ;sip.conf
 
  
 
 [general]
 
 bindport=5064
 
 bindaddr=0.0.0.0
 
 disallow=all
 
 allow=ulaw
 
 allow=alaw
 
 allow=gsm
 
 srvlookup=yes
 
 canreinvite=no
 
 autocreeper=yes
 
 nat=yes
 
  
 
 [2092]
 
 type=friend
 
 username=2092
 
 canreinvite=no
 
 context=from-sip
 
 mailbox=2092
 
 host=dynamic
 
 nat=no
 
 dtmfmode=INFO
 
 disallow=all
 
 allow=alaw
 
 allow=ulaw
 
  
 
 ;extensions.conf
 
 [general]
 
 static=yes

[Asterisk-Users] IAX PBX responds to IAX registration with expires time=0

2005-09-07 Thread Maciek

Hallo
There is the scenario:

client   server

   --- REGREQ with expires=60  ---
   .
   -- REGACK with expires=0   

I did not see such situation previously, I mean PBX always responded with 
expires!=0.
What does it mean? How should it be treated?

greetings



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po raz pierwszy na żywo! Katowice, 22.09, Warszawa, 26.09, 
Bydgoszcz, 27.09. Więcej: WWW.metalopolis.pl
http://klik.wp.pl/?adr=http%3A%2F%2Fadv.reklama.wp.pl%2Fas%2Farena.htmlsid=492


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[Asterisk-Users] Speex codec - Out of buffer space

2005-09-07 Thread David Hajek

Hi,

I'm running Asterisk 1.0.7 and would like to add Speex support. I
downloaded Speex 1.0.5, installed and recompile Asterisk again.

Now trying from X-Lite to connect using Speex but getting lot of weird
erros on Asterisk console:

Sep  7 15:03:25 WARNING[28605]: codec_speex.c:166 speextolin_framein:
Out of buffer space

I was trying to setup Speex on my second Asterisk server and wanted to
use this codec for IAX between these two boxes. But I'm getting unable
to negotiate codecs. Other codecs works like a charm. 

Any ideas?

Thank you.

--
David
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[Asterisk-Users] Polycom 300 with latest 1.5.3 firmware not registering

2005-09-07 Thread Jorge Alayon


Hello,

I got 3 Polycom 300 phones, and upgraded to the latest firmware provided by the 
reseller.

This is my first experience with Polycom and I cannot make them register in my 
Asterisk Box.

I followed every advice I found (including separating [user] and [peer] in 
sip.conf.

Using ethereal, I found that it tries to SUBSCRIBE to the asterisk box and it 
receives a 403 FORBIDDEN message.

I compared to a Grandstream registration, and it tries to REGISTER to the 
asterisk receiving a 200 Message response and effectively registering.

Finding in the packet capture no other great difference, I believe that 
SUBSCRIBE requires a different authentification approach, maybe related to the 
voIpProt.SIP.requestValidation.digest.realm parameter in sip.cfg. I Tried the 
Polycom default, empty, default (voicemail context), from-internal (Extension 
context), the IP of the asterisk box, the name of the asterisk box, asterisk, 
etc, with no result. I tried different approaches documented in the wiki and 
related pages with no result. I can makke calls but I cannot receive them.

I've seen mails stating that some installations have more than 100 phones 
working perfectly, can someone point me in the right direction to solve this ?

Regards,

Jorge A.
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Re: [Asterisk-Users] Asterisk overheating on VIA Epia M Seriesmotherboard

2005-09-07 Thread Dustin Wildes

Angus - I have several mini-itx systems based on the Epia MII6000 
(fanless) system.
They all run great, and I have no problems.  I also run 'mpg123' with 
several mp3s.

I run it in an embedded configuration (in house).

However, I do remember one board that I got where the heatsink on the 
CPU was loose which caused the thermal compound to be detached from the CPU.
I removed the heatsink and put a silver compound in the place of the 
other compound, and we were okay again.


My systems usually run around 45C-50C under load.


Angus Comber wrote:

But the systems are sold in this configuration.  There is a fan 
option.  I chose the fanless option.


Angus

- Original Message - From: C F [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com

Sent: Tuesday, September 06, 2005 1:28 AM
Subject: Re: [Asterisk-Users] Asterisk overheating on VIA Epia M 
Seriesmotherboard



As you suspected, the problem is the fact that you don't have a fan.
Since a machine that runs just a file server does not require much CPU
power, the CPU doesn't get too hot. However Asterisk does use lots of
CPU, therefore the CPU is hot, and yes the problem of stopping to work
is because of the CPU being overheated, you are lucky that the
computer booted after that, in most cases the overheating of a CPU
means that the CPU expanded too much, when you shut it down it cools
off, and shrinks, which could result in cracking the CPU. You should
never run a CPU without it's fan if it's meant to run with a fan. Even
if running it just as a file server. The fact that you are lucky
doesn't mean that you don't need a fan.

On 9/5/05, Angus Comber [EMAIL PROTECTED] wrote:



Hello

I am running Asterisk on SUSE Linux Professional 9.3 on a VIA Epia M 
Series
motherboard - CPU runs at 1GHz.  There is no fan - just a large 
heatsink.
Currently system is running off standard IDE hard drive - because I 
couldn't
get astlinux to run with my Digium TDM04B card (only PCI card in 
system).


Strangely I also have the same system also running SUSE Linux running 
as a

file server and that does not run so hot and does not overheat?  Why the
difference?

Just booting up both systems for 15 minutes you can tell the Asterisk 
box is
quite a bit hotter.  Also the Asterisk box overheated (well think 
that was

the problem) and stopped operating as PBX at one stage.

Anyone any experience of this sort of thing?  any ideas how to fix - 
ideally

I don't want to have to fit a fan.

Is SUSE not the best distro to use for this sort of thing?  Should it be
something to take up with VIA?

Angus


__




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RES: [Asterisk-Users] Billing - Disable accounts when balance gets0 value

2005-09-07 Thread itn

[EMAIL PROTECTED]

Simoni,

Thank you for your copersation. If you need routes in Brazil I have very
high quality ones ok...  

Atenciosamente
 
 
Reduzimos ao mínimo a sua conta de Telefone
Liguetel - ITN Info - 15 anos em Telecomunicações
 
Diretoria Comercial - Newton Medina
PABX11.3891.2434
Fax  11.38980112
msn [EMAIL PROTECTED] 
 
Rua Augusta 2.212 SL 26 Jardins 01412001
São Paulo - Brasil 
 
Visite a Loja www.liguetel.com.br ou www.liguetel.com 
e conheça produtos e serviços para reduzir definitivamente a sua conta
de telefone. 
 

-Mensagem original-
De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Em nome de
Simone Cittadini
Enviada em: segunda-feira, 5 de setembro de 2005 05:54
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Assunto: Re: [Asterisk-Users] Billing - Disable accounts when balance
gets0 value


This billing is also able to set accounts balance and for each call.
Now I
need to disable accounts which balance gets a determined value. I was
thinking on changing account pass for that specif account which we need
to
disable. And then in the sip.com reload info.

Can you help me with new (new ways for doing so) or programing ideas
too
once billing server has not the same public IP than Asterisk server. I
ll
appreciate your comments ok.

  

I use ser+radius to do authentication, this way I can disable users or 
groups of users in a standard way, without using tricks like changing 
passwords.
(when your customer pays he expect to have the same password as before, 
have you saved it ? where ? in a safe way ?)
radius has a mysql backend, so also no need to reload config files.
Asterisk and radius share the same db, with some not-too-complex agi 
before the actual Dial you can do stuff like setting the call timeout 
based on the remaining credit, blocking the call if the credit is too 
much in the red, and so on...
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Checked by AVG Anti-Virus.
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RE: [Asterisk-Users] How to connect many analog lines to Asterisk?

2005-09-07 Thread Darren Wright

Wow, first of all, if you have a hundred analog lines, you are doing
yourself a disservice.a 4 T1's would be much much cheaper, and much
easier to manage.

Anyway, for 100 analog lines, I'd get 3 adit 600 channel banks and fill
them with FXO cards, and then buy 1 TE406 Quad T1 card for your asterisk
box.

-Darren
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Josip
Gracin
Sent: Wednesday, September 07, 2005 5:20 AM
To: [EMAIL PROTECTED]: [Asterisk-Users] How to
connect many analog lines to Asterisk?

Hello!

If I have more than a hundred analog telephones (analog lines) that need

to be connected to Asterisk PBX, what kind of hardware do I need, and 
where can I buy it?

Thanks in advance!
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[Asterisk-Users] Extensions - Realtime

CVS HEAD/Asterisk 1.2: Is there a way to have the entire extensions.conf file coming from the realtime? It appears that RealTimefor the extensions.conf file is on a context by context basis, but you have to create each new context in the
extensions.conf file then add a switch = Realtime line (then reload). I want to be able to add phones without having to edit any files. I'm using a PHP w/MySQLAGI app to route my calls and all this would all be find except for blind transfers as I could use the channel variable itself to determine the source of the call. But, the blind transfer comes into the dialplanon the channel of the transferree with the context of the transferer. So, I need to have each phone in it's own context to determine what channel is actually requesting the transfer (forcdr, parking, routing, etc.reasons). As such, I have to create a new context each time I add a phone.
Idon't mind reloading so much and it looks like I'm not going to be able to avoid that anyways with the SIP RealTime cached (--oxymoron)for MWI. The reason I don't want to edit files is that I'm sharing the dialplan between multiple boxes (the PHP app takes care of figuring out which box it is). I don't want to have to a. have to save the file on each box or b. map the files between boxes.
!!!- Alternatively, if there is a way to determine in the dialplan who is the transferer without having each phone in it's own context, that would be fine. -- Automated Signature: This message is from Flobi of
Flobi.com.Visit my website if you like: http://www.flobi.com/Please remember to tip your waitress and bartender.They are doing their best to serve you and your indignant, malcontent attitude.
--
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Re: [Asterisk-Users] channels VHF/ HF radio in asterisk

2005-09-07 Thread makevuy



Is then possible using app_rpt solution for both VHF and HF channels?

Regards.

Mark Phillips escribió:


2 ways.

1) buy into the app_rpt system. They have a bespoke card for your PC 
that can drive radio's etc. It's mainly aimed at repeater owners.


2) connect a phone patch between an ATA and your HF rig. This solution 
is currently being used to provied phone services from a few Red Cross 
shelters to the ARC HQ in Montgomery, AL. It works well.


Mark, KC2ENI

makevuy wrote:


Hy,
I have a network with WIFI communication and VHF/ HF channels.
I have integrated asterisk in the network using SIP, ZAP and IAX2 
channels for WIFI communications, but I don't Know How I could 
integrate the VHF/ HF channels.


I have heard speaking about app_rpt project, but I don't Know very 
much about this.


Could I integrate VHF/ HF channels with this application? if the 
answer is yes, How?


Regards.

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[Asterisk-Users] Re: ISDN PBX integration

2005-09-07 Thread Shahar Livne

Well, I just answer myself here:

Since the ISDN PBX is just the same as ISDN phone as far as the
asterisk is concerned, NT mode on the ISDN card should be used as
well.

The difference is that the phone uses p2mp (point to multi point)
protocol, as the PBX uses p2p (point to point) protocol.
Using the bri_net signalling instead of bri_net_ptmp solves the problem.

Shahar

On 9/7/05, Shahar Livne [EMAIL PROTECTED] wrote:
 Hello list,
 
 I am trying to connect an old ISDN PBX to my asterisk system.
 The setup includes an asterisk (1.0.9) running on the Soekris
 hardware, with an ISDN card (Billion BIPAC PCI), and I run
 zaphfc-bristuff-0.2.0-RC8k kernel module in NT mode (modes=1).
 
 When I connect an ISDN phone to the card (using cross ISDN cable + 40v
 power supply), I manage to make calls from and to the asterisk.
 
 When I try to connect the ISDN PBX instead, still using the same cross
 ISDN cable and the power supply, I get rings to the right extensions
 of the ISDN PBX, but no call setup happens, and after 2 rings comes
 silence.
 
 I have tried this with NT and TE modes of the zaphfc kernel module (I
 thought it has to do with point-to-point or point-to-multipoint
 issue). I have tried it both with cross ISDN cable and normal ISDN
 cable (I just used Ethernet cable), all without any success (not even
 the mentioned rings).
 
 When this ISDN PBX is connected to normal ISDN line, it just works.
 
 If there is anyone with some knowledge about ISDN equipment and its
 integration with asterisk – I will be happy to get some advice – what
 to check or how is it actually supposed to work.
 
 Thanks,
 Shahar
 
 --
 Shahar Livne
 LivneX - Open Source Development and Services
 http://LivneX.com
 


-- 
Shahar Livne
LivneX - Open Source Development and Services
http://LivneX.com
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RE: [Asterisk-Users] channels VHF/ HF radio in asterisk

2005-09-07 Thread Jonathan k. Creasy

The VHF or HF is determined by the radio equipment you have attached, not the 
software. 

-Jonathan

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of makevuy
Sent: Wednesday, September 07, 2005 10:01 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] channels VHF/ HF radio in asterisk

Is then possible using app_rpt solution for both VHF and HF channels?

Regards.

Mark Phillips escribió:

 2 ways.

 1) buy into the app_rpt system. They have a bespoke card for your PC
 that can drive radio's etc. It's mainly aimed at repeater owners.

 2) connect a phone patch between an ATA and your HF rig. This solution
 is currently being used to provied phone services from a few Red Cross 
 shelters to the ARC HQ in Montgomery, AL. It works well.

 Mark, KC2ENI

 makevuy wrote:

 Hy,
 I have a network with WIFI communication and VHF/ HF channels. I have 
 integrated asterisk in the network using SIP, ZAP and IAX2 channels 
 for WIFI communications, but I don't Know How I could integrate the 
 VHF/ HF channels.

 I have heard speaking about app_rpt project, but I don't Know very
 much about this.

 Could I integrate VHF/ HF channels with this application? if the
 answer is yes, How?

 Regards.

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Re: [Asterisk-Users] Which Linux distribution?

2005-09-07 Thread [EMAIL PROTECTED]

I use Centos 3.5 with great success.  It is a RHEL3 binary compatible clone.
Don't know if I would use development cutting edge software in the
enterprise.



--- John Daragon [EMAIL PROTECTED] wrote:
 Tzafrir Cohen wrote:
  On Wed, Sep 07, 2005 at 10:10:05AM +0100, John Daragon wrote:
  
 YT Lim wrote:
 
 We have tried Asterisk 1.0.9 on FC4 and have never
 been able to get CAPI (with Fritz card, fcpci) to work
 properly. Apart from that Asterisk works fine in
 switching internal calls. But it's useless if we can't
 make outgoing calls on our ISDN line.
 
 We are considering abandoning FC4 for Debian or SuSe.
 What is the general concensus on the best Linux to run
 Asterisk with CAPI?
 
 SUSE (as far as I know) is the only distro that really *expects* you to 
 be using ISDN2e as a matter of course.
  
  
  Only Linux distro that is generally something that is a bit hasty to
  say, given the fact that there are so many of them ;-) .
 
 You're absolutely right.
 
  Mandrake is quite Europe-centric as well. I'm not sure about ISDN
  support. 
 
 It's shipped with the packages; I looked at it when I first started 
 installing *, but couldn't get fcpci to work at the time.  CAPI appears 
 to have been written on (or for) SUSE in the first place, and SUSE was 
 the first distro I came across that supported ISDN2e out of the box.
 
  
  Debian has generally a large european installed base and a variety of
  ISDN-related packages as a result.
  
  Sorry, I won't make your life easier :-p
  
 
 You mean it's *supposed* to be easy ?
 
 jd
 -- 
 
 John Daragon  [EMAIL PROTECTED]
 argv[0] limited
 Lambs Lawn Cottage,  Staple Fitzpaine,  Taunton,  TA3 5SL,  UK
 v +44 (0) 1460 234068   f +44 (0) 1460 234069   m +44 (0) 7836 576127
 
 
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[Asterisk-Users] CONNECT ACK timeout in libpri

2005-09-07 Thread gshaw


Hi

I am testing a voip gateway product with Asterisk. We are experiencing
CONNECT ACK timer (T313) timing out on the Asterisk side when an incoming
call is received on the T1-PRI interface. The call is immediately routed to
voice mail.

This doesn't happen if I connect another PRI test equipment to the voip
gateway. The T313 timer is defined as 4000 mS however we see libpri
complaining (i.e., timing out) immediately after sending CONNECT to the voip
gateway. It seems it is not waiting for 4 seconds as before timing out.

Any ponters would be appreciated.

Regards
GS


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RE: [Asterisk-Users] Extensions - Realtime

2005-09-07 Thread Robert Bedell









Ive got some modifications Ive
made to asterisk that create a global switch. It essentially
just adds a check to the end of pbx_find_extension() that will try to look the
extension up in the database if its not found in one of the includes or in
any of the switches attached to the context itself.



Its rather hackish (it uses a
global context not linked in with the regular context list), and so probably
has some issues, but I can clean it up and post the patch somewhere if others
are interested. It sounds like you would be.



Cheers!



Robert Bedell











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Flobi
Sent: Wednesday, September 07,
2005 9:57 AM
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users]
Extensions - Realtime





CVS HEAD/Asterisk 1.2: Is there a way to have
the entire extensions.conf file coming from the realtime? 

It appears that RealTimefor the extensions.conf file is on a context by
context basis, but you have to create each new context in the extensions.conf
file then add a switch = Realtime line (then reload). I
want to be able to add phones without having to edit any files. 

I'm using a PHP w/MySQLAGI app to route my calls and all this would
all be find except for blind transfers as I could use the channel variable
itself to determine the source of the call. But, the blind transfer comes
into the dialplanon the channel of the transferree with the context of
the transferer. So, I need to have each phone in it's own context to determine
what channel is actually requesting the transfer (forcdr, parking,
routing, etc.reasons). As such, I have to create a new context each
time I add a phone. 

Idon't mind reloading so much and it looks like I'm not going to be able
to avoid that anyways with the SIP RealTime cached (--oxymoron)for
MWI. The reason I don't want to edit files is that I'm sharing the
dialplan between multiple boxes (the PHP app takes care of figuring out which
box it is). I don't want to have to a. have to save the file on each box
or b. map the files between boxes. 

!!!- Alternatively, if there is a way to determine in the dialplan who is
the transferer without having each phone in it's own context, that would be
fine. 
-- 
Automated Signature: This message is from Flobi of Flobi.com.
Visit my website if you like: http://www.flobi.com/

Please remember to tip your waitress and bartender.They are doing
their best to serve you and your indignant, malcontent attitude. 
-- 






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RE: [Asterisk-Users] PHP and ASterisk Manager

2005-09-07 Thread Anton Krall

I fixed the problem using preg_replace but you are right, I completely
forgot We are using open source ! :) silly of me, I should have checked
that.

Thx for reopening my eyes Christoph

 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Christoph Eicke
|Sent: Miércoles, 07 de Septiembre de 2005 02:12 a.m.
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] PHP and ASterisk Manager
|
|I looked into the source code of Asterisk to figure out how 
|the printf() statements were spaced. That's the power of open 
|source, you can look under the hood for these questions. It's 
|easy to find, even for non-C-Gurus. Just do a grep for the 
|string that you want inside of the Asterisk source directory 
|and it will give you the file that the string you are looking 
|for is in. Then simply open the file, search for the string 
|and look at the
|printf() statement.
|
|Christoph
|
|On Tuesday 06 September 2005 21:16, Anton Krall wrote:
| I was able to do and if and while loops to get the block of 
|lines I want..
| Now.. Another issue.
|
| I need to parse the line read to insert it into a table but seems 
| Asterisk inserts TABS or SPACES inconsistantly.. For example:
|
| Xxx(TAB)xxx(5 spaces)xxx
| Next line
| Xxx(TAB)xxx(3 spaces)xxx
|
| Im having a hard time figuring out how Asterisk Manager returns the 
| stuff
| :)
|
| Well..s o far so good...
|
| |-Original Message-
| |From: [EMAIL PROTECTED]
| |[mailto:[EMAIL PROTECTED] On Behalf 
|Of Matthew 
| |Boehm
| |Sent: Martes, 06 de Septiembre de 2005 01:49 p.m.
| |To: Asterisk Users Mailing List - Non-Commercial Discussion
| |Subject: Re: [Asterisk-Users] PHP and ASterisk Manager
| |
| |Anton Krall wrote:
| | Guys, is anybody using PHP sockets to connect to the Manager
| |
| |and send
| |
| | command like show voicemail users for example or any other?
| |
| | My question is, how to parse the return info in a way that can be 
| | shown back to the user via web (discard all the manager
| |
| |responses not needed)?
| |
| |Use preg_match() to match the lines you want the user to see on the 
| |website.
| |
| |$socket = fsockopen(localhost,5038, $errno, $errstr, 30);
| |
| |if(!$socket) {
| | print No socket;
| |exit();
| |}
| |
| |fputs($socket, Action: Login\r\n);
| |fputs($socket, Events: Off\r\n);
| |fputs($socket, UserName: bleh\r\n); fputs($socket, Secret: 
| |bleh\r\n\r\n);
| |
| |fputs($socket, Action: Command\r\n); fputs($socket, 
|Command: show 
| |channels\r\n\r\n);
| |
| |fputs($socket, Action: Logoff\r\n\r\n);
| |
| |while(!feof($socket)) {
| | $buff = fgets($socket,1024);
| | if(preg_match(/SIP\/.*/, $buff)) {
| |print I found a SIP call;
| | }
| |}
| |
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Re: [Asterisk-Users] Extensions - Realtime

2005-09-07 Thread Matthew Boehm

 CVS HEAD/Asterisk 1.2: Is there a way to have the entire
 extensions.conffile coming from the realtime?

Yes. Go read the wiki on RealTime Static.

-Matthew


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Re: [Asterisk-Users] TDM400 and Phone does not 'ring'

2005-09-07 Thread Wilson Pickett

 1 phone 'ringes' a bit cripled (instead of ring-ring... ring-ring..., it
   does  'ring-ri... ri ring... ri...)
 and the 3rd one does not ring at all when Asterisk says 'Ringing Zap/6'.
 However, when I do an 'off-hook' on this phone, I get tone signal and
 can dial and talk perfectly.
 
 I have phones compliant to the Belgium (Belgacom) Telco specs.
 Are there differences in 'Ring Voltage' ?

There is an issue here in France with our Siemens DECT phones that
required a patch to change the ring _frequency_. It was given here
ages ago, but now I can't find it. I believe it is still in bugs. It
requires a change to one line in wcfxs.c
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Re: [Asterisk-Users] Speex codec - Out of buffer space


 I'm running Asterisk 1.0.7 and would like to add Speex support. I
 downloaded Speex 1.0.5, installed and recompile Asterisk again.
 
 Now trying from X-Lite to connect using Speex but getting lot of weird
 erros on Asterisk console:
 
 Sep  7 15:03:25 WARNING[28605]: codec_speex.c:166 speextolin_framein:
 Out of buffer space
 
 I was trying to setup Speex on my second Asterisk server and wanted to
 use this codec for IAX between these two boxes. But I'm getting unable
 to negotiate codecs. Other codecs works like a charm. 

v1.0.7 is pretty old. Current cvs-head has speex built in. Would suggest
upgrading asterisk code.


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[Asterisk-Users] PBX Replacement

2005-09-07 Thread Sean Cook

I am getting ready to spec out a replacement for a Merlin Legend system
with asterisk.  There are a couple of things that holding me up that
hopefully someone here can answer.

1.  How well do modems work through a channel back to a PRI/T1
interface?

2.  Is there a decent receptionist phone (I don't want to use FOP) to
replace the system that our receptionists are already familiar with?  I
know that there is some discussion on the cisco 7914s and the SNom side
car... but do they work?  and how well...

3.  Network suggestions... this is more open ended.. we are currently
100mb to the desktop, however the switches are garbage and I have heard
that it is best to vlan the voice traffic away from the data traffic...
thoughts?


Regards,

Sean Cook
Network Engineer
Kinex Networking

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Re: [Asterisk-Users] How to connect many analog lines to Asterisk?

2005-09-07 Thread C F

I don't know why Darren syas 3 Adits since each one can handle 48
FXO/FXS channels, so 2 make 96. Anyhow each Adit connects to 2 T1
ports on a TE405/6. With Adit 600 I don't see why TE406 is required
since I believe the Adit 600 will take care of the echo, I might be
wrong on this last one about the echo.
In any case, if what you want is FXS cards, then I would strongly
recommend to get Adit 600 with CMG cards, then you can get by with a
$400 PC for asterisk, since you don't need any digium cards for that,
because the CMG in the Adit will handle the main load of transcoding.
The one limitation is, that if you need faxing, or FXO ports then
don't go with the CMG. It might still be worthwhile to get a single
span T1 card from Digium, use an Adit to cross connect the FXS
cards/channels that are connected to fax machines to that T1, as well
as any FXO cards that need to go to asterisk.

On 9/7/05, Darren Wright [EMAIL PROTECTED] wrote:
 Wow, first of all, if you have a hundred analog lines, you are doing
 yourself a disservice.a 4 T1's would be much much cheaper, and much
 easier to manage.
 
 Anyway, for 100 analog lines, I'd get 3 adit 600 channel banks and fill
 them with FXO cards, and then buy 1 TE406 Quad T1 card for your asterisk
 box.
 
 -Darren
 
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Josip
 Gracin
 Sent: Wednesday, September 07, 2005 5:20 AM
 To: [EMAIL PROTECTED]: [Asterisk-Users] How to
 connect many analog lines to Asterisk?
 
 Hello!
 
 If I have more than a hundred analog telephones (analog lines) that need
 
 to be connected to Asterisk PBX, what kind of hardware do I need, and
 where can I buy it?
 
 Thanks in advance!
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Re: [Asterisk-Users] PRI in and out

2005-09-07 Thread C F

Sorry my mistake. The span to provider is pri_cpe, and the span to the
avaya is pri_net.

On 9/7/05, Rod Bacon [EMAIL PROTECTED] wrote:
 It DOES help, thanks.
 
 Except for this
 
 
  the only difference between the first set of channels (1-23) and the
  second set of channels (25-47) is:
  signalling=pri_net
  group=1
  context = fromprovider
  channel = 1-23
  signalling = pri_cpe
  group=2
  context=fromavaya
  channel=25-47
 
 I thought the signalling setting was from the perspective of the * server, not
 the other side. For example, my PRIs to my provider are configured as pri_cpe,
 as I am the CPE.
 
 Your example seems to suggest the other way around.
 

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Re: [Asterisk-Users] Extensions - Realtime

It states that the conf file overrides the static db info, but what about the ael file? Does that override also? 

BTW, RealTime Static...talk about oxymoron :-) Gotta love it!
Flobi
On 9/7/05, Matthew Boehm [EMAIL PROTECTED] wrote:
 CVS HEAD/Asterisk 1.2: Is there a way to have the entire extensions.conffile coming from the realtime?
 Yes. Go read the wiki on RealTime Static.-Matthew___--Bandwidth and Colocation sponsored by Easynews.com --
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[Asterisk-Users] ArtDio IPF-2000 unable to send audio to Cisco 7940 until placed on hold and resumed

2005-09-07 Thread Michael Coburn









The issue appears to be between the Cisco 7940 and the
ArtDio IPF-2000, when a call is initiated between these phones the ArtDio phone
receives the audio stream fine from the Cisco, but the Cisco cant hear
anything from the ArtDio, until the Cisco user places the call on hold and then
picks the line back up. When the Cisco user places the call on hold,
music streams fine from the server to the ArtDio and is played.
Incoming/outgoing calls through the Asterisk server with the PSTN network are
flawless with either the ArtDio or Cisco phone, and calls between ArtDio-only
or Cisco-only are flawless. What gives??



I tried setting specific audio codecs in sip.conf for each
phone with disallow=all and then allow=ulaw, I set the SIPMAC.cnf file
for the Cisco phone to be 711u as the preferred codec, and through the web
interface configured the ArtDio to use 711u as its preferred codec, to
no avail. I tcpdumped the data during the call and set the ArtDio
phone to use RTP port and Control ports within the 17000  32000 range
(SIPmac.cnf sets the Cisco phones to use ports dynamically from this
range, based on tcpdump analysis) Were using Asterisk 1.0.9 (Xorcom
Rapid 1.1 latest release). 



I can post further configuration files if required. Any
pointers would be much appreciated!

--

Michael
 Coburn

Network Solutions Manager

MidWest Technical
Associates

c: 614-425-9203 

p: 614-336-3640 x501

f: 614-336-3645

www.midwesttechnical.net








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Re: [Asterisk-Users] How to connect many analog lines to Asterisk?

2005-09-07 Thread Josip Gracin


Darren Wright wrote:

Wow, first of all, if you have a hundred analog lines, you are doing
yourself a disservice.a 4 T1's would be much much cheaper, and much
easier to manage.


Let me clear this up a little bit.  There are hundreds of telephone 
devices inside the building, all connected to a PBX, and there is an 
E1/T1 connection to the PSTN (being statistically multiplexed, 
obviously).  What I'd like to do is to replace the PBX with Asterisk.


I don't see how I can make the situation better by using 4 T1's?


Anyway, for 100 analog lines, I'd get 3 adit 600 channel banks and fill
them with FXO cards, and then buy 1 TE406 Quad T1 card for your asterisk
box.


Thanks, I think that's what I need.
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Re: [Asterisk-Users] Occasional quiet voicemails

2005-09-07 Thread Anthony Rodgers

Indeed I do - but I read bug 2023 before posting and thought it was to 
do with the system-wide problem, not with occasional occurrences. I'll 
go back and read it again. Has the problem been solved with the 411P?


Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp

On Sep 6, 2005, at 7:32 PM, Rich Adamson wrote:


 Having solved a system-wide problem on 1.0.7 with low volume on
 voicemail messages by using format=wav, some users are still
 complaining that the occasional voicemail message (no apparent 
pattern

 in terms of call origination) is still so quiet as to be barely
 audible. Normal conversations and the majority of voicemail messages
 are fine.

 Has anyone else experienced something similar?

Yup. Bet you have an x100p or TDM card. See bug #2023 from a long
time ago.


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[Asterisk-Users] Re: Polycom 300 with latest 1.5.3 firmware not registering

2005-09-07 Thread Noah Miller


Hi Jorge -

I got 3 Polycom 300 phones, and upgraded to the latest firmware  
provided by the reseller.


This is my first experience with Polycom and I cannot make them  
register in my Asterisk Box.


I followed every advice I found (including separating [user] and  
[peer] in sip.conf.


Using ethereal, I found that it tries to SUBSCRIBE to the asterisk  
box and it receives a 403 FORBIDDEN message.


I compared to a Grandstream registration, and it tries to REGISTER  
to the asterisk receiving a 200 Message response and effectively  
registering.


Finding in the packet capture no other great difference, I believe  
that SUBSCRIBE requires a different authentification approach,  
maybe related to the voIpProt.SIP.requestValidation.digest.realm  
parameter in sip.cfg. I Tried the Polycom default, empty, default  
(voicemail context), from-internal (Extension context), the IP of  
the asterisk box, the name of the asterisk box, asterisk, etc, with  
no result. I tried different approaches documented in the wiki and  
related pages with no result. I can makke calls but I cannot  
receive them.



To start off, can you post your Polycom phone.cfg file and your  
asterisk sip.conf files?


Thanks,
Noah
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Re: [Asterisk-Users] Extensions - Realtime

Okay, this doesn't seem to be working.  I've gone and deleted my ael
file also.  I do know my MySQL is set up cause I have my sip, iax and
voicemail going through it too.

here's the line in extconfig.conf:
[settings]
extensions.conf = mysql,asterisk,pbx_realtime_extensions


in pbx_realtime_extensions, my db table:
id  name  context  exten  priority  app  appdata  
1  default  default  _.  1  NoOp  Testing 


CLI show dialplan
[ Context 'parkedcalls' created by 'res_features' ]
  '700' =  1. Park() [res_features]

-= 1 extensions (1 priorities) in 1 contexts. =-

And when I try to call, I get:
Sep  7 12:16:30 NOTICE[14330]: chan_iax2.c:7052 socket_read: Rejected
connect attempt from 206.180.254.97, request '[EMAIL PROTECTED]' does not
exist

Also, this message keeps popping up even when calls aren't going through:
Sep  7 12:15:12 NOTICE[14330]: pbx.c:1688 pbx_extension_helper: Cannot
find extension context 'default'


On 9/7/05, Flobi [EMAIL PROTECTED] wrote:
 
 It states that the conf file overrides the static db info, but what about the 
 ael file?  Does that override also?  
  
 BTW, RealTime Static...talk about oxymoron :-)  Gotta love it!
  
 Flobi
  
 
 On 9/7/05, Matthew Boehm [EMAIL PROTECTED] wrote: 
   CVS HEAD/Asterisk 1.2: Is there a way to have the entire
   extensions.conffile coming from the realtime? 
  
 Yes. Go read the wiki on RealTime Static.
  
  -Matthew
  
  
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Re: [Asterisk-Users] Lock Extension

2005-09-07 Thread Stephen


Hi Robert,

Do you have the sample script for locking the extension?

Thanks,
Stephen

Robert Goodyear wrote:


On Aug 18, 2005, at 3:07 AM, Stephen wrote:


Hi All,

How can I lock the extension in Asterisk?
For example , my extension is 1000 and I am away for business trip.  
I want to lock my extension during my absence.

Can it be done in Asterisk?

regards,
Stephen



You could write a little script to mangle/unmangle your SIP context  
and then SIP RELOAD. You could assign it to a context called  
'disabled' whose only valid extension matching therein is to that  
same macro to authenticate and change your context back.

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Re: [Asterisk-Users] Working example of ALERT_INFO with Cisco ATAs?

2005-09-07 Thread Brian Capouch


Olle E. Johansson wrote:



Try setting _ALERT_INFO



Worked perfectly, thanks.

B.
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[Asterisk-Users] Asterisk crashed?

2005-09-07 Thread Arik Funke


Hi,

I am running Asterisk in production mode but unfortunately every few 
days or so, it crashes, presumably...


Presumably because, when the phones stop working and I look for the 
cause, asterisk is no longer running. Asterisk logs and 
/var/log/messages contain no hints at all.


How can I get mode info on such unpredicable crashes?

Thanks in advance,
Arik
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[Asterisk-Users] Second Line does not Connect - HELP - misdn,sip

2005-09-07 Thread Pascal Speck










About my System:



2 * HFC Cards with misdn. 1 NT mode, 1 TE mode

1 * Sip-Provider (1und1)



On NT-Port  Ritto
(Elmeg) PBX

On TE-Port  NTBA



About my Problem: 



When a SIP-Call from a phone connected to the Ritto PBX is
in progress and someone calls on the ISDN-Line, the greeting works, and the phones
connected to the Ritto PBX are ringing. When I pick up a phone there is no connection
and the caller hears MOH all the time. This only happens when a second call is
in progress.

When no other call is in progress, everything works fine.



About my Configfiles:








extensions.conf









[incoming]

exten = ,1,Goto(anruferannahme,s,1)

exten = ,1,Goto(anruferannahme,s,1)

exten = 922xxx,1,Answer()

exten = 922xxx,2,Dial(misdn/2/922975) ; FAX

exten = 923xxx,1,Answer()

exten = 923xxx,2,Playback(thomas)

exten = 923xxx,3,Dial(misdn/2/9230250,,m) ; Thomas
Durchwahl

exten = 923xxx,1,Answer()

exten = 923xxx,2,Dial(misdn/2/9230251) ; Thomas FAX







[outgoing]





; Anwahl ber normale ISDN-Leitung:

exten = _999.,1,Answer()

exten = _999.,2,Dial(misdn/1/${EXTEN:3},,m)

exten = _999.,3,Playback(dialfailed)



; Faxe ber normalen ISDN-Anschluss verschicken:



exten = _X./922975,1,WaitforDigits(2000) ; mit
Vorwahl

exten = _X./922975,2,Answer()

exten = _X./922975,3,Dial(misdn/1/${EXTEN}) ; wenn IP
nich erfolgreich



; Telefongesprche bei denen die Vorwahl angegeben ist:



exten = _0X.,1,WaitforDigits(4000)

exten = _0X.,2,Answer()

exten = _0X.,3,Dial(SIP/[EMAIL PROTECTED])

exten = _0X.,4,Playback(nosip)

exten = _0X.,5,Dial(misdn/1/${EXTEN}) ; wenn IP nicht
erfolgreich

exten = _0X.,6,Playback(dialfailed)

exten = _0X.,104,Playback(besetzt)



; Telefongesprche bei denen die Vorwahl nicht
angegeben ist:



exten = _X.,1,WaitforDigits(4000)

exten = _X.,2,Answer()

exten = _X.,3,Dial(SIP/[EMAIL PROTECTED])

exten = _X.,4,Playback(nosip)

exten = _X.,5,Dial(misdn/1/${EXTEN})

exten = _X.,6,Playback(dialfailed)

exten = _X.,104,Playback(besetzt)



[aufnahme]

exten = s,1,Background(beep)

exten = 1,1,Record(/var/lib/asterisk/sounds/greeting:gsm)

exten = 2,1,Record(/var/lib/asterisk/sounds/besetzt:gsm)

exten =
3,1,Record(/var/lib/asterisk/sounds/aufnahme:gsm)





[anruferannahme]

exten = s,1,Answer()

exten = s,2,Background(greeting)

exten = s,3,Dial(misdn/2/,15,m)

;exten = s,4,WaitMusicOnHold(2)

;exten = s,5,Dial(misdn/2/9230255,15,m)

;exten = s,6,WaitMusicOnHold(2)

;exten = s,7,Dial(misdn/2/,100,m)

;exten = s,8,Playback(nichterr)

exten = s,4,Hangup()



exten = 7,1,Goto(aufnahme,s,1)







misdn.conf





[general]

context=vs

language=de

immediate=yes

debug=2

allow=alaw

musiconhold=default



[TEport]

context=incoming

ports=1

msns=*



[NTport]

context=outgoing

ports=2





sip.conf





[general]

port = 5060

bindaddr = 0.0.0.0

externip = myip

localnet = 192.168.0.0/255.255.0.0

context = default

srvlookup = yes

disallow = all

allow = ulaw

nat = yes



register = 492774:[EMAIL PROTECTED]/492774



[sip.1und1.de]

type=friend

username=492774

fromuser=492774

secret=mysecret

host=sip.1und1.de

context=incoming

fromdomain=1und1.de

qualify=no

insecure=very

canreinvite=no

nat=yes

allow=g726

dtmfmode=info








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[Asterisk-Users] ztcfg Kills My Dial Tone

2005-09-07 Thread Shaw Terwilliger

I'm using two Rhino channel banks (first 12FXO/12FXS, second 24FXS).
These connect to a Digium TE210P card.  I'm running kernel 2.6.10
and I've tried Asterisk (w/zaptel) 1.0.9, 1.2 beta, and CVS from today.
The results are the same for all versions:

Right after I reboot, and modprobe wct4xxp, my analog phone connected
to port 13 of the first channel bank (first FXS port) gets a dial tone.
Asterisk is not running yet, and I have NOT run ztcfg.

As soon as I run ztcfg, the port goes dead.  No dial tone, but I can
hear things I say into the phone's microphone come out of the speaker,
se there's voltage.  Starting Asterisk makes no difference.  The port is
dead until I unload and reload the wct4xxp module.

HOWEVER, when the port is dead, I can ring it from a SIP extension, and
the analog phone rings.  If I pick up the phone, Asterisk has no clue,
and keeps ringing forever.

My guess is this is a problem with the TE210P card or drivers.  

Any suggestions?

*** Here's my /etc/zaptel.conf (all non comment lines): ***

span=1,1,0,esf,b8zs
span=2,0,0,esf,b8zs
fxsks=1-12
fxoks=13-48
defaultzone = us
loadzone = us

*** Here's the output of ztcfg -vvv: ***

Zaptel Configuration
==

SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1)
SPAN 2: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1)

Channel map:

Channel 01: FXS Kewlstart (Default) (Slaves: 01)
Channel 02: FXS Kewlstart (Default) (Slaves: 02)
Channel 03: FXS Kewlstart (Default) (Slaves: 03)
Channel 04: FXS Kewlstart (Default) (Slaves: 04)
Channel 05: FXS Kewlstart (Default) (Slaves: 05)
Channel 06: FXS Kewlstart (Default) (Slaves: 06)
Channel 07: FXS Kewlstart (Default) (Slaves: 07)
Channel 08: FXS Kewlstart (Default) (Slaves: 08)
Channel 09: FXS Kewlstart (Default) (Slaves: 09)
Channel 10: FXS Kewlstart (Default) (Slaves: 10)
Channel 11: FXS Kewlstart (Default) (Slaves: 11)
Channel 12: FXS Kewlstart (Default) (Slaves: 12)
Channel 13: FXO Kewlstart (Default) (Slaves: 13)
Channel 14: FXO Kewlstart (Default) (Slaves: 14)
Channel 15: FXO Kewlstart (Default) (Slaves: 15)
Channel 16: FXO Kewlstart (Default) (Slaves: 16)
Channel 17: FXO Kewlstart (Default) (Slaves: 17)
Channel 18: FXO Kewlstart (Default) (Slaves: 18)
Channel 19: FXO Kewlstart (Default) (Slaves: 19)
Channel 20: FXO Kewlstart (Default) (Slaves: 20)
Channel 21: FXO Kewlstart (Default) (Slaves: 21)
Channel 22: FXO Kewlstart (Default) (Slaves: 22)
Channel 23: FXO Kewlstart (Default) (Slaves: 23)
Channel 24: FXO Kewlstart (Default) (Slaves: 24)
Channel 25: FXO Kewlstart (Default) (Slaves: 25)
Channel 26: FXO Kewlstart (Default) (Slaves: 26)
Channel 27: FXO Kewlstart (Default) (Slaves: 27)
Channel 28: FXO Kewlstart (Default) (Slaves: 28)
Channel 29: FXO Kewlstart (Default) (Slaves: 29)
Channel 30: FXO Kewlstart (Default) (Slaves: 30)
Channel 31: FXO Kewlstart (Default) (Slaves: 31)
Channel 32: FXO Kewlstart (Default) (Slaves: 32)
Channel 33: FXO Kewlstart (Default) (Slaves: 33)
Channel 34: FXO Kewlstart (Default) (Slaves: 34)
Channel 35: FXO Kewlstart (Default) (Slaves: 35)
Channel 36: FXO Kewlstart (Default) (Slaves: 36)
Channel 37: FXO Kewlstart (Default) (Slaves: 37)
Channel 38: FXO Kewlstart (Default) (Slaves: 38)
Channel 39: FXO Kewlstart (Default) (Slaves: 39)
Channel 40: FXO Kewlstart (Default) (Slaves: 40)
Channel 41: FXO Kewlstart (Default) (Slaves: 41)
Channel 42: FXO Kewlstart (Default) (Slaves: 42)
Channel 43: FXO Kewlstart (Default) (Slaves: 43)
Channel 44: FXO Kewlstart (Default) (Slaves: 44)
Channel 45: FXO Kewlstart (Default) (Slaves: 45)
Channel 46: FXO Kewlstart (Default) (Slaves: 46)
Channel 47: FXO Kewlstart (Default) (Slaves: 47)
Channel 48: FXO Kewlstart (Default) (Slaves: 48)

48 channels configured.

*** Here are the kernel messages emitted when I run ztcfg -vvv: ***

Sep  7 11:46:22 localhost kernel: About to enter spanconfig!
Sep  7 11:46:22 localhost kernel: About to enter startup!
Sep  7 11:46:22 localhost kernel: wct2xxp: Setting yellow alarm on span 1
Sep  7 11:46:22 localhost kernel: Zaptel: Master changed to TE2/0/2
Sep  7 11:46:22 localhost kernel: TE2XXP: Span 1 configured for ESF/B8ZS
Sep  7 11:46:22 localhost kernel: Putting 0 in register 2f on span 1
Sep  7 11:46:22 localhost kernel: Putting 0 in register 30 on span 1
Sep  7 11:46:22 localhost kernel: Putting 0 in register 31 on span 1
Sep  7 11:46:22 localhost kernel: SPAN 1: Primary Sync Source
Sep  7 11:46:22 localhost kernel: Completed startup!
Sep  7 11:46:22 localhost kernel: About to enter spanconfig!
Sep  7 11:46:22 localhost kernel: About to enter startup!
Sep  7 11:46:22 localhost kernel: wct2xxp: Setting yellow alarm on span 2
Sep  7 11:46:22 localhost kernel: TE2XXP: Span 2 configured for ESF/B8ZS
Sep  7 11:46:22 localhost kernel: Putting 0 in register 2f on span 2
Sep  7 11:46:22 localhost kernel: Putting 0 in register 30 on span 2
Sep  7 11:46:22 localhost kernel: Putting 0 in register 31 on span 2
Sep  7 11:46:22 localhost kernel: Completed startup!
Sep  7 11:46:22 localhost

Re: [Asterisk-Users] Extensions - Realtime

Okay, after noticing an error on this mysql statement after i switched to odbc:
SELECT * FROM 
pbx_realtime_extensions 
WHERE filename='extensions.conf' and commented=0 
ORDER BY filename,cat_metric desc,var_metric asc,category,var_name,var_val,id

I added those fields and reloaded...* immediately crashed.  I
restarted.  Now, I'm getting this:
*CLI show dialplan
[ Context 'NoOp' created by 'pbx_config' ]

[ Context 'parkedcalls' created by 'res_features' ]
  '700' =  1. Park() [res_features]

-= 1 extensions (1 priorities) in 2 contexts. =-


out of this table:
  id  name  context  exten  priority  app  appdata  filename 
commented  cat_metric  var_metric  category  var_name  var_val
  1 default default _. 1 NoOp Testing. extensions.conf 0 NULL NULL
NULL NULL NULL


On 9/7/05, Flobi [EMAIL PROTECTED] wrote:
 Okay, this doesn't seem to be working.  I've gone and deleted my ael
 file also.  I do know my MySQL is set up cause I have my sip, iax and
 voicemail going through it too.
 
 here's the line in extconfig.conf:
 [settings]
 extensions.conf = mysql,asterisk,pbx_realtime_extensions
 
 
 in pbx_realtime_extensions, my db table:
 id  name  context  exten  priority  app  appdata
 1  default  default  _.  1  NoOp  Testing
 
 
 CLI show dialplan
 [ Context 'parkedcalls' created by 'res_features' ]
  '700' =  1. Park() 
 [res_features]
 
 -= 1 extensions (1 priorities) in 1 contexts. =-
 
 And when I try to call, I get:
 Sep  7 12:16:30 NOTICE[14330]: chan_iax2.c:7052 socket_read: Rejected
 connect attempt from 206.180.254.97, request '[EMAIL PROTECTED]' does not
 exist
 
 Also, this message keeps popping up even when calls aren't going through:
 Sep  7 12:15:12 NOTICE[14330]: pbx.c:1688 pbx_extension_helper: Cannot
 find extension context 'default'
 
 
 On 9/7/05, Flobi [EMAIL PROTECTED] wrote:
 
  It states that the conf file overrides the static db info, but what about 
  the ael file?  Does that override also?
 
  BTW, RealTime Static...talk about oxymoron :-)  Gotta love it!
 
  Flobi
 
 
  On 9/7/05, Matthew Boehm [EMAIL PROTECTED] wrote:
CVS HEAD/Asterisk 1.2: Is there a way to have the entire
extensions.conffile coming from the realtime?
  
  Yes. Go read the wiki on RealTime Static.
  
   -Matthew
  
  
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Re: [Asterisk-Users] Extensions - Realtime

Nevermind, I figured out that the table is used way differently when
doing static.  Here's my fixed table.  I'll try to explain this in the
voip-info doc.

  id  cat_metric  var_metric  commented  filename  category  var_name  var_val  
  1 0 0 0 extensions.conf default exten _.,1,NoOp(Testing) 


On 9/7/05, Flobi [EMAIL PROTECTED] wrote:
 Okay, after noticing an error on this mysql statement after i switched to 
 odbc:
 SELECT * FROM
 pbx_realtime_extensions
 WHERE filename='extensions.conf' and commented=0
 ORDER BY filename,cat_metric desc,var_metric asc,category,var_name,var_val,id
 
 I added those fields and reloaded...* immediately crashed.  I
 restarted.  Now, I'm getting this:
 *CLI show dialplan
 [ Context 'NoOp' created by 'pbx_config' ]
 
 [ Context 'parkedcalls' created by 'res_features' ]
  '700' =  1. Park() 
 [res_features]
 
 -= 1 extensions (1 priorities) in 2 contexts. =-
 
 
 out of this table:
  id  name  context  exten  priority  app  appdata  filename
 commented  cat_metric  var_metric  category  var_name  var_val
  1 default default _. 1 NoOp Testing. extensions.conf 0 NULL NULL
 NULL NULL NULL
 
 
 On 9/7/05, Flobi [EMAIL PROTECTED] wrote:
  Okay, this doesn't seem to be working.  I've gone and deleted my ael
  file also.  I do know my MySQL is set up cause I have my sip, iax and
  voicemail going through it too.
 
  here's the line in extconfig.conf:
  [settings]
  extensions.conf = mysql,asterisk,pbx_realtime_extensions
 
 
  in pbx_realtime_extensions, my db table:
  id  name  context  exten  priority  app  appdata
  1  default  default  _.  1  NoOp  Testing
 
 
  CLI show dialplan
  [ Context 'parkedcalls' created by 'res_features' ]
   '700' =  1. Park() 
  [res_features]
 
  -= 1 extensions (1 priorities) in 1 contexts. =-
 
  And when I try to call, I get:
  Sep  7 12:16:30 NOTICE[14330]: chan_iax2.c:7052 socket_read: Rejected
  connect attempt from 206.180.254.97, request '[EMAIL PROTECTED]' does not
  exist
 
  Also, this message keeps popping up even when calls aren't going through:
  Sep  7 12:15:12 NOTICE[14330]: pbx.c:1688 pbx_extension_helper: Cannot
  find extension context 'default'
 
 
  On 9/7/05, Flobi [EMAIL PROTECTED] wrote:
  
   It states that the conf file overrides the static db info, but what about 
   the ael file?  Does that override also?
  
   BTW, RealTime Static...talk about oxymoron :-)  Gotta love it!
  
   Flobi
  
  
   On 9/7/05, Matthew Boehm [EMAIL PROTECTED] wrote:
 CVS HEAD/Asterisk 1.2: Is there a way to have the entire
 extensions.conffile coming from the realtime?
   
   Yes. Go read the wiki on RealTime Static.
   
-Matthew
   
   
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Re: [Asterisk-Users] How to connect many analog lines to Asterisk?

2005-09-07 Thread Francesco Peeters

On Wed, September 7, 2005 18:11, Josip Gracin said:
 Darren Wright wrote:
 Wow, first of all, if you have a hundred analog lines, you are doing
 yourself a disservice.a 4 T1's would be much much cheaper, and much
 easier to manage.

 Let me clear this up a little bit.  There are hundreds of telephone
 devices inside the building, all connected to a PBX, and there is an
 E1/T1 connection to the PSTN (being statistically multiplexed,
 obviously).  What I'd like to do is to replace the PBX with Asterisk.

 I don't see how I can make the situation better by using 4 T1's?


You said you had 100 analog lines... What you meant is you have 100 analog
phones... Big difference... (OTOH, only a single letter: FXO - FXS)  ;-)

But seriously, there really is a big difference whether you are trying to
connect 100 analog lines (i.o.w. 100 incoming POTS lines from the PSTN) or
100 analog phones... If you had 100 incoming POTS lines, 4 PRI spans would
be way cheaper and way easier, hence the advice!

 Anyway, for 100 analog lines, I'd get 3 adit 600 channel banks and fill
 them with FXO cards, and then buy 1 TE406 Quad T1 card for your asterisk
 box.

 Thanks, I think that's what I need.

If you want 100 analog phones, make sure you get FXS cards in stead of FXO
cards... FXO cards are for incoming lines, FXS for phones...

(FXO stands for Foreign Exchange Office, ie PABX or PSTN, FXS for Foreign
Exchange Subscriber, ie telephones)

HTH

-- 
Francesco Peeters

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RE: [Asterisk-Users] How to connect many analog lines to Asterisk?

2005-09-07 Thread Jonathan k. Creasy

You asked how to connect lines, so he answered that question. The answer
is basically the same just change the FXO in the channel bank to FXS. 
-Jonathan

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Josip
Gracin
Sent: Wednesday, September 07, 2005 12:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] How to connect many analog lines to
Asterisk?

Darren Wright wrote:
 Wow, first of all, if you have a hundred analog lines, you are doing 
 yourself a disservice.a 4 T1's would be much much cheaper, and 
 much easier to manage.

Let me clear this up a little bit.  There are hundreds of telephone 
devices inside the building, all connected to a PBX, and there is an 
E1/T1 connection to the PSTN (being statistically multiplexed, 
obviously).  What I'd like to do is to replace the PBX with Asterisk.

I don't see how I can make the situation better by using 4 T1's?

 Anyway, for 100 analog lines, I'd get 3 adit 600 channel banks and 
 fill them with FXO cards, and then buy 1 TE406 Quad T1 card for your 
 asterisk box.

Thanks, I think that's what I need.
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Re: [Asterisk-Users] Extensions - Realtime

2005-09-07 Thread Matthew Boehm

I don't see your swich statement anywhere.

You must define a context [default] then add in the correct switch=
statement.

-Matthew


 From: Flobi [EMAIL PROTECTED]
 Reply-To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial
 Discussion asterisk-users@lists.digium.com
 Date: Wed, 7 Sep 2005 12:18:26 -0400
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Subject: Re: [Asterisk-Users] Extensions - Realtime
 
 Okay, this doesn't seem to be working.  I've gone and deleted my ael
 file also.  I do know my MySQL is set up cause I have my sip, iax and
 voicemail going through it too.
 
 here's the line in extconfig.conf:
 [settings]
 extensions.conf = mysql,asterisk,pbx_realtime_extensions
 
 
 in pbx_realtime_extensions, my db table:
 id  name  context  exten  priority  app  appdata
 1  default  default  _.  1  NoOp  Testing
 
 
 CLI show dialplan
 [ Context 'parkedcalls' created by 'res_features' ]
   '700' =  1. Park()
 [res_features]
 
 -= 1 extensions (1 priorities) in 1 contexts. =-
 
 And when I try to call, I get:
 Sep  7 12:16:30 NOTICE[14330]: chan_iax2.c:7052 socket_read: Rejected
 connect attempt from 206.180.254.97, request '[EMAIL PROTECTED]' does not
 exist
 
 Also, this message keeps popping up even when calls aren't going through:
 Sep  7 12:15:12 NOTICE[14330]: pbx.c:1688 pbx_extension_helper: Cannot
 find extension context 'default'
 
 
 On 9/7/05, Flobi [EMAIL PROTECTED] wrote:
 
 It states that the conf file overrides the static db info, but what about the
 ael file?  Does that override also?
  
 BTW, RealTime Static...talk about oxymoron :-)  Gotta love it!
  
 Flobi
  
 
 On 9/7/05, Matthew Boehm [EMAIL PROTECTED] wrote:
 CVS HEAD/Asterisk 1.2: Is there a way to have the entire
 extensions.conffile coming from the realtime?
 
Yes. Go read the wiki on RealTime Static.
 
 -Matthew
 
 
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Re: [Asterisk-Users] Extensions - Realtime

2005-09-07 Thread Matthew Boehm

The wiki doc's are correct. You are trying to combine two different methods
of pulling RealTime extensions and that is why it isn't working as you are
expecting.

Pick 1 method and all will be revealed. Both are very simple to do.

-Matthew

 From: Flobi [EMAIL PROTECTED]
 Reply-To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial
 Discussion asterisk-users@lists.digium.com
 Date: Wed, 7 Sep 2005 13:00:26 -0400
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Subject: Re: [Asterisk-Users] Extensions - Realtime
 
 Nevermind, I figured out that the table is used way differently when
 doing static.  Here's my fixed table.  I'll try to explain this in the
 voip-info doc.
 
   id  cat_metric  var_metric  commented  filename  category  var_name  var_val
   1 0 0 0 extensions.conf default exten _.,1,NoOp(Testing)
 
 
 On 9/7/05, Flobi [EMAIL PROTECTED] wrote:
 Okay, after noticing an error on this mysql statement after i switched to
 odbc:
 SELECT * FROM
 pbx_realtime_extensions
 WHERE filename='extensions.conf' and commented=0
 ORDER BY filename,cat_metric desc,var_metric asc,category,var_name,var_val,id
 
 I added those fields and reloaded...* immediately crashed.  I
 restarted.  Now, I'm getting this:
 *CLI show dialplan
 [ Context 'NoOp' created by 'pbx_config' ]
 
 [ Context 'parkedcalls' created by 'res_features' ]
  '700' =  1. Park()
 [res_features]
 
 -= 1 extensions (1 priorities) in 2 contexts. =-
 
 
 out of this table:
  id  name  context  exten  priority  app  appdata  filename
 commented  cat_metric  var_metric  category  var_name  var_val
  1 default default _. 1 NoOp Testing. extensions.conf 0 NULL NULL
 NULL NULL NULL
 
 
 On 9/7/05, Flobi [EMAIL PROTECTED] wrote:
 Okay, this doesn't seem to be working.  I've gone and deleted my ael
 file also.  I do know my MySQL is set up cause I have my sip, iax and
 voicemail going through it too.
 
 here's the line in extconfig.conf:
 [settings]
 extensions.conf = mysql,asterisk,pbx_realtime_extensions
 
 
 in pbx_realtime_extensions, my db table:
 id  name  context  exten  priority  app  appdata
 1  default  default  _.  1  NoOp  Testing
 
 
 CLI show dialplan
 [ Context 'parkedcalls' created by 'res_features' ]
  '700' =  1. Park()
 [res_features]
 
 -= 1 extensions (1 priorities) in 1 contexts. =-
 
 And when I try to call, I get:
 Sep  7 12:16:30 NOTICE[14330]: chan_iax2.c:7052 socket_read: Rejected
 connect attempt from 206.180.254.97, request '[EMAIL PROTECTED]' does not
 exist
 
 Also, this message keeps popping up even when calls aren't going through:
 Sep  7 12:15:12 NOTICE[14330]: pbx.c:1688 pbx_extension_helper: Cannot
 find extension context 'default'
 
 
 On 9/7/05, Flobi [EMAIL PROTECTED] wrote:
 
 It states that the conf file overrides the static db info, but what about
 the ael file?  Does that override also?
 
 BTW, RealTime Static...talk about oxymoron :-)  Gotta love it!
 
 Flobi
 
 
 On 9/7/05, Matthew Boehm [EMAIL PROTECTED] wrote:
 CVS HEAD/Asterisk 1.2: Is there a way to have the entire
 extensions.conffile coming from the realtime?
 
Yes. Go read the wiki on RealTime Static.
 
 -Matthew
 
 
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 http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [Asterisk-Users] How to connect many analog lines to Asterisk?

2005-09-07 Thread Josip Gracin


Jonathan k. Creasy wrote:

You asked how to connect lines, so he answered that question. The answer
is basically the same just change the FXO in the channel bank to FXS. 


Well, actually, I said: If I have more than a hundred analog telephones 
(analog lines) that need...  But, that doesn't help my case, does it? :-)


Anyway, thanks everybody for the info!
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RE: [Asterisk-Users] How to connect many analog lines to Asterisk?

2005-09-07 Thread Jonathan k. Creasy

Ohmy bad...I picked up the thread later :) 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Josip
Gracin
Sent: Wednesday, September 07, 2005 2:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] How to connect many analog lines to
Asterisk?


Jonathan k. Creasy wrote:
 You asked how to connect lines, so he answered that question. The 
 answer is basically the same just change the FXO in the channel bank 
 to FXS.

Well, actually, I said: If I have more than a hundred analog telephones

(analog lines) that need...  But, that doesn't help my case, does it?
:-)

Anyway, thanks everybody for the info!
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Re: [Asterisk-Users] Extensions - Realtime

It's not that, it's just that the wiki wasn't very clear on the fact
that all the tables for a static load had to be the same.  I had
thought that I was supposed to use the table on this page:
http://www.voip-info.org/tiki-index.php?page=Asterisk+RealTime+Extensions
even when doing realtime static, which isn't the case, I had to use
the table on 
http://www.voip-info.org/tiki-index.php?page=Asterisk+RealTime+Static
.

Also, I wasn't trying to say the wiki was incorrect, just a little
unclear.  I didn't change any info, just added some clarification for
those who might miss that part, like I did.

On 9/7/05, Matthew Boehm [EMAIL PROTECTED] wrote:
 The wiki doc's are correct. You are trying to combine two different methods
 of pulling RealTime extensions and that is why it isn't working as you are
 expecting.
 
 Pick 1 method and all will be revealed. Both are very simple to do.
 
 -Matthew
 
  From: Flobi [EMAIL PROTECTED]
  Reply-To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial
  Discussion asterisk-users@lists.digium.com
  Date: Wed, 7 Sep 2005 13:00:26 -0400
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  asterisk-users@lists.digium.com
  Subject: Re: [Asterisk-Users] Extensions - Realtime
 
  Nevermind, I figured out that the table is used way differently when
  doing static.  Here's my fixed table.  I'll try to explain this in the
  voip-info doc.
 
id  cat_metric  var_metric  commented  filename  category  var_name  
  var_val
1 0 0 0 extensions.conf default exten _.,1,NoOp(Testing)
 
 
  On 9/7/05, Flobi [EMAIL PROTECTED] wrote:
  Okay, after noticing an error on this mysql statement after i switched to
  odbc:
  SELECT * FROM
  pbx_realtime_extensions
  WHERE filename='extensions.conf' and commented=0
  ORDER BY filename,cat_metric desc,var_metric 
  asc,category,var_name,var_val,id
 
  I added those fields and reloaded...* immediately crashed.  I
  restarted.  Now, I'm getting this:
  *CLI show dialplan
  [ Context 'NoOp' created by 'pbx_config' ]
 
  [ Context 'parkedcalls' created by 'res_features' ]
   '700' =  1. Park()
  [res_features]
 
  -= 1 extensions (1 priorities) in 2 contexts. =-
 
 
  out of this table:
   id  name  context  exten  priority  app  appdata  filename
  commented  cat_metric  var_metric  category  var_name  var_val
   1 default default _. 1 NoOp Testing. extensions.conf 0 NULL NULL
  NULL NULL NULL
 
 
  On 9/7/05, Flobi [EMAIL PROTECTED] wrote:
  Okay, this doesn't seem to be working.  I've gone and deleted my ael
  file also.  I do know my MySQL is set up cause I have my sip, iax and
  voicemail going through it too.
 
  here's the line in extconfig.conf:
  [settings]
  extensions.conf = mysql,asterisk,pbx_realtime_extensions
 
 
  in pbx_realtime_extensions, my db table:
  id  name  context  exten  priority  app  appdata
  1  default  default  _.  1  NoOp  Testing
 
 
  CLI show dialplan
  [ Context 'parkedcalls' created by 'res_features' ]
   '700' =  1. Park()
  [res_features]
 
  -= 1 extensions (1 priorities) in 1 contexts. =-
 
  And when I try to call, I get:
  Sep  7 12:16:30 NOTICE[14330]: chan_iax2.c:7052 socket_read: Rejected
  connect attempt from 206.180.254.97, request '[EMAIL PROTECTED]' does not
  exist
 
  Also, this message keeps popping up even when calls aren't going through:
  Sep  7 12:15:12 NOTICE[14330]: pbx.c:1688 pbx_extension_helper: Cannot
  find extension context 'default'
 
 
  On 9/7/05, Flobi [EMAIL PROTECTED] wrote:
 
  It states that the conf file overrides the static db info, but what about
  the ael file?  Does that override also?
 
  BTW, RealTime Static...talk about oxymoron :-)  Gotta love it!
 
  Flobi
 
 
  On 9/7/05, Matthew Boehm [EMAIL PROTECTED] wrote:
  CVS HEAD/Asterisk 1.2: Is there a way to have the entire
  extensions.conffile coming from the realtime?
 
 Yes. Go read the wiki on RealTime Static.
 
  -Matthew
 
 
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Re: [Asterisk-Users] Asterisk crashed?

Which version of * are you using?  I had a problem with 1.0.7 crashing
unexplainably at one point, but 1.0.9 was out then and I upgraded and
it stopped.

On 9/7/05, Arik Funke [EMAIL PROTECTED] wrote:
 Hi,
 
 I am running Asterisk in production mode but unfortunately every few
 days or so, it crashes, presumably...
 
 Presumably because, when the phones stop working and I look for the
 cause, asterisk is no longer running. Asterisk logs and
 /var/log/messages contain no hints at all.
 
 How can I get mode info on such unpredicable crashes?
 
 Thanks in advance,
 Arik
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[Asterisk-Users] asterisk.org blocked - rejecting connections


Address lookup
canonical name  asterisk.org.
aliases 
addresses   216.27.40.102

Service scan
FTP - 21Error: TimedOut
SMTP - 25   Error: ConnectionRefused
HTTP - 80   Error: ConnectionRefused
POP3 - 110  Error: TimedOut
NNTP - 119  Error: TimedOut

digium.com is ok though

Address lookup
canonical name  digium.com.
aliases 
addresses   216.207.245.1


Service scan
FTP - 21Error: TimedOut
SMTP - 25   Error: TimedOut
HTTP - 80   HTTP/1.1 302 Found
Date: Wed, 07 Sep 2005 18:38:11 GMT
Server: Apache
X-Powered-By: PHP/4.3.10
Location: http://www.digium.com/
Connection: close
Content-Type: text/html; charset=ISO-8859-1
POP3 - 110  Error: TimedOut
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Re: [Asterisk-Users] asterisk.org blocked - rejecting connections

I'm not having any problems connecting to asterisk.org port 80.  

On 9/7/05, Martin [EMAIL PROTECTED] wrote:
 
 Address lookup
 canonical name  asterisk.org.
 aliases
 addresses   216.27.40.102
 
 Service scan
 FTP - 21Error: TimedOut
 SMTP - 25   Error: ConnectionRefused
 HTTP - 80   Error: ConnectionRefused
 POP3 - 110  Error: TimedOut
 NNTP - 119  Error: TimedOut
 
 digium.com is ok though
 
 Address lookup
 canonical name  digium.com.
 aliases
 addresses   216.207.245.1
 
 
 Service scan
 FTP - 21Error: TimedOut
 SMTP - 25   Error: TimedOut
 HTTP - 80   HTTP/1.1 302 Found
 Date: Wed, 07 Sep 2005 18:38:11 GMT
 Server: Apache
 X-Powered-By: PHP/4.3.10
 Location: http://www.digium.com/
 Connection: close
 Content-Type: text/html; charset=ISO-8859-1
 POP3 - 110  Error: TimedOut
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Re: [Asterisk-Users] Occasional quiet voicemails

I don't believe 2023 has anything to do with the 411P; it was basically
an digium analog card issue (eg, TDM04b  x100p).

Based on my tests and findings, the issue is the digium cards record
voicemail messages at a very low audio level (very different from
recording a voicemail from a sip phone). If the person leaving a VM
message called in via the digium card, and that user was located close
to their central office, the VM level is acceptable to poor. But, if
that same person is further from their central office (adding additional
transmission path loss), then that loss plus the digium analog card loss
makes the VM difficult if not impossible to hear.

So, thinking that statement through very carefully, you might have some
users complain and other not, and the problem will not track against
anything that you have control over (eg, where the remote user is calling
from and the transmission loss they incur).

If the digium analog cards passed audio through without any additional
loss, your user's probably would not be complaining. But that extra loss
is what I believe is the issue.

Sounds like there might be a workaround coming for this.

Rich


 Indeed I do - but I read bug 2023 before posting and thought it was to 
 do with the system-wide problem, not with occasional occurrences. I'll 
 go back and read it again. Has the problem been solved with the 411P?
 
 Regards,
 -- 
 Anthony Rodgers
 Business Systems Analyst
 District of North Vancouver
 Web: http://www.dnv.org
 RSS Feed: http://www.dnv.org/rss.asp
 
 On Sep 6, 2005, at 7:32 PM, Rich Adamson wrote:
 
   Having solved a system-wide problem on 1.0.7 with low volume on
   voicemail messages by using format=wav, some users are still
   complaining that the occasional voicemail message (no apparent 
  pattern
   in terms of call origination) is still so quiet as to be barely
   audible. Normal conversations and the majority of voicemail messages
   are fine.
  
   Has anyone else experienced something similar?
 
  Yup. Bet you have an x100p or TDM card. See bug #2023 from a long
  time ago.
 
 
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Re: [Asterisk-Users] asterisk.org blocked - rejecting connections

On Wednesday 07 September 2005 13:47, Flobi wrote:
 I'm not having any problems connecting to asterisk.org port 80.



They came up again. Finally.  That check wasn't from where I am but another 
location once I couldn't get onto the site.  Nothing more to see here...move 
on  ;-0

Regards...Martin
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Re: [Asterisk-Users] Occasional quiet voicemails

On Wednesday 07 September 2005 14:41, Rich Adamson wrote:
 I don't believe 2023 has anything to do with the 411P; it was basically
 an digium analog card issue (eg, TDM04b  x100p).

 Based on my tests and findings, the issue is the digium cards record
 voicemail messages at a very low audio level (very different from
 recording a voicemail from a sip phone). If the person leaving a VM
 message called in via the digium card, and that user was located close
 to their central office, the VM level is acceptable to poor. But, if
 that same person is further from their central office (adding additional
 transmission path loss), then that loss plus the digium analog card loss
 makes the VM difficult if not impossible to hear.

 So, thinking that statement through very carefully, you might have some
 users complain and other not, and the problem will not track against
 anything that you have control over (eg, where the remote user is calling
 from and the transmission loss they incur).

 If the digium analog cards passed audio through without any additional
 loss, your user's probably would not be complaining. But that extra loss
 is what I believe is the issue.

 Sounds like there might be a workaround coming for this.

 Rich


zapata.conf

usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=800

rxgain=5.0
txgain=5.0

group=0
callgroup=1
pickupgroup=1
immediate=no

improved it for me.  YMMV.

Regards...Martin
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RE: [Asterisk-Users] TDM card and voicemail volume

2005-09-07 Thread tmassey

[EMAIL PROTECTED] wrote on 06/28/2005 07:52:50 AM:

 I was able to raise the volume from inaudible to acceptable by
 increasing the RxGain in zapata.conf by 5db.  I'd rather not go the
 uncomressed wav route, as it will chew up storage in my email system. 

I know I'm way behind on reading this, but I thought I would follow up.

According to this message:

 
http://lists.digium.com/pipermail/asterisk-users/2004-November/072990.html

the reason that uncompressed WAV files are louder is that the software 
that saves the WAV file is amplifying the volume of the files by shifting 
the data two bits to the left (or making it 4x louder).  It is in no way 
fixing the underlying problem of the file being too quiet;  it is just 
throwing away dynamic range in order to amplify the file.

Now that may not be a bad solution:  if you don't need the dynamic range, 
but you *do* need the volume, so be it:  you would prefer the off-chance 
of some clipping.  It *has* to be a better solution to using the rxgain 
setting if you don't need to:  rxgain is going to affect echo for the 
worse.  Also notice that the volume of these files is sufficient when they 
are played back over the telephone:  it's only when you play them back via 
a sound card that you have the volume problem.  So, you can't just 
willy-nilly amplify everything.

Hope this helps.

Tim Massey

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Re: [Asterisk-Users] Speaking of Polycom phones...updated ROM: ouch!

2005-09-07 Thread Doug


At 16:16 9/6/2005 -0700, Jesse Keating wrote:

On Tue, 2005-09-06 at 17:41 -0500, Doug wrote:
 After I did this, it appears that the Web interface
 for the phone won't change the settings, nor will
 it actually reboot the phone now.  What do I have
 to set on the phone itself, so I can update info
 in the Web interface, and then restart the phone?


What you need to do is 'clear local config' before you start making
changes.

Menu - Settings - Advanced ( - password ) - Admin Settings - Reset
to Default - Reset Local Config

Once you've done that and rebooted, you should be able to make your
changes through web or on the phone itself.



Hey Jesse,

Thanks for the advice.  It worked like charm.  Now
I can set fields in the Web interface, and reboot.

However, now I can't seem to the phone to register.
Have you seen this before?

I again followed instructions here:
http://www.voip-info.org/tiki-index.php?page=Polycom+SoundPoint+IP+501

Would you be willing to save and zip up your config
pages for the phone and Asterisk for me to compare
with mine?

http://192.168.2.5/netConf.htm
http://192.168.2.5/appConf.htm
http://192.168.2.5/reg.htm
http://xxx.xxx.xx.xx/admin/config.php?display=3extdisplay=10100

In the boot log, it shows something like:

x....cfg could not be downloaded.  Getting next file...

That file does not exist on the TFTP server.  Is this a
problem?

Also, in the boot log it shows DNS referring to 192.168.2.1,
even though it's setup properly at the phone and its
corresponding Web interface.

Any other ideas?

Thanks!



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[Asterisk-Users] sip - aastra 9133i

Hello.

Just rx'd the sip - aastra 9133i.

Haven't done sip before.

My initial attempt at setup has failed.

No Service

Anyone want to contact me off-list or on irc ?

Regards...Martin
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RE: [Asterisk-Users] TDM card and voicemail volume