Re: [asterisk-users] Zaptel-1.4.1 error cross compile

[Alan Lord]

Satish Patel wrote:
snip /
 I am using cross compile so i can't update GCC other wise it will effect on
 my other packages anyway... tell me one thing i have host system kernel
 version is 2.6.18 and i am compiling ARM embedded rootbuild with other
 kernel version 2.6.22 so i need to compile my zaptel package with 2.6.22
 kernel caz i will use it on target ARM hardware ( IXP425 ). I am doing that
 but after porting rootfs on target host and when i run insmod zaptel
 command on target board i got error
snip /
 clfs:/mnt/clfs$ file lib/modules/2.6.22.6/misc/zaptel.ko
 lib/modules/2.6.22.6/misc/zaptel.ko: ELF 32-bit MSB relocatable, ARM,  
 version 1, not stripped
 
 i dont know why this error coming i think its caz confusion between  
 host kernel and target kernel

I think you may be right.

Can you not extract a set of kernel headers for 2.6.18 and point the 
zaptel build to them when you are making it in your cross-compile 
environment? I can't remember the switch off hand but I am sure there is 
a way to point the make scripts at whatever headers you wish.

HTH

Al


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[asterisk-users] Fwd: Fonolo: Visually Navigate Dial IVR Phone Menus in Web/Mobile Browser

[Babcock, Michael Alex]

this might be classifyed to some of you as ot but i wonder what this  
will do for the asterisk community?


Begin forwarded message:


Date: October 5, 2008 5:17:36 PM GMT-08:00
Subject: Fonolo: Visually Navigate  Dial IVR Phone Menus in Web/ 
Mobile Browser

Source: Tech[dot]Blog
Author: Abdul Aziz

It is a fact that everyone hates to listen to automated phone menus-  
Interactive Voice Response (IVR)- and go through endless options to  
reach a human being. A service called Fonolo has tried to make this  
experience easier by listing the entire phone menu tree visually on  
one page and provides call buttons to skip right to that part of the  
menu. The best part is that it actually calls you when it’s time to  
talk to someone and you don’t even have to do any dialing.


Fonolo transcribes the phone menus of large companies to navigate  
them visually. Pick the company you need, scan through their phone  
menu visually, then just click the spot you need to call. Fonolo  
will automatically dial, navigate their menu and then dial your  
phone. When you answer, you will be connected to the right spot in  
the menu. You can even bookmark any point in a phone menu and access  
that bookmark as a simple URL through your browser or smartphone.




Fonolo also provides an “Intelligent Call History” that allows you  
to keep track of your calls, notes and recordings. It automatically  
organizes all of your calls to a given company, regardless of which  
phone you used or which number was dialed. It stores recordings of  
all the calls that you can review at any time or forward to someone  
by email. It also allows you to write text notes during a call that  
get stored with the history. You can later search and review those  
notes.


Fonolo’s revolutionary technology “spiders” the phone menu system,  
much like a search engine spider crawls a website. Their system  
dials companies, navigates their menus and uses a combination of  
speech recognition, signal processing and human editing to maintain  
the IVR visually.






Read more…



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

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[asterisk-users] Alarm events + asterisk dies

[Roberts Klotins]

Hi All,

I am getting these events in asterisk message log:
NOTICE[16647] chan_zap.c: Got event 4 (Alarm)...
NOTICE[16647] chan_zap.c: Alarm cleared on channel 1

after that asterisk exits silently until I restart it. Sometimes zapata
drivers also get in a state where I need to physically restart the
machine. Does anyone have any suggestions how to troubleshoot these
alarm events?

Roberts


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Re: [asterisk-users] R key with Siemens Gigaset IP (was MWI with Siemens Gigaset S450IP)

[Olivier]

2008/10/5 [EMAIL PROTECTED] [EMAIL PROTECTED]

 Kevin P. Fleming wrote:
  Olivier wrote:
 
 
  2. R Hook-flash key is now available to transfer calls.
  In s450IP web management server, its defaults settings are :
  Application-type: dtmf-relay
  Application-signal: 16
 
  Is there anything to configure in features.conf, extensionsconf or
  elsewhere to trigger transfers when R key is pressed ?
 
 
  I don't believe there is any support for hook-flash style transfers over
  SIP in Asterisk; that key should be programmed to use standard SIP
  transfer methods, not DTMF emulation methods.
 
 
 do you have a suggestion, there is only two fields that can be filled in
 that to refer to the R key,

 Application-type:  I think this is content type
 Application-signal: what it sends?


Hello,

Reading this thread, I think I should have opened in the first place, 2
different threads as a common title is misleading to this R Hook-Flash key
topic.

Now, Gigaset S450IP base configuration web offers 2 fields to set R key :
Application-type:
Application-signal:

When those 2 fields are respectively valued to
Application-type:  dtmf-relay
Application-signal:  16

... anytime this R-key is pressed, the base station would send a SIP INFO
message to Asterisk.
This SIP info is ended with :
...
User-Agent: S450 IP02123000
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/dtmf-relay
Content-Length: 22

Signal=16
Duration=86

This 16 signal is interpreted as :
Receiving INFO!
* DTMF-relay event received: FLASH

In my testing, I changed values like this
Application-type:  foo
Application-signal:  16 2

and got a (single) SIP INFO message like this:
User-Agent: S450 IP02123000
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/foo
Content-Length: 22

Signal=16 2



As Kevin told previously, Hook Flash transfer is not supported by Asterisk
SIP stack.

At the same time, it is written here (
http://www.voip-info.org/wiki/view/Siemens+Gigaset+S450IP) that :

   - Enable the R-button in SIP mode *fixed 14/09/2007*


So, what does this exactly mean ?
Which values are to be typed in Application type and Application signal to
make this R key be of any use ?
Is it possible to pass several DTMF signals in a single SIP INFO so that
Asterisk would receive a *2 anytime the R-key is pressed ?


Regards
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Re: [asterisk-users] MS Exchange IMAP Voicemail

[Olivier]

2008/10/6 Andrew Joakimsen [EMAIL PROTECTED]

 Yes, IMAP is IMAP... at least it is supposed to.

 But not all IMAP servers use the same configuration. Not all IMAP
 servers will use the same Master User IMAP setup, what works in
 Dovecot might not work in UW or Exchange due to a prefix or some other
 fairly trivial setting. Remember there are two pieces of software that
 need to be configured for this to work properly.

 So I am asking if someone has a configuration that they *know works*
 with Exchange 2003 and if they could please share that.

 On Sun, Oct 5, 2008 at 9:04 PM, David Backeberg [EMAIL PROTECTED]
 wrote:
  Isn't IMAP IMAP? Does MS not actually follow the protocol? Why would
  it be different?
 
  On Sun, Oct 5, 2008 at 8:38 PM, Andrew Joakimsen [EMAIL PROTECTED]
 wrote:
  Has anyone successfully used the IMAP voicemail storage with Microsoft
  Exchange 2003? Can someone provide a working example configuration?
 
 


I have heard Exchange 2003 would allow a single Exchange account to have
read/write access to other user accounts but Exchange 2007 wouldn't allow
this anymore.

So I don't have personal experience to share but I would be delighted to
know is this specific is a concern or not when using Exchange 2003.

My 0,002 cents
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Re: [asterisk-users] Ok message

[michel freiha]

Dear Sir,

I'm sending them Session Progress as you can see in the attached log
fle...Please let me know if they ahve any reason to not sending DTMF to me

Regards




On Fri, Oct 3, 2008 at 6:54 PM, Alex Balashov [EMAIL PROTECTED]wrote:

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

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

 A packet capture would be most helpful.

 michel freiha wrote:

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

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--- SIP read from 81.201.82.39:5060 ---
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
From: anonymous sip:[EMAIL PROTECTED];tag=70665
To: sip:[EMAIL PROTECTED]
Via: SIP/2.0/UDP 
81.201.82.39:5060;branch=z9hG4bKd865aecb62fe5bc6e374f67b61eabce7
Max-Forwards: 69
Content-Type: application/sdp
Contact: sip:[EMAIL PROTECTED]:5060;transport=udp
User-Agent: Vox Callcontrol
Content-Length: 311

v=0
o=root 16790 16790 IN IP4 81.201.82.23
s=session
c=IN IP4 81.201.82.23
t=0 0
m=audio 11564 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

-
--- (11 headers 15 lines) ---
Sending to 81.201.82.39 : 5060 (no NAT)
Using INVITE request as basis request - [EMAIL PROTECTED]
Found peer 'sip_proxy1'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 81.201.82.23:11564
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0x4 (ulaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 
(nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 81.201.82.23:11564
Looking for 155469877445 in stations (domain Asterisk_IP)
list_route: hop: sip:[EMAIL PROTECTED]:5060;transport=udp
localhost*CLI 
--- Transmitting (no NAT) to 81.201.82.39:5060 ---
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 
81.201.82.39:5060;branch=z9hG4bKd865aecb62fe5bc6e374f67b61eabce7;received=81.201.82.39
From: anonymous sip:[EMAIL PROTECTED];tag=70665
To: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:[EMAIL PROTECTED]
Content-Length: 0




--- Transmitting (no NAT) to 81.201.82.39:5060 ---
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 
81.201.82.39:5060;branch=z9hG4bKd865aecb62fe5bc6e374f67b61eabce7;received=81.201.82.39
From: anonymous sip:[EMAIL PROTECTED];tag=70665
To: sip:[EMAIL PROTECTED];tag=as78e4c405
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:[EMAIL PROTECTED]
Content-Type: application/sdp
Content-Length: 238

v=0
o=root 637 637 IN IP4 Asterisk_IP
s=session
c=IN IP4 Asterisk_IP
t=0 0
m=audio 17750 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv


localhost*CLI 
--- SIP read from 81.201.82.39:5060 ---
CANCEL sip:[EMAIL PROTECTED] SIP/2.0
Call-ID: [EMAIL PROTECTED]
CSeq: 102 CANCEL
From: anonymous sip:[EMAIL PROTECTED];tag=70665
To: sip:[EMAIL PROTECTED]
Via: SIP/2.0/UDP 

Re: [asterisk-users] queue members randomly become paused after upgrade to Asterisk 1.4

[Giorgio Incantalupo]

Hi Mark,
made some other tests but the problem remains. I installed 1.4.22-rc5 
but nothing changed. I opened an issue on mantis waiting for a fix.

Giorgio


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Re: [asterisk-users] asterisk, phpagi and singleton

[Giedrius Augys]

2008/10/6 Alex Balashov [EMAIL PROTECTED]

 I think the problem is that every [Dead]AGI call is still a distinct
 invocation of the script, even if the interpreter stays loaded as an ELF
 module or whatnot.

 A good solution to this problem would be to use a FastAGI service,
 wherein a daemon runs persistently with a reusable DB handle.  Calls to
 AGI can connect to that using a service mode of operation rather than
 invoking a local script.

 Giedrius Augys wrote:

  Hello,
 
 I've this situation: 300+ simultaneous calls and dialplan like this:
  exten = _X.,1,Answer()
  exten = _X.,2,DEADAGI(check_status.php)
  exten = _X.,3,Dial(SIP/other/${NUMBER})
  exten = _X.,4,Hangup
 
  exten = h,1,DEADAGI(cdr.php)
 
  When project is running , I had  a lot of defunct php scripts (I've
  exceed mysql connection limits and so on, deadagi help a bit). The
  scripts check_status.php and cdr.php connects to database to
  retrieve/store data. So one call - 2 connections to database. So I want
  to do like this: 100 simultaneous calls , make 200 queries per one mysql
  connection. WEB developers uses singleton to avoid this issue. Maybe
  somebody has experience with singleton and phpagi.
  thanks...
 
  --
  Pagarbiai  / Best Regards,
  Giedrius Augys
 
 
  
 
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 --
 Alex Balashov
 Evariste Systems
 Web: http://www.evaristesys.com/
 Tel: (+1) (678) 954-0670
 Direct : (+1) (678) 954-0671
 Mobile : (+1) (706) 338-8599

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What tools and programming (scripting) language do you use for FastAGI?
Thanks
-- 
Pagarbiai  / Best Regards,
Giedrius Augys
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Re: [asterisk-users] Tribox

[Steven Howes]

Hi

triXbox.org can answer these questions. Google may also give a  
balanced view. But yes, i can assure you, people are using Trixbox  
from Fonality.

Steve

On 6 Oct 2008, at 10:24, broadband Voice wrote:

 Anyone using Tribox from Fonality. I understand its open source and  
 free. Can I use it for a call center functionality? Thanks.
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[asterisk-users] Tribox

[broadband Voice]

Anyone using Tribox from Fonality. I understand its open source and free.
Can I use it for a call center functionality? Thanks.
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Re: [asterisk-users] asterisk, phpagi and singleton

[Alex Balashov]

Giedrius Augys wrote:

 What tools and programming (scripting) language do you use for FastAGI?

Whatever languages FastAGI APIs are available for.  You are pretty much 
limited to languages whose interpreter lends itself to invocation as a 
standalone daemon, which may or may not exclude PHP and other languages 
designed to be web scripting languages and whose state is expected to be 
determined in terms of serial HTTP requests.

I use Perl, personally:

http://search.cpan.org/~jaywhy/Asterisk-FastAGI-0.02/lib/Asterisk/FastAGI.pm

-- Alex

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

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Re: [asterisk-users] asteriskt38.com

[Daniel Ferenci]

Hi

...
and there is a new application called fax gateway (
http://bugs.digium.com/view.php?id=13405)
that can do gatewaying between T30 and T38 and vice versa.

Best regards
Daniel.


 asterisk.
 

 Actually it exists. 1.4 had passtrough mode and 1.6 can send and receive.

 Regards,
 Atis

 --
 Atis Lezdins,
 VoIP Project Manager / Developer,
 [EMAIL PROTECTED]
 Skype: atis.lezdins
 Cell Phone: +371 28806004
 Cell Phone: +1 800 7300689
 Work phone: +1 800 7502835

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Re: [asterisk-users] asteriskt38.com

[Atis Lezdins]

On Mon, Oct 6, 2008 at 7:39 AM, Andrew Joakimsen [EMAIL PROTECTED] wrote:
 I was going to write a blog once about the non-existent T.38 support
 in asterisk hence my purchase of the above domain. It expires in 10
 days. T.38 support in asterisk still does not exist but I don't have
 any time. If someone wants this domain I will offer it for free and
 can send push it to your enom account since I was going to allow it to
 expire anyways. The only condition would be that you do not use it for
 a commercial use, i.e. you don't try to sell a t.38 module for
 asterisk.


Actually it exists. 1.4 had passtrough mode and 1.6 can send and receive.

Regards,
Atis

-- 
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835

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

[Olivier]

2008/10/3 Olivier [EMAIL PROTECTED]

 Hi,

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

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

 Has anyone been any further ?
 Regards


Replying to myself, for an unknown reason, MWI is weirdly working  :
- Phone icon inconsistently shows awaiting voicemails,
- NOTIFY message from Asterisk are still replied with 481 Call
Leg/Transaction Does Not Exist

When base station is restarted, it will SUBSCRIBE its endpoints to Voicemail
Notifications :
- you can see SUBSCRIBE message
- you can see NOTIFY answer
- you can't see any 481 Call Leg/Transaction Does Not Exist reply to this
NOTIFY message

From then on, further NOTIFY messages are replied with 481 Call
Leg/Transaction Does Not Exist and obviously not taken into account as
endpoint GUI remains unchanged.

Looking deeper into this here are :

NOTIFY message accepted by S450IP

NOTIFY sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.254:5060;branch=z9hG4bK06adc48b;rport
From: asterisk sip:[EMAIL PROTECTED];tag=as4ea953db
To: sip:sip:[EMAIL PROTECTED]:5060;tag=2580238520
Contact: sip:[EMAIL PROTECTED] [EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Subscription-State: active
Content-Length: 89

Messages-Waiting: yes
Message-Account: sip:[EMAIL PROTECTED]
Voice-Message: 2/0 (0/0)



NOTIFY message rejected by S450IP (rejected means 481 reply)

NOTIFY sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.254:5060;branch=z9hG4bK3d83e7f6;rport
From: asterisk sip:[EMAIL PROTECTED][EMAIL PROTECTED]
;tag=as5e574490
To: sip:[EMAIL PROTECTED]:5060
Contact: sip:[EMAIL PROTECTED] [EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 96

Messages-Waiting: yes
Message-Account: sip:[EMAIL PROTECTED][EMAIL PROTECTED]
Voice-Message: 3/0 (0/0)



The only difference I see between both is that new NOTIFY don't include :
Subscription-State: active

Do you see something else ?
Is it possible to easily add this Subscription-State field without patching
Asterisk source (I'm unable to do that) ?
Your thoughts ?

Regards
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Re: [asterisk-users] Zaptel-1.4.1 error cross compile

[satish patel]

 

Satish Patel wrote:
snip /

I think you may be right.

Can you not extract a set of kernel headers for 2.6.18 and point the zaptel
build to them when you are making it in your cross-compile environment? I
can't remember the switch off hand but I am sure there is a way to point
the make scripts at whatever headers you wish.

HTH

Al



I have set env on shell

KVERS=2.6.22.5  
KSRC=/path/to/kernel-2.6.22.5/source

This to veriable give you that option to tell zaptel use it and compile with
that specified kernel I have do it at zaptel compile time but things not
work..error which I had mentioned on last mail. Astfin using same thing and
they people using zaptel on embedded system I don't know how ??


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Re: [asterisk-users] Tribox

[Tarek Sawah]

i am on Trixbox and it works as you want it to work.. if you need further help 
i can offer some.. regards
 
AHD Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP
Syria: +963 944 618286
USA: +1 347 562 2308

Date: Mon, 6 Oct 2008 05:24:14 -0400From: [EMAIL PROTECTED]: [EMAIL PROTECTED]: 
[asterisk-users] Tribox
Anyone using Tribox from Fonality. I understand its open source and free. Can I 
use it for a call center functionality? Thanks. 
_
See how Windows connects the people, information, and fun that are part of your 
life.
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Re: [asterisk-users] Zaptel-1.4.1 error cross compile

[Alan Lord]

satish patel wrote:
snip /
 I have set env on shell
 
 KVERS=2.6.22.5  
 KSRC=/path/to/kernel-2.6.22.5/source
 

Maybe I misunderstood you then. I thought you said that your ARM system 
was using a 2.6.18 kernel? If that is the case, then surely you need to 
build your zaptel module against that and *not* the kernel on your 
cross-compiling host machine.

But probably I misunderstood what you meant.

Al



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[asterisk-users] Conneting Asterisk to Swyx pri

[Geraint Lee]

Hi all, I've done this a few times with other PBX's but swyx has stumped me!
I'm having some trouble getting Asterisk connected to a Swyx system using a
sangoma A104dx... currently the setup is:
BT - Swyx

The above setup works fine... what i'm trying to achieve is
BT  SIP Trunks - Asterisk - Swyx

I have connected to our BT (2 x ISDN30 UK) with asterisk and have no errors
and can make and receive calls and it never dies... the problem comes when i
try and connect asterisk to swyx...
I can make calls from asterisk to the swyx system with no problems or
errors, but... when i try and place a call from Swyx to asterisk i receive
the following error:
[Oct  6 10:56:20] ERROR[9794]: chan_zap.c:8250 zt_pri_error: !! Unexpected
Channel selection 3

The call does complete as normal but after about 2 or 3 hours of calls
passing through this setup i start receiving errors like the following:
[Oct  6 10:55:25] WARNING[9794]: chan_zap.c:8074 pri_fixup_principle: Can't
fix up channel from 63 to 92 because 92 is already in use
[Oct  6 10:55:25] WARNING[9794]: chan_zap.c:9245 pri_dchannel: Hangup on bad
channel 0/30 on span 3
[Oct  6 10:55:55] WARNING[9794]: chan_zap.c:8074 pri_fixup_principle: Can't
fix up channel from 63 to 92 because 92 is already in use

And eventually no more calls can be placed from swyx to asterisk... time for
some configs... and before anyone says something about wanpipe3 and 4 having
dchan=0, i tried with dchan=16 and no calls can be placed...

I hope someone can point me in the right direction as we're trying to get
rid of swyx since we're tied down by limiting software and excessive
licensing costs.

Thanks!

Geraint

pri show spans shows all spans as up and active.
zap show status shows all as ok
wanrouter status shows all as connected

wanpipe1 and 2:
[devices]
wanpipe1 = WAN_AFT_TE1, Comment

[interfaces]
w1g1 = wanpipe1, , TDM_VOICE, Comment

[wanpipe1]
CARD_TYPE   = AFT
S514CPU = A
CommPort= PRI
AUTO_PCISLOT= NO
PCISLOT = 1
PCIBUS  = 16
FE_MEDIA= E1
FE_LCODE= HDB3
FE_FRAME= CRC4
FE_LINE = 1
TE_CLOCK= NORMAL
TE_REF_CLOCK= 0
TE_SIG_MODE = CCS
TE_HIGHIMPEDANCE= NO
LBO = 120OH
FE_TXTRISTATE   = NO
MTU = 1500
UDPPORT = 9000
TTL = 255
IGNORE_FRONT_END = NO
TDMV_SPAN   = 1
TDMV_DCHAN  = 16
TDMV_HW_DTMF= NO

[w1g1]
ACTIVE_CH   = ALL
TDMV_ECHO_OFF   = NO
TDMV_HWEC   = YES


wanpipe3 and 4:
[devices]
wanpipe3 = WAN_AFT_TE1, Comment

[interfaces]
w3g1 = wanpipe3, , TDM_VOICE, Comment

[wanpipe3]
CARD_TYPE   = AFT
S514CPU = A
CommPort= PRI
AUTO_PCISLOT= NO
PCISLOT = 1
PCIBUS  = 16
FE_MEDIA= E1
FE_LCODE= HDB3
FE_FRAME= CRC4
FE_LINE = 3
TE_CLOCK= MASTER
TE_REF_CLOCK= 1
TE_SIG_MODE = CCS
TE_HIGHIMPEDANCE= NO
LBO = 120OH
FE_TXTRISTATE   = NO
MTU = 1500
UDPPORT = 9000
TTL = 255
IGNORE_FRONT_END = NO
TDMV_SPAN   = 3
TDMV_DCHAN  = 0
TDMV_HW_DTMF= NO

[w3g1]
ACTIVE_CH   = ALL
TDMV_ECHO_OFF   = NO
TDMV_HWEC   = YES

zaptel.conf:
loadzone=uk
defaultzone=uk

#Sangoma A104 port 1 [slot:1 bus:16 span:1] wanpipe1
span=1,0,0,ccs,hdb3,crc4
bchan=1-15,17-31
hardhdlc=16

#Sangoma A104 port 2 [slot:1 bus:16 span:2] wanpipe2
span=2,0,0,ccs,hdb3,crc4
bchan=32-46,48-62
hardhdlc=47

#Sangoma A104 port 3 [slot:1 bus:16 span:3] wanpipe3
span=3,0,0,ccs,hdb3,crc4
bchan=63-77,79-93
dchan=78

#Sangoma A104 port 4 [slot:1 bus:16 span:4] wanpipe4
span=4,0,0,ccs,hdb3,crc4
bchan=94-108,110-124
dchan=109

I have also tried with hardhdlc=109 and have the same problem.

zapata.conf:
[channels]
language=en
usecallerid=yes
hidecallerid=no
callwaiting=no
callwaitingcallerid=yes
restrictcid=no
usecallingpres=no
threewaycalling=yes
callreturn=yes
transfer=yes
cancallforward=yes
musiconhold=default
rxgain=0.0
txgain=0.0
immediate=no

; BT
switchtype=euroisdn
group=1
context=from-bt
signalling=pri_cpe

; Port 1 - BT
channel = 1-15,17-31

; Port 2 - BT
channel = 32-46,48-62

; Swyx
overlapdial=yes
group=2
context=from-swyx
signalling=pri_net

; Port 3 - Swyx
channel = 63-77,79-93

; Port 4 - Swyx
channel = 94-108,110-124
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[asterisk-users] How to implement Ringing through a sound card for overhead paging

[Robert Augustyn]

Hi,
I have followed this guide
http://www.voip-info.org/wiki/view/Setting+up+paging+with+a+sound+card
 and have paging working ok, now I need to implement 'ringing'.
When someone calls I need the ringing to be send to overhead paging through
the sound card.
Any pointers?
 
 
Sincerely,
Robert Augustyn
www.linqone.com http://www.linqone.com/ 
 
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Re: [asterisk-users] OT: headsets

[Jay R. Ashworth]

- Bill Michaelson [EMAIL PROTECTED] wrote:
 The IP330 has a subminiature jack for headset/mic combos.  Are there 
 quality headsets anyone would recommend for in-office use for heavy 
 users with these phones?  Using any wiring path?  I've tried a cell 
 phone earphone/mic, and it sounds OK, but it's flimsy for this
 application.

In our call center (250 agents), we use Panasonic KX-TCA60 headsets, which
I think we get for 7 or 8 bucks apiece, and sell to the agents at cost.

They have covered gooseneck tubes, decent padding on the earpiece, and are
fairly sturdy.  Turnover being what it is, we don't have to replace too 
many of them for breakage.

They have 2.5mm plugs, and really good audio -- I've plugged mine into my
Nextel/RIM BlackBerry 7100i, and called my best friend, who is almost as
picky as I am... his opinion is that it not only sounds better than my 
Plantronics Voyager 510, it sounds better than the mic inside the phone.

My opinion is the converse: receive audio is nice too.

Recommended.

Cheers,
-- jra
-- 
Jay R. Ashworth   Baylink  [EMAIL PROTECTED]
Designer The Things I Think   RFC 2100
Ashworth  Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA  http://photo.imageinc.us +1 727 647 1274

 Those who cast the vote decide nothing.
 Those who count the vote decide everything.
   -- (Josef Stalin)


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[asterisk-users] AEL and swap from macros to contexts

[Pavel Jezek]

Hi, according to discussion on asterisk IRC, where people said, that 
macros will be depracated, I tried to migrate from macros to contexts 
and Gosub
but if I try to use gosub in extensions.ael, ael compiler complains, 
that I shouln't use Gosub app,
but I can't find ael keyword, that will be Gosub equivalent, or can I 
ignore this ael warnings? thanks
PJ


LOG: lev:3 file:pval.c  line:2521 func: check_pval_item  Warning: file 
/etc/asterisk/extensions.ael, line 36-36: application call to Gosub 
affects flow of control, and needs to be re-written using AEL if, while, 
goto, etc. keywords instead!


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Re: [asterisk-users] No route to destination error

[Martin Seebach]

- Andres wrote: 
 After looking at your iax.conf and extensions.conf I believe you are 
 under the misconception that if you 'register' to a provider, then you 
 can send and receive calls. The fact is that you 'register' to receive 
 calls, but you must define a trunk in order to Dial Out. Your iax.conf 
 [88821268] entry is not a trunk as you have not defined a host. That is 
 why you get cause 3 - No route to destination. Asterisk does not have 
 any host defined in order to route that call. You need to talk to your 
 provider for instructions on how to setup the trunk. 

That was indeed the problem. I added this to iax.conf: 

[myprovider] 
type=friend 
username=88821268 
secret=xxzzyy 
host=s1.core.myprovid.er 

And used this in extensions.conf: 
exten = _ZXXX,2,Dial(IAX2/myprovider/${EXTEN:0},30,r) 

Thank you for the assistance. 

Regards, 
Martin Seebach 


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Re: [asterisk-users] Conneting Asterisk to Swyx pri

[Gordon Henderson]

On Mon, 6 Oct 2008, Geraint Lee wrote:

 Hi all, I've done this a few times with other PBX's but swyx has stumped me!
 I'm having some trouble getting Asterisk connected to a Swyx system using a
 sangoma A104dx... currently the setup is:
 BT - Swyx

 The above setup works fine... what i'm trying to achieve is
 BT  SIP Trunks - Asterisk - Swyx

 I have connected to our BT (2 x ISDN30 UK) with asterisk and have no errors
 and can make and receive calls and it never dies... the problem comes when i
 try and connect asterisk to swyx...
 I can make calls from asterisk to the swyx system with no problems or
 errors, but... when i try and place a call from Swyx to asterisk i receive
 the following error:
 [Oct  6 10:56:20] ERROR[9794]: chan_zap.c:8250 zt_pri_error: !! Unexpected
 Channel selection 3

 The call does complete as normal but after about 2 or 3 hours of calls
 passing through this setup i start receiving errors like the following:
 [Oct  6 10:55:25] WARNING[9794]: chan_zap.c:8074 pri_fixup_principle: Can't
 fix up channel from 63 to 92 because 92 is already in use
 [Oct  6 10:55:25] WARNING[9794]: chan_zap.c:9245 pri_dchannel: Hangup on bad
 channel 0/30 on span 3
 [Oct  6 10:55:55] WARNING[9794]: chan_zap.c:8074 pri_fixup_principle: Can't
 fix up channel from 63 to 92 because 92 is already in use

 And eventually no more calls can be placed from swyx to asterisk... time for
 some configs... and before anyone says something about wanpipe3 and 4 having
 dchan=0, i tried with dchan=16 and no calls can be placed...

 I hope someone can point me in the right direction as we're trying to get
 rid of swyx since we're tied down by limiting software and excessive
 licensing costs.

So go in one Saturday morning, wire it up as you want (BT - Asterisk) and 
the re-configure all the SIP phones to talk directly to the asterisk box 
and not the swyx box, then arrange the the swyx box to misteriously die, 
then tell everyone what a good job it was that you were in on the weekend 
to re-configure the phones to use the asterisk box ;-)

Gordon

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Re: [asterisk-users] How to implement Ringing through a sound card for overhead paging

[Doug Lytle]

Robert Augustyn wrote:
 Hi,
 I have followed this guide
 http://www.voip-info.org/wiki/view/Setting+up+paging+with+a+sound+card
  and have paging working ok, now I need to implement 'ringing'.
 When someone calls I need the ringing to be send to overhead paging 
 through the sound card.

I have recorded a sound effect, use a callfile to play the file via the 
sound card.  I have a very short timeout for that extension.  I just 
jump back to the beginning on the context, play the sound effect and 
then ring the phone again.  Code below:


;**
;* If Press extension is dialed after 5pm, play bull
;* Horn sound effect to get pressman's attention
;**

[night_bell]

exten = 4173,1,GotoIfTime(07:45-16:59|mon-fri|*|*?press-officehours,s,1)
exten = 4173,2,System(/bin/cp /usr/local/bin/bullhorn.call 
/var/spool/asterisk/outgoing/bullhorn`date +%s`.call)
exten = 4173,3,Dial(SIP/4173,15,tTkK)
exten = 4173,4,Goto(night_bell,4173,1)


bullhorn.call

Channel: Console/dsp
MaxRetries: 0
Application: playback
Data: /var/lib/asterisk/sounds/local/bullhorn


Doug

-- 
 
Ben Franklin quote:

Those who would give up Essential Liberty to purchase a little Temporary 
Safety, deserve neither Liberty nor Safety.


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Re: [asterisk-users] Conneting Asterisk to Swyx pri

[Geraint Lee]

brilliant idea - except it would be a sunday morning and another problem
the handsets that come with swyx aren't sip compatible :S

Cheers

Geraint

2008/10/6 Gordon Henderson
[EMAIL PROTECTED][EMAIL PROTECTED]


 On Mon, 6 Oct 2008, Geraint Lee wrote:

  Hi all, I've done this a few times with other PBX's but swyx has stumped
 me!
  I'm having some trouble getting Asterisk connected to a Swyx system using
 a
  sangoma A104dx... currently the setup is:
  BT - Swyx
 
  The above setup works fine... what i'm trying to achieve is
  BT  SIP Trunks - Asterisk - Swyx
 
  I have connected to our BT (2 x ISDN30 UK) with asterisk and have no
 errors
  and can make and receive calls and it never dies... the problem comes
 when i
  try and connect asterisk to swyx...
  I can make calls from asterisk to the swyx system with no problems or
  errors, but... when i try and place a call from Swyx to asterisk i
 receive
  the following error:
  [Oct  6 10:56:20] ERROR[9794]: chan_zap.c:8250 zt_pri_error: !!
 Unexpected
  Channel selection 3
 
  The call does complete as normal but after about 2 or 3 hours of calls
  passing through this setup i start receiving errors like the following:
  [Oct  6 10:55:25] WARNING[9794]: chan_zap.c:8074 pri_fixup_principle:
 Can't
  fix up channel from 63 to 92 because 92 is already in use
  [Oct  6 10:55:25] WARNING[9794]: chan_zap.c:9245 pri_dchannel: Hangup on
 bad
  channel 0/30 on span 3
  [Oct  6 10:55:55] WARNING[9794]: chan_zap.c:8074 pri_fixup_principle:
 Can't
  fix up channel from 63 to 92 because 92 is already in use
 
  And eventually no more calls can be placed from swyx to asterisk... time
 for
  some configs... and before anyone says something about wanpipe3 and 4
 having
  dchan=0, i tried with dchan=16 and no calls can be placed...
 
  I hope someone can point me in the right direction as we're trying to get
  rid of swyx since we're tied down by limiting software and excessive
  licensing costs.

 So go in one Saturday morning, wire it up as you want (BT - Asterisk) and
 the re-configure all the SIP phones to talk directly to the asterisk box
 and not the swyx box, then arrange the the swyx box to misteriously die,
 then tell everyone what a good job it was that you were in on the weekend
 to re-configure the phones to use the asterisk box ;-)

 Gordon

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[asterisk-users] Hook Flash

[Lucas Alvarez]

Hi, I'm having a problem conecting my asterisk 1.4.21 with zaptel 1.4.11  
to a Panasonic PBX. I'm using dynamic features to send hook flash to the  
zap channels to make a call transfer to the pbx without tying a channel.  
When I call from asterisk to the Panasonic PBX I haven't any no problem,  
but when the call is from the Panasonic PBX, the dynamic features doesn't  
work. I have already tried all possible combinations in feature.conf:

zapflash = *3,peer/both,flash
zapflash2 = *4,callee,flash
zapflash2 = *5,caller,flash

In all cases I am setting the variable DYNAMIC_FEATURES before the Dial().  
And is not a dtmf problem because I can see in the console the debug of  
the DTMF:


chan_zap.c:1233 zt_digit_begin: Started VLDTMF digit '*'
chan_zap.c:1268 zt_digit_end: Ending VLDTMF digit '*'
chan_zap.c:1233 zt_digit_begin: Started VLDTMF digit '3'
chan_zap.c:1268 zt_digit_end: Ending VLDTMF digit '3'

The problem is that the application mapped in feature.conf it isn't been  
triggered. I would appreciate any help, I have already googled to death  
and I couldn't find anything. Thanks in advance.



Lucas Alvarez
-- 

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[asterisk-users] PoE switch recommendations?

[Ken D'Ambrosio]

Hey, all.  We're rolling out VoIP, and I'm wondering about PoE
recommendations, as we're going to have to replace our current network
equipment.  My first inclination would be to just plunk down the cash and
do a Cisco system, but I'm relatively certain that would get shot down by
finance.  Any recommendations for a couple-hundred-port solution with
VLANs, PoE, and QoS?  Don't care much if it's in a single chassis or not,
so long as it has Gbit uplinks.

Thanks!

-Ken


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Re: [asterisk-users] PoE switch recommendations?

[Geraint Lee]

Linksys SRW248P or something like that... something from linksys anyway are
quite capable of all you mentioned... maximum 24 port powered though iirc.

Geraint

2008/10/6 Ken D'Ambrosio [EMAIL PROTECTED]

 Hey, all.  We're rolling out VoIP, and I'm wondering about PoE
 recommendations, as we're going to have to replace our current network
 equipment.  My first inclination would be to just plunk down the cash and
 do a Cisco system, but I'm relatively certain that would get shot down by
 finance.  Any recommendations for a couple-hundred-port solution with
 VLANs, PoE, and QoS?  Don't care much if it's in a single chassis or not,
 so long as it has Gbit uplinks.

 Thanks!

 -Ken


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Re: [asterisk-users] PoE switch recommendations?

[Singer Wang]

We've had some bad experiences with Linksys in general (prior to going
VOIP) and avoided them. We're running now fully on the NetGear FS728TP
switch (24 port 10/100 POE, 4 port 1000 uplink, and 2 slots for fiber
modules).
 

Geraint Lee wrote:
 Linksys SRW248P or something like that... something from linksys
 anyway are quite capable of all you mentioned... maximum 24 port
 powered though iirc.

 Geraint

 2008/10/6 Ken D'Ambrosio [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]

 Hey, all.  We're rolling out VoIP, and I'm wondering about PoE
 recommendations, as we're going to have to replace our current network
 equipment.  My first inclination would be to just plunk down the
 cash and
 do a Cisco system, but I'm relatively certain that would get shot
 down by
 finance.  Any recommendations for a couple-hundred-port solution with
 VLANs, PoE, and QoS?  Don't care much if it's in a single chassis
 or not,
 so long as it has Gbit uplinks.

 Thanks!

 -Ken


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Re: [asterisk-users] PoE switch recommendations?

[Chris Bagnall]

We've used Linksys SRW224P units at quite a few places without issue. For a 
little lower cost, we've also used Netgear FS726 series switches.

Personally, I prefer the Linksys ones - they have a serial port for 
administration rather than relying on you doing it over the LAN (though they 
have a pretty web interface, too). The pretty web interface is less fussy than 
the Netgear one (which seems unreliable in non-Internet Exploder browsers).

On the other hand, the Netgear is substantially less deep (an issue in some 
wallmount cabinets) and definitely a lot quieter.

Regards,

Chris


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[asterisk-users] Semi OT: Global Crossing

[Ken Williams]

We're looking at using Global Crossing for our WAN infrastructure that's
spread across 9 states.  We're hoping to gain some stability and one
point of contact for these sites, as our current infrastructure is
pathetic for VoIP.

 

I have a couple of questions.

 

1.   Has anyone on this list used a service such as Global Crossing,
if so what have your results been?

2.   They're offering VoIP long distance minutes as a part of the
plan, has anyone used Global Crossing's VoIP servers through your
Asterisk server?

 

Thanks for any input,

Ken

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Re: [asterisk-users] PoE switch recommendations?

[Geraint Lee]

yes, thats the one i mean, 224p, the one i mentioned isn't capable of vlans
properly (which was strange, since it said it did)... i never had any
problems with them powering phones and cisco access points.

2008/10/6 Chris Bagnall [EMAIL PROTECTED]

 We've used Linksys SRW224P units at quite a few places without issue. For a
 little lower cost, we've also used Netgear FS726 series switches.

 Personally, I prefer the Linksys ones - they have a serial port for
 administration rather than relying on you doing it over the LAN (though they
 have a pretty web interface, too). The pretty web interface is less fussy
 than the Netgear one (which seems unreliable in non-Internet Exploder
 browsers).

 On the other hand, the Netgear is substantially less deep (an issue in some
 wallmount cabinets) and definitely a lot quieter.

 Regards,

 Chris


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Re: [asterisk-users] PoE switch recommendations?

[Gordon Henderson]

On Mon, 6 Oct 2008, Ken D'Ambrosio wrote:

 Hey, all.  We're rolling out VoIP, and I'm wondering about PoE
 recommendations, as we're going to have to replace our current network
 equipment.  My first inclination would be to just plunk down the cash and
 do a Cisco system, but I'm relatively certain that would get shot down by
 finance.  Any recommendations for a couple-hundred-port solution with
 VLANs, PoE, and QoS?  Don't care much if it's in a single chassis or not,
 so long as it has Gbit uplinks.

I'm curious as to why you want Gb uplinks on the switches?

If we assume 100Kb/sec per phone .. (gross rounding, using 100Kb/sec per 
phone, rather than ~80 - make the sums easier and builds in a margin) 10 
calls per Mb/sec.

So for a 24-port switch, 24 phones all talking to 24 extensions off that 
switch, the max the uplink port is going to be pushing out is 2.4Mb/sec.

For 200 extensions, say 9 x 24 port switches, with a single top-level (non 
PoE switch) switch with the PBX plugged in along side the 9 downlinks, 
that single PBX link will be carrying 2.4*9 = 22Mb/sec if all phones are 
in-use at the same time (and the PBX is carrying media)

Now you may not want to build the network like that, but it seems that Gb 
is overkill just for the VoIP side of things. (And with that many 
extensions, I would suggest keeping all the phones on one set of switches)

(Then again, it might not be possible to get big PoE switches without Gb 
uplinks, so it might be a moot point!)

So satisfy my curiosity - why Gb uplinks?

Cheers,

Gordon

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Re: [asterisk-users] Hook Flash

[Jeff Peeler]

- Lucas Alvarez [EMAIL PROTECTED] wrote:

 Hi, I'm having a problem conecting my asterisk 1.4.21 with zaptel
 1.4.11  
 to a Panasonic PBX. I'm using dynamic features to send hook flash to
 the  
 zap channels to make a call transfer to the pbx without tying a
 channel.  
 When I call from asterisk to the Panasonic PBX I haven't any no
 problem,  
 but when the call is from the Panasonic PBX, the dynamic features
 doesn't  
 work. I have already tried all possible combinations in feature.conf:
 
 zapflash = *3,peer/both,flash
 zapflash2 = *4,callee,flash
 zapflash2 = *5,caller,flash
 
 In all cases I am setting the variable DYNAMIC_FEATURES before the
 Dial().  
 And is not a dtmf problem because I can see in the console the debug
 of  
 the DTMF:
 
 
 chan_zap.c:1233 zt_digit_begin: Started VLDTMF digit '*'
 chan_zap.c:1268 zt_digit_end: Ending VLDTMF digit '*'
 chan_zap.c:1233 zt_digit_begin: Started VLDTMF digit '3'
 chan_zap.c:1268 zt_digit_end: Ending VLDTMF digit '3'
 
 The problem is that the application mapped in feature.conf it isn't
 been  
 triggered. I would appreciate any help, I have already googled to
 death  
 and I couldn't find anything. Thanks in advance.
 
 
 
 Lucas Alvarez
 -- 

Perhaps it is a matter of how fast the DTMF is being delivered from the other 
PBX. You can adjust the featuredigittimeout in features.conf to see if that is 
the case.

Jeff

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Re: [asterisk-users] PoE switch recommendations?

[David Gibbons]

We've been EXTREMELY happy with the HP 5400ZL series chassis switch. Price per 
port is about 1/3 that of Cisco when it comes to POE. Price is about $100 per 
port and all ports are 1Gb with POE by default -- you can't get modules that 
don't have 1Gb and POE. 10Gb uplinks are available with other modules.

Dave

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ken D'Ambrosio
Sent: Monday, October 06, 2008 11:03 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] PoE switch recommendations?

Hey, all.  We're rolling out VoIP, and I'm wondering about PoE
recommendations, as we're going to have to replace our current network
equipment.  My first inclination would be to just plunk down the cash and
do a Cisco system, but I'm relatively certain that would get shot down by
finance.  Any recommendations for a couple-hundred-port solution with
VLANs, PoE, and QoS?  Don't care much if it's in a single chassis or not,
so long as it has Gbit uplinks.

Thanks!

-Ken


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Re: [asterisk-users] PoE switch recommendations?

[Alexander Lopez]

Your math is correct but the application is incorrect.

The OP requested a switch with solution with VLANs, PoE, and QoS?  By that 
they would be using the VLANS and QoS for separation of Data / Voice.

Gb uplinks are very useful in Data applications..

Alex


 Kindly consider the environment before printing this e-mail.

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Gordon Henderson
 Sent: Monday, October 06, 2008 11:29 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] PoE switch recommendations?
 
 On Mon, 6 Oct 2008, Ken D'Ambrosio wrote:
 
  Hey, all.  We're rolling out VoIP, and I'm wondering about PoE
  recommendations, as we're going to have to replace our current network
  equipment.  My first inclination would be to just plunk down the cash
 and
  do a Cisco system, but I'm relatively certain that would get shot down
 by
  finance.  Any recommendations for a couple-hundred-port solution with
  VLANs, PoE, and QoS?  Don't care much if it's in a single chassis or
 not,
  so long as it has Gbit uplinks.
 
 I'm curious as to why you want Gb uplinks on the switches?
 
 If we assume 100Kb/sec per phone .. (gross rounding, using 100Kb/sec per
 phone, rather than ~80 - make the sums easier and builds in a margin) 10
 calls per Mb/sec.
 
 So for a 24-port switch, 24 phones all talking to 24 extensions off that
 switch, the max the uplink port is going to be pushing out is 2.4Mb/sec.
 
 For 200 extensions, say 9 x 24 port switches, with a single top-level (non
 PoE switch) switch with the PBX plugged in along side the 9 downlinks,
 that single PBX link will be carrying 2.4*9 = 22Mb/sec if all phones are
 in-use at the same time (and the PBX is carrying media)
 
 Now you may not want to build the network like that, but it seems that Gb
 is overkill just for the VoIP side of things. (And with that many
 extensions, I would suggest keeping all the phones on one set of switches)
 
 (Then again, it might not be possible to get big PoE switches without Gb
 uplinks, so it might be a moot point!)
 
 So satisfy my curiosity - why Gb uplinks?
 
 Cheers,
 
 Gordon
 
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Re: [asterisk-users] PoE switch recommendations?

[David Gibbons]

Obviously we don't need 1Gb connections for VOIP :)

Phones support pass through to the desktop and VLAN tagging.

The need for 1Gb ports comes from wanting to have 1Gb at the desktop.

Dave

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gordon Henderson
Sent: Monday, October 06, 2008 11:29 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PoE switch recommendations?

On Mon, 6 Oct 2008, Ken D'Ambrosio wrote:

 Hey, all.  We're rolling out VoIP, and I'm wondering about PoE
 recommendations, as we're going to have to replace our current network
 equipment.  My first inclination would be to just plunk down the cash and
 do a Cisco system, but I'm relatively certain that would get shot down by
 finance.  Any recommendations for a couple-hundred-port solution with
 VLANs, PoE, and QoS?  Don't care much if it's in a single chassis or not,
 so long as it has Gbit uplinks.

I'm curious as to why you want Gb uplinks on the switches?

If we assume 100Kb/sec per phone .. (gross rounding, using 100Kb/sec per
phone, rather than ~80 - make the sums easier and builds in a margin) 10
calls per Mb/sec.

So for a 24-port switch, 24 phones all talking to 24 extensions off that
switch, the max the uplink port is going to be pushing out is 2.4Mb/sec.

For 200 extensions, say 9 x 24 port switches, with a single top-level (non
PoE switch) switch with the PBX plugged in along side the 9 downlinks,
that single PBX link will be carrying 2.4*9 = 22Mb/sec if all phones are
in-use at the same time (and the PBX is carrying media)

Now you may not want to build the network like that, but it seems that Gb
is overkill just for the VoIP side of things. (And with that many
extensions, I would suggest keeping all the phones on one set of switches)

(Then again, it might not be possible to get big PoE switches without Gb
uplinks, so it might be a moot point!)

So satisfy my curiosity - why Gb uplinks?

Cheers,

Gordon

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Re: [asterisk-users] How to implement Ringing through a sound card for overhead paging

[Robert Augustyn]

Doug,
That is interesting concept.
How do you add this to a ring group and does it stop when an extension is
picked up?
Thank you very much.
robert  

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Doug Lytle
 Sent: Monday, October 06, 2008 10:34 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] How to implement Ringing 
 through a sound card for overhead paging
 
 Robert Augustyn wrote:
  Hi,
  I have followed this guide
  
 http://www.voip-info.org/wiki/view/Setting+up+paging+with+a+sound+card
   and have paging working ok, now I need to implement 'ringing'.
  When someone calls I need the ringing to be send to overhead paging 
  through the sound card.
 
 I have recorded a sound effect, use a callfile to play the 
 file via the sound card.  I have a very short timeout for 
 that extension.  I just jump back to the beginning on the 
 context, play the sound effect and then ring the phone again. 
  Code below:
 
 
 ;**
 ;* If Press extension is dialed after 5pm, play bull
 ;* Horn sound effect to get pressman's attention
 ;**
 
 [night_bell]
 
 exten = 
 4173,1,GotoIfTime(07:45-16:59|mon-fri|*|*?press-officehours,s,1)
 exten = 4173,2,System(/bin/cp /usr/local/bin/bullhorn.call 
 /var/spool/asterisk/outgoing/bullhorn`date +%s`.call) exten 
 = 4173,3,Dial(SIP/4173,15,tTkK) exten = 
 4173,4,Goto(night_bell,4173,1)
 
 
 bullhorn.call
 
 Channel: Console/dsp
 MaxRetries: 0
 Application: playback
 Data: /var/lib/asterisk/sounds/local/bullhorn
 
 
 Doug
 
 -- 
  
 Ben Franklin quote:
 
 Those who would give up Essential Liberty to purchase a 
 little Temporary Safety, deserve neither Liberty nor Safety.
 
 
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Re: [asterisk-users] PoE switch recommendations?

[Robert Augustyn]

Most phones support only 100M switching though  Unless you run separate
cabling for VoIP and data but then you would not need the 1G uplink. 

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 David Gibbons
 Sent: Monday, October 06, 2008 11:48 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] PoE switch recommendations?
 
 Obviously we don't need 1Gb connections for VOIP :)
 
 Phones support pass through to the desktop and VLAN tagging.
 
 The need for 1Gb ports comes from wanting to have 1Gb at the desktop.
 
 Dave
 
 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Gordon Henderson
 Sent: Monday, October 06, 2008 11:29 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] PoE switch recommendations?
 
 On Mon, 6 Oct 2008, Ken D'Ambrosio wrote:
 
  Hey, all.  We're rolling out VoIP, and I'm wondering about PoE 
  recommendations, as we're going to have to replace our 
 current network 
  equipment.  My first inclination would be to just plunk 
 down the cash 
  and do a Cisco system, but I'm relatively certain that 
 would get shot 
  down by finance.  Any recommendations for a couple-hundred-port 
  solution with VLANs, PoE, and QoS?  Don't care much if it's in a 
  single chassis or not, so long as it has Gbit uplinks.
 
 I'm curious as to why you want Gb uplinks on the switches?
 
 If we assume 100Kb/sec per phone .. (gross rounding, using 
 100Kb/sec per phone, rather than ~80 - make the sums easier 
 and builds in a margin) 10 calls per Mb/sec.
 
 So for a 24-port switch, 24 phones all talking to 24 
 extensions off that switch, the max the uplink port is going 
 to be pushing out is 2.4Mb/sec.
 
 For 200 extensions, say 9 x 24 port switches, with a single 
 top-level (non PoE switch) switch with the PBX plugged in 
 along side the 9 downlinks, that single PBX link will be 
 carrying 2.4*9 = 22Mb/sec if all phones are in-use at the 
 same time (and the PBX is carrying media)
 
 Now you may not want to build the network like that, but it 
 seems that Gb is overkill just for the VoIP side of things. 
 (And with that many extensions, I would suggest keeping all 
 the phones on one set of switches)
 
 (Then again, it might not be possible to get big PoE switches 
 without Gb uplinks, so it might be a moot point!)
 
 So satisfy my curiosity - why Gb uplinks?
 
 Cheers,
 
 Gordon
 
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Re: [asterisk-users] AEL and swap from macros to contexts

[Atis Lezdins]

On Mon, Oct 6, 2008 at 5:21 PM, Pavel Jezek [EMAIL PROTECTED] wrote:
 Hi, according to discussion on asterisk IRC, where people said, that
 macros will be depracated, I tried to migrate from macros to contexts
 and Gosub
 but if I try to use gosub in extensions.ael, ael compiler complains,
 that I shouln't use Gosub app,
 but I can't find ael keyword, that will be Gosub equivalent, or can I
 ignore this ael warnings? thanks
 PJ


 LOG: lev:3 file:pval.c  line:2521 func: check_pval_item  Warning: file
 /etc/asterisk/extensions.ael, line 36-36: application call to Gosub
 affects flow of control, and needs to be re-written using AEL if, while,
 goto, etc. keywords instead!

Hi,

In definition use:

macro set_record(A,B) {
  // do something
}

And for calling:

set_record(${CALLERID(NUM)},${EXTEN});

It will automatically be translated to GoSub in 1.6, but will remain
as Macro in 1.4.

Regards,
Atis



-- 
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835

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Re: [asterisk-users] AEL and swap from macros to contexts

[Pavel Jezek]

Atis Lezdins wrote:
 On Mon, Oct 6, 2008 at 5:21 PM, Pavel Jezek [EMAIL PROTECTED] wrote:
   
 Hi, according to discussion on asterisk IRC, where people said, that
 macros will be depracated, I tried to migrate from macros to contexts
 and Gosub
 but if I try to use gosub in extensions.ael, ael compiler complains,
 that I shouln't use Gosub app,
 but I can't find ael keyword, that will be Gosub equivalent, or can I
 ignore this ael warnings? thanks
 PJ


 LOG: lev:3 file:pval.c  line:2521 func: check_pval_item  Warning: file
 /etc/asterisk/extensions.ael, line 36-36: application call to Gosub
 affects flow of control, and needs to be re-written using AEL if, while,
 goto, etc. keywords instead!
 

 Hi,

 In definition use:

 macro set_record(A,B) {
   // do something
 }

 And for calling:

 set_record(${CALLERID(NUM)},${EXTEN});

 It will automatically be translated to GoSub in 1.6, but will remain
 as Macro in 1.4.
   

yes, I know, but I hear on IRC, that macros will be deprecated and 
suggestion was to move to contexts,
personaly I would like also move away from macros, because macros have 
some limitations, eg. variable number of arguments isn't possible with 
classic macros,
macros also require variable to be defined in macro definition (that is 
needless, because I'm referecing to ARG1, ARG2 etc. inside macros)
so I definitively agree with moving from macros to contexts, only one 
bad thing is compiler warning, when I try to Gosub to context (as macro 
replacement)
PJ



 Regards,
 Atis



   

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Re: [asterisk-users] How to implement Ringing through a sound card for overhead paging

[Doug Lytle]

Robert Augustyn wrote:
 Doug,
 That is interesting concept.
 How do you add this to a ring group and does it stop when an extension is
 picked up?
   

It depends on how you have your ring group setup, I personally only do 
this with a single extension.  And yes, the bullhorn sound stops when 
the phone is answered.

Doug


-- 
 
Ben Franklin quote:

Those who would give up Essential Liberty to purchase a little Temporary 
Safety, deserve neither Liberty nor Safety.


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Re: [asterisk-users] OT: headsets

[Bill Michaelson]

Jay R. Ashworth wrote:
In our call center (250 agents), we use Panasonic KX-TCA60 headsets, which
I think we get for 7 or 8 bucks apiece, and sell to the agents at cost.

Thanks for that - they look good, and I found several recommendations for them 
after I got yours and started looking for them.

Further to this, I'm in the client office today and dealing directly with the 
users who are reporters and editors for a periodical and conduct many telephone 
interviews.  They want to use their old recording devices with the new phones, 
but are finding unpleasant audio experiences when they switch them over from 
the Nortel meridians to the Polycom IP330s.  So I'm looking for kit to use here 
as well.  Recommendations most welcome.

And in the case of one user, she is adamant she not be required to use a 
different recording device.  I don't know how to approach this except to try a 
different telephone or mess with Polycom gain settings that the manual advises 
not to touch.  Anybody been down this road - have any wisdom?



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Re: [asterisk-users] AEL and swap from macros to contexts

[Kevin P. Fleming]

Pavel Jezek wrote:

 yes, I know, but I hear on IRC, that macros will be deprecated and 
 suggestion was to move to contexts,
 personaly I would like also move away from macros, because macros have 
 some limitations, eg. variable number of arguments isn't possible with 
 classic macros,
 macros also require variable to be defined in macro definition (that is 
 needless, because I'm referecing to ARG1, ARG2 etc. inside macros)
 so I definitively agree with moving from macros to contexts, only one 
 bad thing is compiler warning, when I try to Gosub to context (as macro 
 replacement)

You are confusing AEL macros with traditional dialplan macros; they are
no longer the same thing. As of Asterisk 1.6, AEL macros are implemented
using Gosub, but this is transparent to the AEL programmer... the AEL
dialplan still calls it a 'macro'.

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

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Re: [asterisk-users] asterisk, phpagi and singleton

[Steve Edwards]

On Mon, 6 Oct 2008, Alex Balashov wrote:

 Giedrius Augys wrote:

 What tools and programming (scripting) language do you use for FastAGI?

 Whatever languages FastAGI APIs are available for.  You are pretty much
 limited to languages whose interpreter lends itself to invocation as a
 standalone daemon, which may or may not exclude PHP and other languages
 designed to be web scripting languages and whose state is expected to be
 determined in terms of serial HTTP requests.

 I use Perl, personally:

While not an interpreted scripting language, I would use C :)

Thanks in advance,

Steve Edwards  [EMAIL PROTECTED]  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000

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Re: [asterisk-users] OT: headsets

[Michael Graves]

Two options worth considering:

1. Use a soft phone that supports call recording. The convenience of
recording directly to the PC might win some converts. X-Lite and
Ebeybeam do this nicely, amongst others.

2. Using the Polycom IP650 which has onboard call recording to a USB
device when the optional software productivity suite is installed. It's
an extra $12/phone. The phones are great! But more costly. Street
prices running around $260 each.

Michael

On Mon, 06 Oct 2008 12:35:18 -0400, Bill Michaelson wrote:

Jay R. Ashworth wrote:
In our call center (250 agents), we use Panasonic KX-TCA60 headsets, which
I think we get for 7 or 8 bucks apiece, and sell to the agents at cost.

Thanks for that - they look good, and I found several recommendations for them 
after I got yours and started looking for them.

Further to this, I'm in the client office today and dealing directly with the 
users who are reporters and editors for a periodical and conduct many 
telephone interviews.  They want to use their old recording devices with the 
new phones, but are finding unpleasant audio experiences when they switch them 
over from the Nortel meridians to the Polycom IP330s.  So I'm looking for kit 
to use here as well.  Recommendations most welcome.

And in the case of one user, she is adamant she not be required to use a 
different recording device.  I don't know how to approach this except to try a 
different telephone or mess with Polycom gain settings that the manual advises 
not to touch.  Anybody been down this road - have any wisdom?



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--
Michael Graves
mgravesatmstvp.com
http://blog.mgraves.org
o713-861-4005
c713-201-1262
sip:[EMAIL PROTECTED]
skype mjgraves
fwd 54245




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Re: [asterisk-users] asteriskt38.com

[Andrew Joakimsen]

That isn't real T.38 support, it's just Packet2Packet bridging that
works correctly. Still need to use a Cisco gateway to support sending
the faxes somewhere on the PSTN. But it does work and it is reliable,
I use it every day.

On Mon, Oct 6, 2008 at 7:32 AM, Atis Lezdins [EMAIL PROTECTED] wrote:

 Actually it exists. 1.4 had passtrough mode and 1.6 can send and receive.


Hopefully it works. The one in CallWeaver doesn't.

On Mon, Oct 6, 2008 at 8:12 AM, Daniel Ferenci
[EMAIL PROTECTED] wrote:
 and there is a new application called fax gateway
 (http://bugs.digium.com/view.php?id=13405)
 that can do gatewaying between T30 and T38 and vice versa.

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Re: [asterisk-users] PoE switch recommendations?

[David Gibbons]

Right, it takes some doing to find a 1Gb switching phone though we ended up 
going with a system based on the Cisco 7941G-GE. This model supports all of the 
needed features including vlan tagging and 1Gb switching.


-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Augustyn
Sent: Monday, October 06, 2008 12:01 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] PoE switch recommendations?

Most phones support only 100M switching though  Unless you run separate
cabling for VoIP and data but then you would not need the 1G uplink.

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of
 David Gibbons
 Sent: Monday, October 06, 2008 11:48 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] PoE switch recommendations?

 Obviously we don't need 1Gb connections for VOIP :)

 Phones support pass through to the desktop and VLAN tagging.

 The need for 1Gb ports comes from wanting to have 1Gb at the desktop.

 Dave

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of
 Gordon Henderson
 Sent: Monday, October 06, 2008 11:29 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] PoE switch recommendations?

 On Mon, 6 Oct 2008, Ken D'Ambrosio wrote:

  Hey, all.  We're rolling out VoIP, and I'm wondering about PoE
  recommendations, as we're going to have to replace our
 current network
  equipment.  My first inclination would be to just plunk
 down the cash
  and do a Cisco system, but I'm relatively certain that
 would get shot
  down by finance.  Any recommendations for a couple-hundred-port
  solution with VLANs, PoE, and QoS?  Don't care much if it's in a
  single chassis or not, so long as it has Gbit uplinks.

 I'm curious as to why you want Gb uplinks on the switches?

 If we assume 100Kb/sec per phone .. (gross rounding, using
 100Kb/sec per phone, rather than ~80 - make the sums easier
 and builds in a margin) 10 calls per Mb/sec.

 So for a 24-port switch, 24 phones all talking to 24
 extensions off that switch, the max the uplink port is going
 to be pushing out is 2.4Mb/sec.

 For 200 extensions, say 9 x 24 port switches, with a single
 top-level (non PoE switch) switch with the PBX plugged in
 along side the 9 downlinks, that single PBX link will be
 carrying 2.4*9 = 22Mb/sec if all phones are in-use at the
 same time (and the PBX is carrying media)

 Now you may not want to build the network like that, but it
 seems that Gb is overkill just for the VoIP side of things.
 (And with that many extensions, I would suggest keeping all
 the phones on one set of switches)

 (Then again, it might not be possible to get big PoE switches
 without Gb uplinks, so it might be a moot point!)

 So satisfy my curiosity - why Gb uplinks?

 Cheers,

 Gordon

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Re: [asterisk-users] PoE switch recommendations?

[Jonathan C. Bailey]

We're using D-Link DES-3028P switches (24 10/100 + 4 gbit). They also have the 
DES-3052P which is a 48 port version of the switch. We're paying ~$500, I think 
for the 24 port version from Graybar.

-Jon


- Original Message -
From: David Gibbons [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: Monday, October 6, 2008 12:04:44 PM GMT -06:00 US/Canada Central
Subject: Re: [asterisk-users] PoE switch recommendations?

Right, it takes some doing to find a 1Gb switching phone though we ended up 
going with a system based on the Cisco 7941G-GE. This model supports all of the 
needed features including vlan tagging and 1Gb switching.


-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Augustyn
Sent: Monday, October 06, 2008 12:01 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] PoE switch recommendations?

Most phones support only 100M switching though  Unless you run separate
cabling for VoIP and data but then you would not need the 1G uplink.

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of
 David Gibbons
 Sent: Monday, October 06, 2008 11:48 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] PoE switch recommendations?

 Obviously we don't need 1Gb connections for VOIP :)

 Phones support pass through to the desktop and VLAN tagging.

 The need for 1Gb ports comes from wanting to have 1Gb at the desktop.

 Dave

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of
 Gordon Henderson
 Sent: Monday, October 06, 2008 11:29 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] PoE switch recommendations?

 On Mon, 6 Oct 2008, Ken D'Ambrosio wrote:

  Hey, all.  We're rolling out VoIP, and I'm wondering about PoE
  recommendations, as we're going to have to replace our
 current network
  equipment.  My first inclination would be to just plunk
 down the cash
  and do a Cisco system, but I'm relatively certain that
 would get shot
  down by finance.  Any recommendations for a couple-hundred-port
  solution with VLANs, PoE, and QoS?  Don't care much if it's in a
  single chassis or not, so long as it has Gbit uplinks.

 I'm curious as to why you want Gb uplinks on the switches?

 If we assume 100Kb/sec per phone .. (gross rounding, using
 100Kb/sec per phone, rather than ~80 - make the sums easier
 and builds in a margin) 10 calls per Mb/sec.

 So for a 24-port switch, 24 phones all talking to 24
 extensions off that switch, the max the uplink port is going
 to be pushing out is 2.4Mb/sec.

 For 200 extensions, say 9 x 24 port switches, with a single
 top-level (non PoE switch) switch with the PBX plugged in
 along side the 9 downlinks, that single PBX link will be
 carrying 2.4*9 = 22Mb/sec if all phones are in-use at the
 same time (and the PBX is carrying media)

 Now you may not want to build the network like that, but it
 seems that Gb is overkill just for the VoIP side of things.
 (And with that many extensions, I would suggest keeping all
 the phones on one set of switches)

 (Then again, it might not be possible to get big PoE switches
 without Gb uplinks, so it might be a moot point!)

 So satisfy my curiosity - why Gb uplinks?

 Cheers,

 Gordon

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Re: [asterisk-users] PoE switch recommendations?

[Norman Franke]

On Oct 6, 2008, at 12:56 PM, [EMAIL PROTECTED]  
wrote:

 We've been EXTREMELY happy with the HP 5400ZL series chassis switch.


Same here. We have 4 of them and they have worked very, very well. I  
have 25 polycom phones at present doing PoE from them and everything  
is working great. They are reasonably priced, come with a lifetime  
warranty and free software updates. (Unlike with Cisco!)

Norman Franke
Answering Service for Directors, Inc.
www.myasd.com



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[asterisk-users] Nice recording interfaces

[Chris Bagnall]

Greetings list,

What are people using for nice pretty recording/playback interfaces on their 
asterisk servers? I'm aware of ARI included with FreePBX, but are there any 
others that aren't linked to a larger GUI?

I'm looking for something that'll integrate nicely with a non-GUI, non-AGI 
asterisk box, purely to allow users to play back recordings that have been 
created with *1 during a call.

Alternatively, has anyone done some dialplan magic that'll store recordings in 
the user's mailbox (which can then be emailed via VM to email) that they'd be 
willing to share?

Thanks in advance.

Regards,

Chris





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Re: [asterisk-users] PoE switch recommendations?

[Jerry Jones]

The times they are a changing - or something like that.

while gb on phones is not the norm today, it s becoming more so on the  
higher end flavors and will continue to do so

since the life span of your switches will be several years, thinking  
ahead is a good thing

my only concern is having too many poe ports in a single switch,  
especially if it is a 1U model, running many with 24 ports poe I have  
had failures after a year or so. And with the new POE+ spec coming  
this will get even worse. Think adding more fans = more noise to get  
rid of the additional heat they generate


On Oct 6, 2008, at 12:04 PM, David Gibbons wrote:

 Right, it takes some doing to find a 1Gb switching phone though we  
 ended up going with a system based on the Cisco 7941G-GE. This model  
 supports all of the needed features including vlan tagging and 1Gb  
 switching.


 -Original Message-
 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 
 ] On Behalf Of Robert Augustyn
 Sent: Monday, October 06, 2008 12:01 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: Re: [asterisk-users] PoE switch recommendations?

 Most phones support only 100M switching though  Unless you run  
 separate
 cabling for VoIP and data but then you would not need the 1G uplink.

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of
 David Gibbons
 Sent: Monday, October 06, 2008 11:48 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] PoE switch recommendations?

 Obviously we don't need 1Gb connections for VOIP :)

 Phones support pass through to the desktop and VLAN tagging.

 The need for 1Gb ports comes from wanting to have 1Gb at the desktop.

 Dave

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of
 Gordon Henderson
 Sent: Monday, October 06, 2008 11:29 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] PoE switch recommendations?

 On Mon, 6 Oct 2008, Ken D'Ambrosio wrote:

 Hey, all.  We're rolling out VoIP, and I'm wondering about PoE
 recommendations, as we're going to have to replace our
 current network
 equipment.  My first inclination would be to just plunk
 down the cash
 and do a Cisco system, but I'm relatively certain that
 would get shot
 down by finance.  Any recommendations for a couple-hundred-port
 solution with VLANs, PoE, and QoS?  Don't care much if it's in a
 single chassis or not, so long as it has Gbit uplinks.

 I'm curious as to why you want Gb uplinks on the switches?

 If we assume 100Kb/sec per phone .. (gross rounding, using
 100Kb/sec per phone, rather than ~80 - make the sums easier
 and builds in a margin) 10 calls per Mb/sec.

 So for a 24-port switch, 24 phones all talking to 24
 extensions off that switch, the max the uplink port is going
 to be pushing out is 2.4Mb/sec.

 For 200 extensions, say 9 x 24 port switches, with a single
 top-level (non PoE switch) switch with the PBX plugged in
 along side the 9 downlinks, that single PBX link will be
 carrying 2.4*9 = 22Mb/sec if all phones are in-use at the
 same time (and the PBX is carrying media)

 Now you may not want to build the network like that, but it
 seems that Gb is overkill just for the VoIP side of things.
 (And with that many extensions, I would suggest keeping all
 the phones on one set of switches)

 (Then again, it might not be possible to get big PoE switches
 without Gb uplinks, so it might be a moot point!)

 So satisfy my curiosity - why Gb uplinks?

 Cheers,

 Gordon

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[asterisk-users] cdr,gsm file format

[Sriram]

Hi

1. What is the best way to convert wav (44000 Khz) to gsm format for 
asterisk ? I;ve tried sox command but the outcome is not satisfying...The 
built-in gsm files shipped with asterisk are simply superb ..How do i create 
gsm files of similar quality ? Can anyone help me out ? if sox is the only 
way can anyone tell me the exact command ?

2. Can Freepbx 2.5 installed above asterisk 1.6.0 or trunk versions ? in 
short does it support dahdi ?

3. If i cannot use Freepbx 2.5 where will the CDRs get stored and how i 
access them for writing my custom cdr program ? i saw that there is a cdr.so 
module that gets loaded - can it help me in anyway

Thanks in advance

Sriram
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[asterisk-users] Missing 'Queue' Application in 1.4.21.2

[Josiah Bryan]

Hey All -

Slight problem here - my install of 1.4.21.2 seems to be missing the 
Queue application:

asterisk*CLI core show version
Asterisk 1.4.21.2 built by root @ asterisk.productiveconcepts.com on a 
i686 running Linux on 2008-09-02 18:15:03 UTC

asterisk*CLI core show application Queue
Your application(s) is (are) not registered

I checked 'make meuselect' and it *seems* to indicate that app_queue was 
built:


**
Asterisk Module and Build Option Selection
**

Press 'h' for help.

   [*] 37. app_nbscat
   XXX 38. app_osplookup
   [*] 39. app_page
   [*] 40. app_parkandannounce
   [*] 41. app_playback
   [*] 42. app_privacy
   [*] 43. app_queue
   [*] 44. app_random
   [*] 45. app_read
   [*] 46. app_readfile
   ... More ...


 True Call Queueing
 Depends on: res_monitor(M)


Any ideas on how to figure this out?

Many thanks,
-josiah

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


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Re: [asterisk-users] asteriskt38.com

[Atis Lezdins]

On Mon, Oct 6, 2008 at 8:04 PM, Andrew Joakimsen [EMAIL PROTECTED] wrote:
 That isn't real T.38 support, it's just Packet2Packet bridging that
 works correctly. Still need to use a Cisco gateway to support sending
 the faxes somewhere on the PSTN. But it does work and it is reliable,
 I use it every day.

 On Mon, Oct 6, 2008 at 7:32 AM, Atis Lezdins [EMAIL PROTECTED] wrote:

 Actually it exists. 1.4 had passtrough mode and 1.6 can send and receive.


 Hopefully it works. The one in CallWeaver doesn't.

How do you mean - it doesn't? We currently use CallWeaver - Asterisk
1.4 - SIP Provider for sending and receiving faxes.

Whenever we'll switch to 1.6, we plan to get rid of CallWeaver, as it
has T.38 support in SendFax and ReceoveFax.

Regards,
Atis

-- 
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835

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Re: [asterisk-users] How to implement Ringing through a sound card for overhead paging

[Robert Augustyn]

Ok then how do you make that an night_bell as your extension?
Thanks 

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Doug Lytle
 Sent: Monday, October 06, 2008 12:31 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] How to implement Ringing 
 through a sound card for overhead paging
 
 Robert Augustyn wrote:
  Doug,
  That is interesting concept.
  How do you add this to a ring group and does it stop when 
 an extension 
  is picked up?

 
 It depends on how you have your ring group setup, I 
 personally only do this with a single extension.  And yes, 
 the bullhorn sound stops when the phone is answered.
 
 Doug
 
 
 -- 
  
 Ben Franklin quote:
 
 Those who would give up Essential Liberty to purchase a 
 little Temporary Safety, deserve neither Liberty nor Safety.
 
 
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Re: [asterisk-users] PoE switch recommendations?

[Dave Walker]

Ken D'Ambrosio wrote:
 Hey, all.  We're rolling out VoIP, and I'm wondering about PoE
 recommendations, as we're going to have to replace our current network
 equipment.  My first inclination would be to just plunk down the cash and
 do a Cisco system, but I'm relatively certain that would get shot down by
 finance.  Any recommendations for a couple-hundred-port solution with
 VLANs, PoE, and QoS?  Don't care much if it's in a single chassis or not,
 so long as it has Gbit uplinks.
   
Hi Ken,

I am rather impressed with Zyxel ES2024PWR,  I've used at least 40 of
these this year and not had any problems. I also can't recommend Zyxel's
support enough, I had initial concerns about the PoE budget and within a
couple of rings, I was through to someone who actually knew the product
inside out.

Kind Regards,
Dave Walker

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Re: [asterisk-users] Missing 'Queue' Application in 1.4.21.2

[Andres]

Josiah Bryan wrote:

Hey All -

Slight problem here - my install of 1.4.21.2 seems to be missing the 
Queue application:
  

What does the CLI output say when you start asterisk and it gets to the 
part where it tries to load app_queue.so?

Andres
http://www.neuroredes.com

asterisk*CLI core show version
Asterisk 1.4.21.2 built by root @ asterisk.productiveconcepts.com on a 
i686 running Linux on 2008-09-02 18:15:03 UTC

asterisk*CLI core show application Queue
Your application(s) is (are) not registered

I checked 'make meuselect' and it *seems* to indicate that app_queue was 
built:


**
Asterisk Module and Build Option Selection
**

Press 'h' for help.

   [*] 37. app_nbscat
   XXX 38. app_osplookup
   [*] 39. app_page
   [*] 40. app_parkandannounce
   [*] 41. app_playback
   [*] 42. app_privacy
   [*] 43. app_queue
   [*] 44. app_random
   [*] 45. app_read
   [*] 46. app_readfile
   ... More ...


 True Call Queueing
 Depends on: res_monitor(M)


Any ideas on how to figure this out?

Many thanks,
-josiah

  



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Re: [asterisk-users] cdr,gsm file format

[Dave Walker]

Hi Sirum,


Sriram wrote:
 Hi

 1. What is the best way to convert wav (44000 Khz) to gsm format for 
 asterisk ? I;ve tried sox command but the outcome is not satisfying...The 
 built-in gsm files shipped with asterisk are simply superb ..How do i create 
 gsm files of similar quality ? Can anyone help me out ? if sox is the only 
 way can anyone tell me the exact command ?
   
Sox should be suitable for this, however since Asterisk 1.4 inbuilt
conversion is supported.

*CLI help file convert
Usage: file convert file_in file_out

 2. Can Freepbx 2.5 installed above asterisk 1.6.0 or trunk versions ? in 
 short does it support dahdi ?
   
Why not just use what is shipped? Is there a killer feature that 1.6 or
trunk provides that the recommended versions don't support?  Although
Dahdi is now available in newer releases, it doesn't mean that legacy
Zaptel will suddenly stop working.  The Dahdi channel driver can also
present as Zap/channel still, so I would imagine it wouldn't cause too
many problems.  However, you would be better to seek assistance from the
FreePBX forum.
 3. If i cannot use Freepbx 2.5 where will the CDRs get stored and how i 
 access them for writing my custom cdr program ? i saw that there is a cdr.so 
 module that gets loaded - can it help me in anyway
   
The FreePBX forum would be a better place for this, I would imagine you
will get an answer sooner.

HTH

Kind Regards,
Dave Wa;ler

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Re: [asterisk-users] OT: headsets

[Jay R. Ashworth]

- Bill Michaelson [EMAIL PROTECTED] wrote:
 Further to this, I'm in the client office today and dealing directly
 with the users who are reporters and editors for a periodical and
 conduct many telephone interviews.  They want to use their old
 recording devices with the new phones, but are finding unpleasant
 audio experiences when they switch them over from the Nortel meridians
 to the Polycom IP330s.  So I'm looking for kit to use here as well. 
 Recommendations most welcome.

Are you switching from Nortel kit to Asterisk?

Why not set up a user function that starts a recording of the call inside
Asterisk itself and save the results to a Samba share where the users can 
drag them to their desktops?  Or not.

 And in the case of one user, she is adamant she not be required to use
 a different recording device.  I don't know how to approach this
 except to try a different telephone or mess with Polycom gain settings
 that the manual advises not to touch.  Anybody been down this road -
 have any wisdom?

What is she using now?  Some kind of analog recording adapter in the 4p4c
handset cord?

You may need to leave her for last, get a good solution going and prove it out 
with others, and then sell it to her boss and let *him* sell it to her.

Asterisk will do a *much* better job of recording than anything on the 
analog side, I would expect.

Cheers,
-- jra
-- 
Jay R. Ashworth   Baylink  [EMAIL PROTECTED]
Designer The Things I Think   RFC 2100
Ashworth  Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA  http://photo.imageinc.us +1 727 647 1274

 Those who cast the vote decide nothing.
 Those who count the vote decide everything.
   -- (Josef Stalin)


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Re: [asterisk-users] PoE switch recommendations?

[Jay R. Ashworth]

- Singer Wang [EMAIL PROTECTED] wrote:
 We've had some bad experiences with Linksys in general (prior to
 going VOIP) and avoided them. We're running now fully on the NetGear
 FS728TP switch (24 port 10/100 POE, 4 port 1000 uplink, and 2 slots for fiber
 modules).

While I haven't worked with their PoE, let me say that every piece of NetGear
kit I have ever touched is still working, solid as a rock, including the 5 
port hub in my bag.  :-)

Cheers,
-- jra
-- 
Jay R. Ashworth   Baylink  [EMAIL PROTECTED]
Designer The Things I Think   RFC 2100
Ashworth  Associates http://baylink.pitas.com '87 e24
St Petersburg FL USA  http://photo.imageinc.us +1 727 647 1274

 Those who cast the vote decide nothing.
 Those who count the vote decide everything.
   -- (Josef Stalin)


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Re: [asterisk-users] Missing 'Queue' Application in 1.4.21.2

[Josiah Bryan]

Gotta love flukes - after stopping asterisk and restarting so I could 
see the startup text, core show application Queue just worked . ???

Oh well. Thanks!
-josiah

Andres wrote:
 Josiah Bryan wrote:
 
 Hey All -

 Slight problem here - my install of 1.4.21.2 seems to be missing the 
 Queue application:
  

 What does the CLI output say when you start asterisk and it gets to the 
 part where it tries to load app_queue.so?
 
 Andres
 http://www.neuroredes.com
 
 asterisk*CLI core show version
 Asterisk 1.4.21.2 built by root @ asterisk.productiveconcepts.com on a 
 i686 running Linux on 2008-09-02 18:15:03 UTC

 asterisk*CLI core show application Queue
 Your application(s) is (are) not registered

 I checked 'make meuselect' and it *seems* to indicate that app_queue was 
 built:


**
Asterisk Module and Build Option Selection
**

Press 'h' for help.

   [*] 37. app_nbscat
   XXX 38. app_osplookup
   [*] 39. app_page
   [*] 40. app_parkandannounce
   [*] 41. app_playback
   [*] 42. app_privacy
   [*] 43. app_queue
   [*] 44. app_random
   [*] 45. app_read
   [*] 46. app_readfile
   ... More ...


 True Call Queueing
 Depends on: res_monitor(M)


 Any ideas on how to figure this out?

 Many thanks,
 -josiah

  

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


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[asterisk-users] Help with remote users

[Steve Anness]

I know I have asked about this before, but I thought that I would ask again
with some more detail and maybe someone will have an idea.  This is my first
time to be setting up an asterisk server and I have a server running.  I
sent Linksys PAP2T¹s to several remote users.  Only one out of the four
users actually work like they should.  One of the other users I am assuming
is behind a firewall on his wireless router and needs to open up the proper
ports.  However, I have two users in New York on a DSL connection and I
can¹t understand why things are happening like they are.

Here Is the situation.  Both users can plug in their ATAs and I can watch
the server output, they register and then they can make calls and I can call
them. Some time later (usually within minutes) the ATAs show to be
³unreachable² and I can no longer call; however, they can still make calls.

I am at a loss as to why this is happening.  They are hooking up right into
a DSL modem, I was thinking maybe it is because they are using DSL but I
don¹t understand why that makes a big difference because I have seen people
using vonage phones with the same PAP2T¹s on worse DSL connections than we
have in New York. 

I am new to asterisk, so any thoughts would be helpful.  I will be glad to
provide any more information that might be useful.

I thank you for your time.

Steve Anness
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[asterisk-users] Asterisk 1.4 or 1.6 ???

[Alejandro Facultad]

Dear all, I know there are two actual versions of Asterisk: 1.4 and 1.6.

My scenario is: SIP server with 100-150 SIP users, voice mail and maybe IVR. I 
will use GSM audio codec.

Maybe in the future I'll connect a E1 line to the PSTN.

What Asterisk version is better to me and why ???

Thank you.


A.F.



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Re: [asterisk-users] PoE switch recommendations?

[Karl Fife]

If you happen to be looking for a SMALL poe switch for a home or lab:

Think twice before you buy a netgear FS1xxP.  While they're great
because fanless, I've had 2 Netgear FS116p POE switches, and so far BOTH
have developed one or more 'dead' POE ports.  The manufacturer has a
LIFETIME warranty, but they have an advance-replacement charge, plus you
have to pay for your own shipping.  $60 so far this year on warranty
replacements.  According to support there is no 'Second Gen' hardware
design to fix the problem so I expect it will happen again.  Has anyone
else seen this?  

-Karl





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Re: [asterisk-users] Help with remote users

[Jerry Jones]

On Oct 6, 2008, at 1:53 PM, Steve Anness wrote:

I know I have asked about this before, but I thought that I would  
ask again with some more detail and maybe someone will have an  
idea.  This is my first time to be setting up an asterisk server and  
I have a server running.  I sent Linksys PAP2T’s to several remote  
users.  Only one out of the four users actually work like they  
should.  One of the other users I am assuming is behind a firewall  
on his wireless router and needs to open up the proper ports.   
However, I have two users in New York on a DSL connection and I  
can’t understand why things are happening like they are.


Here Is the situation.  Both users can plug in their ATAs and I can  
watch the server output, they register and then they can make calls  
and I can call them. Some time later (usually within minutes) the  
ATAs show to be “unreachable” and I can no longer call; however,  
they can still make calls.


do you have qualify=yes ??
Is asterisk on a public IP?


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Re: [asterisk-users] asteriskt38.com

[Andrew Joakimsen]

Maybe it works in more recent versions? I don't know. Anyways this is
getting rather off-topic.

On Mon, Oct 6, 2008 at 2:23 PM, Atis Lezdins [EMAIL PROTECTED] wrote:
 On Mon, Oct 6, 2008 at 8:04 PM, Andrew Joakimsen [EMAIL PROTECTED] wrote:
 Hopefully it works. The one in CallWeaver doesn't.

 How do you mean - it doesn't? We currently use CallWeaver - Asterisk
 1.4 - SIP Provider for sending and receiving faxes.

 Whenever we'll switch to 1.6, we plan to get rid of CallWeaver, as it
 has T.38 support in SendFax and ReceoveFax.

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Re: [asterisk-users] Help with remote users

[Steve Anness]

I am using NAT so the ATAs are configured with a proxy server.  Qualify is
set to yes.  Here is what is happening.  After they plug in the ATA on the
otherside, and things register and I can call and they can call.  After
several minutes I try to call and then get the ³no-service² message.  This
is with Qualify=yes.

   -- Executing [EMAIL PROTECTED]:1] Set(SIP/10.10.30.213-b7823fc0,
CDR(accountcode)=Hiramine) in new stack
-- Executing [EMAIL PROTECTED]:2] Set(SIP/10.10.30.213-b7823fc0,
CALLERID(all)=(Hiramine)  2545239280) in new stack
-- Executing [EMAIL PROTECTED]:3] Dial(SIP/10.10.30.213-b7823fc0,
SIP/17110-1SIP/17112-1|20| w) in new stack
[Oct  6 14:43:17] WARNING[11094]: app_dial.c:1196 dial_exec_full: Unable to
create channel of type 'SIP' (cause 3 - No route to destination)
[Oct  6 14:43:17] WARNING[11094]: app_dial.c:1196 dial_exec_full: Unable to
create channel of type 'SIP' (cause 3 - No route to destination)
  == Everyone is busy/congested at this time (2:0/0/2)
-- Executing [EMAIL PROTECTED]:4]
Playback(SIP/10.10.30.213-b7823fc0, ss-noservice) in new stack

If qualify is equal to no, then it just trys to ring, I get no errors it
just keeps trying (except the phone doesn¹t actually ring).

I just wrote an email to find out more about their network settings there.
To see if the ATAs are actually getting a private or public address.  If
they are getting a public address I suppose I can just set NAT=no and as
long as I can ping the public address and port 5060 isn¹t blocked by a
firewall than I should be able to resolve these issues.

Thanks for your time.

Steve Anness



On 10/6/08 2:20 PM, Jerry Jones [EMAIL PROTECTED] wrote:

 
 On Oct 6, 2008, at 1:53 PM, Steve Anness wrote:
 
  I know I have asked about this before, but I thought that I would ask again
 with some more detail and maybe someone will have an idea.  This is my first
 time to be setting up an asterisk server and I have a server running.  I sent
 Linksys PAP2T¹s to several remote users.  Only one out of the four users
 actually work like they should.  One of the other users I am assuming is
 behind a firewall on his wireless router and needs to open up the proper
 ports.  However, I have two users in New York on a DSL connection and I can¹t
 understand why things are happening like they are.
  
  Here Is the situation.  Both users can plug in their ATAs and I can watch
 the server output, they register and then they can make calls and I can call
 them. Some time later (usually within minutes) the ATAs show to be
 ³unreachable² and I can no longer call; however, they can still make calls.
 
 do you have qualify=yes ??
 Is asterisk on a public IP?
 
 
 
 
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Re: [asterisk-users] Tribox

[Guillermo Salas M.]

El lun, 06-10-2008 a las 10:57 +0100, Steven Howes escribió:
 Hi
 
 triXbox.org can answer these questions. Google may also give a  
 balanced view. But yes, i can assure you, people are using Trixbox  
 from Fonality.
 
 Steve
 
 On 6 Oct 2008, at 10:24, broadband Voice wrote:
 
  Anyone using Tribox from Fonality. I understand its open source and  
  free. Can I use it for a call center functionality? Thanks.


Give a try to elastix [1], it haves a very complete callcenter module.



[1] www.elastix.org


Best regards,


-- 
Guillermo Salas M.
Telconet S.A.
Calle 15 y Avenida 24 Esquina
Edificio Barre #2 Primer Piso
Telefono  : +593 5 262 7815
Celular   : +593 9 985 5138
International : +1 360 968 1701
e-mail: [EMAIL PROTECTED]
www   : http://www.telconet.net
SIP   : [EMAIL PROTECTED]

Linux User: 255902

Beat me, whip me, make me use Windows!

Please avoid the Top Posting, see
http://es.wikipedia.org/wiki/Top-posting


smime.p7s
Description: S/MIME cryptographic signature
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Re: [asterisk-users] Asterisk 1.4 or 1.6 ???

[Gordon Henderson]

On Mon, 6 Oct 2008, Alejandro Facultad wrote:

 Dear all, I know there are two actual versions of Asterisk: 1.4 and 1.6.

There is also 1.2. It may not be supported but there are 1000's of people 
out there (myself included) who are still using it.

 My scenario is: SIP server with 100-150 SIP users, voice mail and maybe 
 IVR. I will use GSM audio codec.

 Maybe in the future I'll connect a E1 line to the PSTN.

 What Asterisk version is better to me and why ???

The answer you are looking for is that you should be using a supported, 
stable version, and right now, 1.4 is the only one that fits. If I were 
starting today, I'd go with 1.4.

But I have to ask: Why GSM? If everything is in-house on the same LAN, 
then why not G711a? E1 is G711a, so you'd have to get the box to transcode 
to G711, which depending on the number of calls and CPU, might be an 
issue...

Gordon

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[asterisk-users] Matching *, + and # in the dialplan

[Karl Fife]

In several places online, and in the Asterisk F.O.T. book, there is a
warning against using '_.' saying: 
[it] should probably never be used.

However, the need often arises act on numeric extensions that begin with
*'s and #'s, and '+', and of course _X. does not match 

I have tried  exten = _[0-9*#+]. but that seems to be the functional
equivalent to _X. ignoring the addition of +,* and #.

Can someone suggest the best way to deal with this without resoring to a
highly repetitive/iterative dialplan?

Thanks in advance!

-Karl 



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Re: [asterisk-users] How to implement Ringing through a sound card for overhead paging

[Doug Lytle]

Robert Augustyn wrote:
 Ok then how do you make that an night_bell as your extension?
   

We have an after hours IVR, press 1 if you know the party that you're 
trying to reach, press 2 for Dial By Directory and press 3 for the night 
bell.


[incoming]

;
;* Check if call is within office hours,
;* if so, jump to the office-hours context
;* If not, continue on in the incoming
;* context.
;

exten = s,1,GotoIfTime(07:59-16:59|mon-fri|*|*?office-hours,s,1)
exten = s,n,Answer()
exten = s,n,Wait(1)

;**
;* If after hours then play the 'Welcome'
;* and office hours message Press 1 if you know
;* the extension or 2 for dial by name directory
;**

exten = s,n,Background(local/welcome)
exten = s,n,Background(local/business-hours)
exten = s,n,Background(local/8am-5pm)
exten = s,n,Background(local/press1-extension)
exten = s,n,Background(local/press2-directory)
exten = s,n,Background(local/press3-night-bell)

;*
;* Set timeouts
;*

exten = s,11,Set(TIMEOUT(response)=15)
exten = s,12,Set(TIMEOUT(digit)=2)

;*
;* If 1 is pressed, go to Dial by extension
;*

exten = 1,1,Goto(dial-by-extension,s,1)

;
;* If 2 is pressed, go to Dial by name
;

exten = 2,1,Goto(directory,s,1)

;
;* If 3 is pressed, go to Night Bell
;

exten = 3,1,Goto(night_bell,4173,1)


Doug


-- 
 
Ben Franklin quote:

Those who would give up Essential Liberty to purchase a little Temporary 
Safety, deserve neither Liberty nor Safety.


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Re: [asterisk-users] Asterisk 1.4 or 1.6 ???

[Brendan Martens]

On Oct 6, 2008, at 3:52 PM, Gordon Henderson wrote:

 The answer you are looking for is that you should be using a  
 supported,
 stable version, and right now, 1.4 is the only one that fits. If I  
 were
 starting today, I'd go with 1.4.

1.6.0 has just been released.
Personally I'd start with that because then you don't stuck with  
generation old features, and as you are just starting you aren't  
locked into any feature sets or syntax issues, etc.

Of course as it has just been released there are undoubtedly some bugs  
yet to be discovered, 1.4 has been around a while and will probably be  
easier to find support/documentation for.

Brendan

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Re: [asterisk-users] Tribox

[Tarek Sawah]

I would suggst th same solution if you haven't started using Trixbox yet.. 
maybe you shoul give Elastix a try .. it has modules made spcialy for call 
centers.. 
 
AHD Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP
Syria: +963 944 618286
USA: +1 347 562 2308

 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Mon, 6 
 Oct 2008 10:44:29 -0500 Subject: Re: [asterisk-users] Tribox  El lun, 
 06-10-2008 a las 10:57 +0100, Steven Howes escribió:  HitriXbox.org 
 can answer these questions. Google may also give a   balanced view. But 
 yes, i can assure you, people are using Trixbox   from Fonality.
 SteveOn 6 Oct 2008, at 10:24, broadband Voice wrote: Anyone 
 using Tribox from Fonality. I understand its open source andfree. Can 
 I use it for a call center functionality? Thanks.   Give a try to elastix 
 [1], it haves a very complete callcenter module.[1] www.elastix.org 
   Best regards,   --  Guillermo Salas M. Telconet S.A. Calle 15 y 
 Avenida 24 Esquina Edificio Barre #2 Primer Piso Telefono : +593 5 262 
 7815 Celular : +593 9 985 5138 International : +1 360 968 1701 e-mail : 
 [EMAIL PROTECTED] www : http://www.telconet.net SIP : [EMAIL PROTECTED]  
 Linux User : 255902  Beat me, whip me, make me use Windows!  Please avoid 
 the Top Posting, see http://es.wikipedia.org/wiki/Top-posting
_
Get more out of the Web. Learn 10 hidden secrets of Windows Live.
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Re: [asterisk-users] Tribox

[Ron Stephan]

 

 

I would add $.02…

 

I found the install on Elastix less than error free.  When the ISO can’t get
MySQL loaded without errors – I worry.

 

And the documentation (not that trixbox is well documented ) was weak IMHO.

 

 

Elvis

 

 

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tarek Sawah
Sent: Monday, October 06, 2008 1:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Tribox

 

I would suggst th same solution if you haven't started using Trixbox yet..
maybe you shoul give Elastix a try .. it has modules made spcialy for call
centers.. 

 

AHD Tarek Sawah

Integrated Digital Systems

CCNA, MCSE, RHCE, VoIP

Syria: +963 944 618286

USA: +1 347 562 2308

 

  _  

 From: [EMAIL PROTECTED]
 To: asterisk-users@lists.digium.com
 Date: Mon, 6 Oct 2008 10:44:29 -0500
 Subject: Re: [asterisk-users] Tribox
 
 El lun, 06-10-2008 a las 10:57 +0100, Steven Howes escribió:
  Hi
  
  triXbox.org can answer these questions. Google may also give a 
  balanced view. But yes, i can assure you, people are using Trixbox 
  from Fonality.
  
  Steve
  
  On 6 Oct 2008, at 10:24, broadband Voice wrote:
  
   Anyone using Tribox from Fonality. I understand its open source and 
   free. Can I use it for a call center functionality? Thanks.
 
 
 Give a try to elastix [1], it haves a very complete callcenter module.
 
 
 
 [1] www.elastix.org
 
 
 Best regards,
 
 
 -- 
 Guillermo Salas M.
 Telconet S.A.
 Calle 15 y Avenida 24 Esquina
 Edificio Barre #2 Primer Piso
 Telefono : +593 5 262 7815
 Celular : +593 9 985 5138
 International : +1 360 968 1701
 e-mail : [EMAIL PROTECTED]
 www : http://www.telconet.net
 SIP : [EMAIL PROTECTED]
 
 Linux User : 255902
 
 Beat me, whip me, make me use Windows!
 
 Please avoid the Top Posting, see
 http://es.wikipedia.org/wiki/Top-posting

  _  

Get more out of the Web. Learn 10 hidden secrets of Windows Live. Learn Now
http://windowslive.com/connect/post/jamiethomson.spaces.live.com-Blog-cns!5
50F681DAD532637!5295.entry?ocid=TXT_TAGLM_WL_getmore_092008 

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The message was checked by ESET Smart Security.

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Re: [asterisk-users] Asterisk 1.4 or 1.6 ???

[Jason Aarons (US)]

I would stick with 1.4 in production, how mad would you be if I gave you a cell 
phone with new code and it didn’t work?  Would you throw your cell phone at me 
if it cut us off during phone calls from a bug?  Some people are ok with trying 
new stuff, others it costs money when they lose business due to their phone 
system not working.  

 

I’ve noticed you can mess up computers, but phones get people mad when they 
don’t work.

 

From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alejandro 
Facultad
Sent: Monday, October 06, 2008 3:02 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Asterisk 1.4 or 1.6 ???

 

Dear all, I know there are two actual versions of Asterisk: 1.4 and 1.6.

My scenario is: SIP server with 100-150 SIP users, voice mail and maybe IVR. I 
will use GSM audio codec.

Maybe in the future I'll connect a E1 line to the PSTN.

What Asterisk version is better to me and why ???

Thank you.


A.F.

 




Yahoo! Cocina
Recetas prácticas y comida saludable
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[asterisk-users] ldap usage in 1.6.0

[Brendan Martens]

Hello, I'm trying to figure out how to implement 1.6.0 with some ldap  
integration, but it's hard to figure out if I can do what I want.
Basically I want to do only some lookup of values from ldap, as  
opposed to storing everything related to my sip users in ldap.

For instance, would there be a way to lookup only certain context  
items from an ldap attribute in extensions.conf? Or in sip.conf?

Something like this:

user.conf
[6000]
hassip = yes
hasiax = yes
userfrom = ldapattribute
insecure = route
secret = anotherldapattribute
type = friend
context = ldapattrib3


It's looking to me like the way that ldap with 1.6.0 is meant to be  
used is more as a replacement for certain .conf files, like how odbc  
is used, and not really for referencing occasionally. I'm pretty new  
to asterisk so any guidance on this issue would be welcomed.


Maybe if I explain a little overview of my end goal someone can help  
me more efficiently.
I have an ldap directory on an OSX server, I want to create asterisk  
extensions for all of those users based on the extension, name, and  
password held in the ldap database. But I do not want to store  
whole .configs in ldap.

Any ideas on how to go about this would be great.

Brendan Martens


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Re: [asterisk-users] Tribox

[Guillermo Salas M.]

El lun, 06-10-2008 a las 10:57 +0100, Steven Howes escribió:
 Hi
 
 triXbox.org can answer these questions. Google may also give a  
 balanced view. But yes, i can assure you, people are using Trixbox  
 from Fonality.
 
 Steve
 
 On 6 Oct 2008, at 10:24, broadband Voice wrote:
 
  Anyone using Tribox from Fonality. I understand its open source and  
  free. Can I use it for a call center functionality? Thanks.


Give a try to elastix [1], it haves a very complete callcenter module.



[1] www.elastix.org


Best regards,


-- 
Guillermo Salas M.
Telconet S.A.
Calle 15 y Avenida 24 Esquina
Edificio Barre #2 Primer Piso
Telefono  : +593 5 262 7815
Celular   : +593 9 985 5138
International : +1 360 968 1701
e-mail: [EMAIL PROTECTED]
www   : http://www.telconet.net
SIP   : [EMAIL PROTECTED]

Linux User: 255902

Beat me, whip me, make me use Windows!

Please avoid the Top Posting, see
http://es.wikipedia.org/wiki/Top-posting


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Description: S/MIME cryptographic signature
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Re: [asterisk-users] Tribox

[Tarek Sawah]

i haven't facedthse tpe of problems you mentioned with mysql.. but there is one 
thing that you need to edit the sip.conf iax.conf or you can use the sample 
ones in the samples folder.. 
other than that.. i've been with trixbox for over three years now.. it has 
problems with it comes to Queues and call center services.. i've been 
struggling with it for months now .. plus as per an earlier post i had here 
i'm having problems convincing trixbox to accept my dia plans on two of my 
three servers.. while i tred installing elastix more than 10 times on different 
machins.. i don't have those problems.. 
besides!!! on trixbox you need to add th ip of the freepbx mirrors to upgrade 
your modules.. and you have to manipulate your php files to be able to upgrade 
your box from the website..
 
AHD Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP
Syria: +963 944 618286
USA: +1 347 562 2308

From: [EMAIL PROTECTED]: [EMAIL PROTECTED]: Mon, 6 Oct 2008 13:23:11 
-0700Subject: Re: [asterisk-users] Tribox






 
 
I would add $.02…
 
I found the install on Elastix less than error free.  When the ISO can’t get 
MySQL loaded without errors – I worry.
 
And the documentation (not that trixbox is well documented ) was weak IMHO.
 
 
Elvis
 
 


From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tarek SawahSent: 
Monday, October 06, 2008 1:15 PMTo: Asterisk Users Mailing List - 
Non-Commercial DiscussionSubject: Re: [asterisk-users] Tribox
 
I would suggst th same solution if you haven't started using Trixbox yet.. 
maybe you shoul give Elastix a try .. it has modules made spcialy for call 
centers.. 
 
AHD Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP
Syria: +963 944 618286
USA: +1 347 562 2308
 



 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Mon, 6 
 Oct 2008 10:44:29 -0500 Subject: Re: [asterisk-users] Tribox  El lun, 
 06-10-2008 a las 10:57 +0100, Steven Howes escribió:  HitriXbox.org 
 can answer these questions. Google may also give a   balanced view. But 
 yes, i can assure you, people are using Trixbox   from Fonality.
 SteveOn 6 Oct 2008, at 10:24, broadband Voice wrote: Anyone 
 using Tribox from Fonality. I understand its open source andfree. Can 
 I use it for a call center functionality? Thanks.   Give a try to elastix 
 [1], it haves a very complete callcenter module.[1] www.elastix.org 
   Best regards,   --  Guillermo Salas M. Telconet S.A. Calle 15 y 
 Avenida 24 Esquina Edificio Barre #2 Primer Piso Telefono : +593 5 262 
 7815 Celular : +593 9 985 5138 International : +1 360 968 1701 e-mail : 
 [EMAIL PROTECTED] www : http://www.telconet.net SIP : [EMAIL PROTECTED]  
 Linux User : 255902  Beat me, whip me, make me use Windows!  Please avoid 
 the Top Posting, see http://es.wikipedia.org/wiki/Top-posting



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[asterisk-users] No reply to our critical packet

[Andrew Joakimsen]

I am using a Polycom 501 SIP phone behind NAT. Asterisk server is
public with no NAT... everything works on the Asterisk end just fine
EXCEPT that I can never check voice mail

After about 30 seconds the call drops with these messagess:

[Sep 30 23:47:48] WARNING[26819]: chan_sip.c:1950 retrans_pkt: Maximum
retries exceeded on transmission
[EMAIL PROTECTED] for seqno 2 (Critical
Response)
[Sep 30 23:47:48] WARNING[26819]: chan_sip.c:1972 retrans_pkt: Hanging
up call [EMAIL PROTECTED] - no reply to our
critical packet.

It seems to me that the problem is the way Asterisk is handling this
critical packet -- of course it can not be sent to 192.168.1.54, the
phone is at that IP behind a NAT and the Asterisk server is not. I can
make any other phone call from this same phone as long as it is not
voicemail and I can be on the line for hours with no problem.

I am really at a loss here. I have searched a bit and come up with
nothing other than blaming the UA. I know the Polycoms dont have the
best NAT support but besides this it works problem-free. It's odd I
can make a call anywhere else even for hours and not have any issues
at all but 30 seconds into a voicemail call it just drops


app5*CLI sip show peer 17865221569
app5*CLI

 * Name   : 17865221569
 Secret   : Set
 MD5Secret: Not set
 Context  : blended-lcr
 Subscr.Cont. : sla_stations
 Language : en
 AMA flags: Unknown
 Transfer mode: closed
 CallingPres  : Presentation Allowed, Not Screened
 Callgroup:
 Pickupgroup  :
 Mailbox  : 17865221569
 VM Extension : 14193016245
 LastMsgsSent : 0/0
 Call limit   : 2
 Dynamic  : Yes
 Callerid :  CENSORED
 MaxCallBR: 256 kbps
 Expire   : 63
 Insecure : no
 Nat  : Always
 ACL  : No
 T38 pt UDPTL : Yes
 CanReinvite  : No
 PromiscRedir : No
 User=Phone   : Yes
 Video Support: No
 Trust RPID   : No
 Send RPID: No
 Subscriptions: Yes
 Overlap dial : No
 DTMFmode : rfc2833
 LastMsg  : 0
 ToHost   :
 Addr-IP : 74.CENSORED.213 Port 5060
 Defaddr-IP  : 0.0.0.0 Port 5060
 Reg. exten   :
 Def. Username: 17865221569
 SIP Options  : (none)
 Codecs   : 0x104 (ulaw|g729)
 Codec Order  : (g729:20,ulaw:20)
 Auto-Framing:  No
 Status   : OK (130 ms)
 Useragent: PolycomSoundPointIP-SPIP_501-UA/3.0.1.0032
 Reg. Contact : sip:[EMAIL PROTECTED]


app5*CLI core show version
Asterisk 1.4.21.1 built by root @ app5 on a i686 running Linux on
2008-07-09 01:41:43 UTC

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Re: [asterisk-users] t1 cards

[Nick B.]

Have you considered fiber?
Nick
On Sun, Oct 05, 2008 at 07:52:54PM -0700, Eric Fort wrote:
 Here's a couple of distances I'm looking to cover (distances are +- 10%):
 
 1 at 400M
 1 at 600M
 1 at 1800M
 1 at 2400M
 
 some of these links may already have pots circuits complete with occasional
 ringing voltage in the same conduit (but likely not the same cable).  how
 far can I push the distance of E1 over copper using only 2 cards back to
 back?
 
 Eric
 
 On Sun, Oct 5, 2008 at 5:19 PM, Andrew Joakimsen [EMAIL PROTECTED]wrote:
 
  How much further than 300m? It might be very well possible to just
  lower the speed to 10M and just use that If you already have some
  quality Cat5 cable between both points it's worth a shot.  I support
  some sites with this arrangement and I've had to find 10M hubs for
  replacement hardware (the previous guy insisted that only a particular
  model HP print server would work, coincidently that model only has a
  10M Ethernet port)... it's not something I would advise someone to
  setup but if cost is a concern I wouldn't rule it out -- it certainly
  can work and be reliable in the real world.
 
 
 
  On Fri, Oct 3, 2008 at 3:14 AM, Eric Fort [EMAIL PROTECTED] wrote:
   yes, more than 300 meters (longer than copper based ethernet allows).
   Yes
   to E1, as I understand it, it's just a config change on many cards
  anyway.
   I'm specificly looking at pci based t1/e1 cards because I'm finding
  single
   port cards on ebay going for 100-200 usd.  in some cases I may want to
  drive
   a channel bank at the far end, thus t1/e1.  anyone have experience on how
   far these pci based cards will drive when wired back to back?
 

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Re: [asterisk-users] Tribox

[Guillermo Salas M.]

El lun, 06-10-2008 a las 13:23 -0700, Ron Stephan escribió:
 And the documentation (not that trixbox is well documented ) was weak
 IMHO.


Try reading:

http://www.elastixconnection.com/downloads/elastix_without_tears.pdf


Best regards,

-- 
Guillermo Salas M.
Telconet S.A.
Calle 15 y Avenida 24 Esquina
Edificio Barre #2 Primer Piso
Telefono  : +593 5 262 7815
Celular   : +593 9 985 5138
International : +1 360 968 1701
e-mail: [EMAIL PROTECTED]
www   : http://www.telconet.net
SIP   : [EMAIL PROTECTED]

Linux User: 255902

Beat me, whip me, make me use Windows!

Please avoid the Top Posting, see
http://es.wikipedia.org/wiki/Top-posting


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Re: [asterisk-users] Vitelity Asterisk configuration help

[Darren Severino]

Stephen,   What exactly are you trying to accomplish? If you want basic call
in/out you're just about there. Changes need to be made in your
extensions.conf. Your phones, by default, are in the [default] context. In
other words when making a call it looks for extensions here. To allow
outbound calling include your outgoing context within the default. To
include it, at the bottom of the default context add include = outgoing
either of these should allow outgoing calling. As for incoming, add a Goto
as follows.

[inbound]
exten = 9045622082,1,Answer
exten = 9045622082,n,Goto(default,101,1)

That equates to goto the default context, extension 101, at the 1st
priority which is your Dial command.

Best Regards,Darren Severino


On Sat, Oct 4, 2008 at 1:30 PM, Stephen Reese [EMAIL PROTECTED] wrote:

 I have a Asterisk server setup and I am able to connect to the server
 using a soft client 'x-lite' and call and leave a message on my second
 extension 102. I have setup a Vitelity account and add what I believe
 to be the correct information to my sip.conf and extension.conf. I
 would like to setup incoming and outgoing calls with voicemail
 support. I've searched all over but many of the full configurations
 that are available are a bit complex. Any tips or recommendations to
 get up and running would be great.

 sip.conf
 Code:

 [general]
 register = rsreese:[EMAIL PROTECTED]:5060
 context=default ; Default context for incoming calls
 realm=ns1.neocipher.net ; Realm for digest authentication
 bindport=5060   ; UDP Port to bind to (SIP standard
 port is 5060)
 bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to
 all)
 srvlookup=yes   ; Enable DNS SRV lookups on outbound calls
 domain=neocipher.net; Set default domain for this host
 [101]
 type=friend ; allows incoming and outgoing calls
 username=101
 secret=test81
 mailbox=101
 callerid=Stephen 101
 host=dynamic
 dtmfmode=rfc2833
 canreinvite=no
 reinvite=no
 disallow=all
 allow=gsm
 [102]
 type=friend ; allows incoming and outgoing calls
 username=102
 secret=test81
 mailbox=102
 callerid=(Bob 101)
 host=dynamic
 dtmfmode=rfc2833
 canreinvite=yes
 allowguest=yes
 insecure=very
 promiscredir=yes
 musicclass=default  ; Sets the default music on hold class
 for all SIP calls
 [authentication]
 [vitel-inbound] ;(exact format/casing required)
 type=friend
 host=inbound18.vitelity.net
 context=inbound ;(ext-did or from-trunk for [EMAIL PROTECTED])
 username=rsreese
 secret=pass
 allow=all
 insecure=very
 canreinvite=no
 [vitel-outbound] ;(exact format/casing required)
 type=friend
 host=outbound.vitelity.net
 context=inbound ;(ext-did or from-trunk for [EMAIL PROTECTED])
 username=rsreese
 fromuser=rsreese
 trustrpid=yes
 sendrpid=yes
 secret=pass
 allow=all
 canreinvite=no


 extensions.conf
 Code:

 [general]
 static=yes
 writeprotect=yes

 [globals]

 [default]

 exten = 101,1,Dial(SIP/101,20)
 exten = 101,2,Voicemail(102)

 exten = 102,1,Dial(SIP/102,20)
 exten = 102,2,Voicemail(102)

 exten=*98,1,VoiceMailMain([EMAIL PROTECTED])   ;This
 automatically calls the right mailbox using the ${CALLERIDNUM}
 variable in the current context (var ${CONTEXT}).

 [outgoing]
 exten = _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED])
 exten = _011.,1,Dial(SIP/[EMAIL PROTECTED])

 exten = _911,1,Dial(SIP/[EMAIL PROTECTED])

 [inbound]
 exten = 9045622082,1,Answer


 voicemail.conf
 Code:

 [general]
 format=wav49|gsm|wav
 serveremail=asterisk
 attach=yes
 skipms=3000
 maxsilence=10
 silencethreshold=128
 maxlogins=3
 emaildateformat=%A, %B %d, %Y at %r
 sendvoicemail=yes   ; Context to Send voicemail from [option 5
 from the advanced menu]
 [zonemessages]
 eastern=America/New_York|'vm-received' Q 'digits/at' IMp
 central=America/Chicago|'vm-received' Q 'digits/at' IMp
 central24=America/Chicago|'vm-received' q 'digits/at' H N 'hours'
 military=Zulu|'vm-received' q 'digits/at' H N 'hours' 'phonetic/z_p'
 [default]
 101 = 123,Stephen Rese,[EMAIL PROTECTED]
 102 = 123,Bob Dole,[EMAIL PROTECTED]
 1234 = 4242,Example Mailbox,[EMAIL PROTECTED]
 [other]
 1234 = 5678,Company2 User,[EMAIL PROTECTED]

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Re: [asterisk-users] AEL and swap from macros to contexts

[Steve Murphy]

On Mon, 2008-10-06 at 18:25 +0200, Pavel Jezek wrote:
 
 Atis Lezdins wrote:
  On Mon, Oct 6, 2008 at 5:21 PM, Pavel Jezek [EMAIL PROTECTED] wrote:

  Hi, according to discussion on asterisk IRC, where people said, that
  macros will be depracated, I tried to migrate from macros to contexts
  and Gosub
  but if I try to use gosub in extensions.ael, ael compiler complains,
  that I shouln't use Gosub app,
  but I can't find ael keyword, that will be Gosub equivalent, or can I
  ignore this ael warnings? thanks
  PJ
 
 
  LOG: lev:3 file:pval.c  line:2521 func: check_pval_item  Warning: file
  /etc/asterisk/extensions.ael, line 36-36: application call to Gosub
  affects flow of control, and needs to be re-written using AEL if, while,
  goto, etc. keywords instead!
  
 
  Hi,
 
  In definition use:
 
  macro set_record(A,B) {
// do something
  }
 
  And for calling:
 
  set_record(${CALLERID(NUM)},${EXTEN});
 
  It will automatically be translated to GoSub in 1.6, but will remain
  as Macro in 1.4.

 
 yes, I know, but I hear on IRC, that macros will be deprecated and 
 suggestion was to move to contexts,
 personaly I would like also move away from macros, because macros have 
 some limitations, eg. variable number of arguments isn't possible with 
 classic macros,
 macros also require variable to be defined in macro definition (that is 
 needless, because I'm referecing to ARG1, ARG2 etc. inside macros)
 so I definitively agree with moving from macros to contexts, only one 
 bad thing is compiler warning, when I try to Gosub to context (as macro 
 replacement)
 PJ
 
 

Pavel--

Yes, you can ignore the warnings and go ahead and hardcoded gosub calls
into your source. I didn't upgrade 1.4 to use gosub-instead-of-macro
because
the key element ended up being calling gosub with arguments, which
didn't
make it into 1.4.

Someday, when you upgrade from 1.4 to 1.6, you will have to change
all your gosub's to use the argument passing feature, if you hardcode
gosubs now. Or, you can backport the gosub-with-arguments feature to
1.4,
and use 1.6 AEL to compile... which will give you some future
portability
when you do move to 1.6...

Sorry to make simple things sound so complicated!

murf


 
  Regards,
  Atis
 
 
 

 
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Re: [asterisk-users] No reply to our critical packet

[SIP]

This message is usually caused by Asterisk not receiving an ACK after 
about 30 seconds of attempts. There are countless misconfigured UAs and 
proxies out there that don't handle ACK well, so it would be nice to be 
able to turn this 'feature' off. What's annoying is that the explanation 
has always been If we can't get an ACK, we can't send any RTP data.   
This is patently false, as the RTP will often work fine even if ACK 
handling is misconfigured (we see it all the time).

But alas. As far as I can tell, there's no way to disable this check. I 
suppose I could code around it, but not being the world's most 
proficient C coder, I'm always afraid I'll break something else. ;)

N.


Andrew Joakimsen wrote:
 I am using a Polycom 501 SIP phone behind NAT. Asterisk server is
 public with no NAT... everything works on the Asterisk end just fine
 EXCEPT that I can never check voice mail

 After about 30 seconds the call drops with these messagess:

 [Sep 30 23:47:48] WARNING[26819]: chan_sip.c:1950 retrans_pkt: Maximum
 retries exceeded on transmission
 [EMAIL PROTECTED] for seqno 2 (Critical
 Response)
 [Sep 30 23:47:48] WARNING[26819]: chan_sip.c:1972 retrans_pkt: Hanging
 up call [EMAIL PROTECTED] - no reply to our
 critical packet.

 It seems to me that the problem is the way Asterisk is handling this
 critical packet -- of course it can not be sent to 192.168.1.54, the
 phone is at that IP behind a NAT and the Asterisk server is not. I can
 make any other phone call from this same phone as long as it is not
 voicemail and I can be on the line for hours with no problem.

 I am really at a loss here. I have searched a bit and come up with
 nothing other than blaming the UA. I know the Polycoms dont have the
 best NAT support but besides this it works problem-free. It's odd I
 can make a call anywhere else even for hours and not have any issues
 at all but 30 seconds into a voicemail call it just drops


 app5*CLI sip show peer 17865221569
 app5*CLI

  * Name   : 17865221569
  Secret   : Set
  MD5Secret: Not set
  Context  : blended-lcr
  Subscr.Cont. : sla_stations
  Language : en
  AMA flags: Unknown
  Transfer mode: closed
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup:
  Pickupgroup  :
  Mailbox  : 17865221569
  VM Extension : 14193016245
  LastMsgsSent : 0/0
  Call limit   : 2
  Dynamic  : Yes
  Callerid :  CENSORED
  MaxCallBR: 256 kbps
  Expire   : 63
  Insecure : no
  Nat  : Always
  ACL  : No
  T38 pt UDPTL : Yes
  CanReinvite  : No
  PromiscRedir : No
  User=Phone   : Yes
  Video Support: No
  Trust RPID   : No
  Send RPID: No
  Subscriptions: Yes
  Overlap dial : No
  DTMFmode : rfc2833
  LastMsg  : 0
  ToHost   :
  Addr-IP : 74.CENSORED.213 Port 5060
  Defaddr-IP  : 0.0.0.0 Port 5060
  Reg. exten   :
  Def. Username: 17865221569
  SIP Options  : (none)
  Codecs   : 0x104 (ulaw|g729)
  Codec Order  : (g729:20,ulaw:20)
  Auto-Framing:  No
  Status   : OK (130 ms)
  Useragent: PolycomSoundPointIP-SPIP_501-UA/3.0.1.0032
  Reg. Contact : sip:[EMAIL PROTECTED]


 app5*CLI core show version
 Asterisk 1.4.21.1 built by root @ app5 on a i686 running Linux on
 2008-07-09 01:41:43 UTC

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Re: [asterisk-users] ldap usage in 1.6.0

[Olivier]

2008/10/6 Brendan Martens [EMAIL PROTECTED]

 Hello, I'm trying to figure out how to implement 1.6.0 with some ldap
 integration, but it's hard to figure out if I can do what I want.
 Basically I want to do only some lookup of values from ldap, as
 opposed to storing everything related to my sip users in ldap.

 For instance, would there be a way to lookup only certain context
 items from an ldap attribute in extensions.conf? Or in sip.conf?

 Something like this:

 user.conf
 [6000]
 hassip = yes
 hasiax = yes
 userfrom = ldapattribute
 insecure = route
 secret = anotherldapattribute
 type = friend
 context = ldapattrib3


 It's looking to me like the way that ldap with 1.6.0 is meant to be
 used is more as a replacement for certain .conf files, like how odbc
 is used, and not really for referencing occasionally. I'm pretty new
 to asterisk so any guidance on this issue would be welcomed.


 Maybe if I explain a little overview of my end goal someone can help
 me more efficiently.
 I have an ldap directory on an OSX server, I want to create asterisk
 extensions for all of those users based on the extension, name, and
 password held in the ldap database. But I do not want to store
 whole .configs in ldap.

 Any ideas on how to go about this would be great.

 Brendan Martens


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Hi,

I don't have much experience with LDAP and Asterisk, but here are some
thoughts about it :

1. I would provide Asterisk its own LDAP directory and synchronize it with
entreprise directory as I think it should be simpler to synchronize 2 LDAP
directories than coordinate Asterisk and Active Directory evolutions.

2. IMHO, many people are confusing SIP secrets (from sip.conf) which somehow
authenticate hardware with user passwords which authenticate persons. I
wouldn't try to make those 2 values equal.

3. Asterisk's LDAP directory should be the reference for anything related to
telephony. Changes could be automatically propagated from Asterisk to
corporate directory.

4. Corporate directory should be the reference for user management. Changes
should be manually propagated from corporate directory to Asterisk as I
don't believe it could be easy to allocate nor free telephony resources
whenever a user is created or deleted in corporate directory.

Hope this helps ...
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Re: [asterisk-users] ldap usage in 1.6.0

[Brendan Martens]

Thanks for the reply. Hmmm


 1. I would provide Asterisk its own LDAP directory and synchronize  
 it with entreprise directory as I think it should be simpler to  
 synchronize 2 LDAP directories than coordinate Asterisk and Active  
 Directory evolutions.

This may work, but my end goal is really to simplify, not complicate.  
If I can't get the information I need for sip users etc from ldap then  
I'll just have to skip it... I need to not be the only person that can  
manage whatever setup I end up with. : (


 2. IMHO, many people are confusing SIP secrets (from sip.conf) which  
 somehow authenticate hardware with user passwords which authenticate  
 persons. I wouldn't try to make those 2 values equal.

Hmm, once again with the integration and the simplifying, one of the  
biggest reasons I want access to ldap is to be able to authenticate  
there, I really don't want to introduce another place to manage  
authentication. Most of my users will be using sip phones and I don't  
want to give them another user/password combo to remember. : (


 3. Asterisk's LDAP directory should be the reference for anything  
 related to telephony. Changes could be automatically propagated from  
 Asterisk to corporate directory.

 4. Corporate directory should be the reference for user management.  
 Changes should be manually propagated from corporate directory to  
 Asterisk as I don't believe it could be easy to allocate nor free  
 telephony resources whenever a user is created or deleted in  
 corporate directory.

Not quite sure I follow here... If a user was deleted from my ldap  
directory, the corresponding sip phone should fail registration, right?





Having thought some more about my issue I think I can perhaps ask my  
question more succinctly: is it possible to get dynamic (or  
realtime) data from ldap within the various .conf files?

If there is not a convenient function for getting this in the .conf  
files, what if I somehow specified a global variable within the  
res_ldap.conf and referenced that value inside the other .conf files?  
Is this possible? Sorry if these are very basic questions, I just  
haven't been able to find answers to them. : (

Brendan

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Re: [asterisk-users] No reply to our critical packet

[Andrew Joakimsen]

The odd thing is on this particular phone it only happens when you
call voicemail.

It is certainly a bug in Asterisk, not the UA. Asterisk is trying to
send to 192.168.1.x which obviously is not possible. Something in the
NAT support is not working right.

On Mon, Oct 6, 2008 at 3:06 PM, SIP [EMAIL PROTECTED] wrote:
 This message is usually caused by Asterisk not receiving an ACK after
 about 30 seconds of attempts. There are countless misconfigured UAs and
 proxies out there that don't handle ACK well, so it would be nice to be
 able to turn this 'feature' off. What's annoying is that the explanation
 has always been If we can't get an ACK, we can't send any RTP data.
 This is patently false, as the RTP will often work fine even if ACK
 handling is misconfigured (we see it all the time).

 But alas. As far as I can tell, there's no way to disable this check. I
 suppose I could code around it, but not being the world's most
 proficient C coder, I'm always afraid I'll break something else. ;)

 N.


 Andrew Joakimsen wrote:
 I am using a Polycom 501 SIP phone behind NAT. Asterisk server is
 public with no NAT... everything works on the Asterisk end just fine
 EXCEPT that I can never check voice mail

 After about 30 seconds the call drops with these messagess:

 [Sep 30 23:47:48] WARNING[26819]: chan_sip.c:1950 retrans_pkt: Maximum
 retries exceeded on transmission
 [EMAIL PROTECTED] for seqno 2 (Critical
 Response)
 [Sep 30 23:47:48] WARNING[26819]: chan_sip.c:1972 retrans_pkt: Hanging
 up call [EMAIL PROTECTED] - no reply to our
 critical packet.

 It seems to me that the problem is the way Asterisk is handling this
 critical packet -- of course it can not be sent to 192.168.1.54, the
 phone is at that IP behind a NAT and the Asterisk server is not. I can
 make any other phone call from this same phone as long as it is not
 voicemail and I can be on the line for hours with no problem.

 I am really at a loss here. I have searched a bit and come up with
 nothing other than blaming the UA. I know the Polycoms dont have the
 best NAT support but besides this it works problem-free. It's odd I
 can make a call anywhere else even for hours and not have any issues
 at all but 30 seconds into a voicemail call it just drops


 app5*CLI sip show peer 17865221569
 app5*CLI

  * Name   : 17865221569
  Secret   : Set
  MD5Secret: Not set
  Context  : blended-lcr
  Subscr.Cont. : sla_stations
  Language : en
  AMA flags: Unknown
  Transfer mode: closed
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup:
  Pickupgroup  :
  Mailbox  : 17865221569
  VM Extension : 14193016245
  LastMsgsSent : 0/0
  Call limit   : 2
  Dynamic  : Yes
  Callerid :  CENSORED
  MaxCallBR: 256 kbps
  Expire   : 63
  Insecure : no
  Nat  : Always
  ACL  : No
  T38 pt UDPTL : Yes
  CanReinvite  : No
  PromiscRedir : No
  User=Phone   : Yes
  Video Support: No
  Trust RPID   : No
  Send RPID: No
  Subscriptions: Yes
  Overlap dial : No
  DTMFmode : rfc2833
  LastMsg  : 0
  ToHost   :
  Addr-IP : 74.CENSORED.213 Port 5060
  Defaddr-IP  : 0.0.0.0 Port 5060
  Reg. exten   :
  Def. Username: 17865221569
  SIP Options  : (none)
  Codecs   : 0x104 (ulaw|g729)
  Codec Order  : (g729:20,ulaw:20)
  Auto-Framing:  No
  Status   : OK (130 ms)
  Useragent: PolycomSoundPointIP-SPIP_501-UA/3.0.1.0032
  Reg. Contact : sip:[EMAIL PROTECTED]


 app5*CLI core show version
 Asterisk 1.4.21.1 built by root @ app5 on a i686 running Linux on
 2008-07-09 01:41:43 UTC

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Re: [asterisk-users] PoE switch recommendations?

[Andrew Joakimsen]

I've used the smaller ones, I think 8pt with 4pt PoE stuck in drop
ceilings and such to power ORiNOCO APs and never had an issue.

As for the larger switches I've used Linksys SRW224P. I have a few
running for a few years without issues. They have GB uplink but the
individual ports are 100M.

On Mon, Oct 6, 2008 at 12:12 PM, Karl Fife
[EMAIL PROTECTED] wrote:
 If you happen to be looking for a SMALL poe switch for a home or lab:

 Think twice before you buy a netgear FS1xxP.  While they're great
 because fanless, I've had 2 Netgear FS116p POE switches, and so far BOTH
 have developed one or more 'dead' POE ports.  The manufacturer has a
 LIFETIME warranty, but they have an advance-replacement charge, plus you
 have to pay for your own shipping.  $60 so far this year on warranty
 replacements.  According to support there is no 'Second Gen' hardware
 design to fix the problem so I expect it will happen again.  Has anyone
 else seen this?

 -Karl





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Re: [asterisk-users] PoE switch recommendations?

[Karl Fife]

 I've used the smaller ones, I think 8pt with 4pt PoE stuck in drop
 ceilings and such to power ORiNOCO APs and never had an issue.

That's a good data point.  We too have an FS108p (like yours) and it has
been reliable so far.  For us it's only been the FS116p's that have
failed.  It seems possible that the 16 port version has one or more
components that are just 'overdriven' variants of the 8 port version and
is therfore being overworked, perhaps leading to failure.  It seems
especially probable being a fanless design. 

-Karl 

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Re: [asterisk-users] No reply to our critical packet

[Atis Lezdins]

On Tue, Oct 7, 2008 at 2:22 AM, Andrew Joakimsen [EMAIL PROTECTED] wrote:
 The odd thing is on this particular phone it only happens when you
 call voicemail.

 It is certainly a bug in Asterisk, not the UA. Asterisk is trying to
 send to 192.168.1.x which obviously is not possible. Something in the
 NAT support is not working right.

Hi,

You should get SIP traces to see why Asterisk is trying to reply to 192.168.1.x.

To do this, enter sip set debug on in asterisk CLI, and post us a
log of call reaching voicemail and disconnecting.

Regards,
Atis


 On Mon, Oct 6, 2008 at 3:06 PM, SIP [EMAIL PROTECTED] wrote:
 This message is usually caused by Asterisk not receiving an ACK after
 about 30 seconds of attempts. There are countless misconfigured UAs and
 proxies out there that don't handle ACK well, so it would be nice to be
 able to turn this 'feature' off. What's annoying is that the explanation
 has always been If we can't get an ACK, we can't send any RTP data.
 This is patently false, as the RTP will often work fine even if ACK
 handling is misconfigured (we see it all the time).

 But alas. As far as I can tell, there's no way to disable this check. I
 suppose I could code around it, but not being the world's most
 proficient C coder, I'm always afraid I'll break something else. ;)

 N.


 Andrew Joakimsen wrote:
 I am using a Polycom 501 SIP phone behind NAT. Asterisk server is
 public with no NAT... everything works on the Asterisk end just fine
 EXCEPT that I can never check voice mail

 After about 30 seconds the call drops with these messagess:

 [Sep 30 23:47:48] WARNING[26819]: chan_sip.c:1950 retrans_pkt: Maximum
 retries exceeded on transmission
 [EMAIL PROTECTED] for seqno 2 (Critical
 Response)
 [Sep 30 23:47:48] WARNING[26819]: chan_sip.c:1972 retrans_pkt: Hanging
 up call [EMAIL PROTECTED] - no reply to our
 critical packet.

 It seems to me that the problem is the way Asterisk is handling this
 critical packet -- of course it can not be sent to 192.168.1.54, the
 phone is at that IP behind a NAT and the Asterisk server is not. I can
 make any other phone call from this same phone as long as it is not
 voicemail and I can be on the line for hours with no problem.

 I am really at a loss here. I have searched a bit and come up with
 nothing other than blaming the UA. I know the Polycoms dont have the
 best NAT support but besides this it works problem-free. It's odd I
 can make a call anywhere else even for hours and not have any issues
 at all but 30 seconds into a voicemail call it just drops


 app5*CLI sip show peer 17865221569
 app5*CLI

  * Name   : 17865221569
  Secret   : Set
  MD5Secret: Not set
  Context  : blended-lcr
  Subscr.Cont. : sla_stations
  Language : en
  AMA flags: Unknown
  Transfer mode: closed
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup:
  Pickupgroup  :
  Mailbox  : 17865221569
  VM Extension : 14193016245
  LastMsgsSent : 0/0
  Call limit   : 2
  Dynamic  : Yes
  Callerid :  CENSORED
  MaxCallBR: 256 kbps
  Expire   : 63
  Insecure : no
  Nat  : Always
  ACL  : No
  T38 pt UDPTL : Yes
  CanReinvite  : No
  PromiscRedir : No
  User=Phone   : Yes
  Video Support: No
  Trust RPID   : No
  Send RPID: No
  Subscriptions: Yes
  Overlap dial : No
  DTMFmode : rfc2833
  LastMsg  : 0
  ToHost   :
  Addr-IP : 74.CENSORED.213 Port 5060
  Defaddr-IP  : 0.0.0.0 Port 5060
  Reg. exten   :
  Def. Username: 17865221569
  SIP Options  : (none)
  Codecs   : 0x104 (ulaw|g729)
  Codec Order  : (g729:20,ulaw:20)
  Auto-Framing:  No
  Status   : OK (130 ms)
  Useragent: PolycomSoundPointIP-SPIP_501-UA/3.0.1.0032
  Reg. Contact : sip:[EMAIL PROTECTED]


 app5*CLI core show version
 Asterisk 1.4.21.1 built by root @ app5 on a i686 running Linux on
 2008-07-09 01:41:43 UTC

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-- 
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: 

Re: [asterisk-users] PoE switch recommendations?

[Daniel Hazelbaker]

On Oct 6, 2008, at 4:31 PM, Andrew Joakimsen wrote:

 As for the larger switches I've used Linksys SRW224P. I have a few
 running for a few years without issues. They have GB uplink but the
 individual ports are 100M.

I recently purchased a few SRW208P switches.  They work fine.  If you  
run Windows.  Granted a lot of people run windows instead of Mac or  
Linux, but be aware (to those looking) that the SRW line of switches  
REQUIRE Internet Explorer on Windows.  The support site says it is  
recommended, but even the login page does not work properly on  
anything but IE on Windows.  For me, as a Mac user, it is enough to  
not buy any more of those ever again.

On the other side, We have a dozen switches in the SGE2000, SGE2000P  
and SGE2010P series that all work perfectly and with any browser I  
have tried.  Some may wonder why I would buy a 24/48-port fully  
gigabit switch.  It is because I don't want to have to think, or even  
keep track, of which port on the wall is PoE and which is Gigabit.  I  
just want to plug it in and work.  I want to be able to tell my staff  
Just plug your phone in and it will work, don't worry about trying to  
find a power adapter.  The extra money is worth not trying to keep  
track of which is which.  The SGE2000 switches we bought before the  
SGE2000P came down in price (it used to be like 4 times the non-PoE  
version).  Now, at a $220 difference ($880 verses $660) there is no  
question.

Beyond that, they work great.  VLAN setup and use is simple.  Link  
Aggregation works perfectly.  STP works like a charm (no more running  
around trying to figure out what idiot patched their wall jack into  
another wall jack).  The ability to transfer the switches  
configuration to a TFTP server (and HTTP in the 2010 version, 2000 is  
using old firmware) makes it easy to backup the configuration and  
restore it to a new switch in the event of complete failure.

Daniel

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[asterisk-users] Asterisk/AJAM Console

[Forrest Beck]

I was just looking to see if anyone knows about an open source app  
using the xml interface.  I just started dabbling with the xml  
interface a little bit and it helps to look at what others are doing.   
I am looking for a console type app for the operator.  Very simple  
operations like transfer, hold, status, park, etc.

We are currently using the FOP, but I always have to update the fop  
configs to add a new button after creating/changing an extension.  Our  
data is in a realtime DB, so I guess I could build a console that uses  
the realtime db and the xml interface.  Anyone else in the same boat?

Thanks.

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Re: [asterisk-users] Asterisk 1.4 or 1.6 ???

[Al Baker]

Brendan Martens wrote:

On Oct 6, 2008, at 3:52 PM, Gordon Henderson wrote:

  
The answer you are looking for is that you should be using a  
supported,
stable version, and right now, 1.4 is the only one that fits. If I  
were

starting today, I'd go with 1.4.



1.6.0 has just been released.
Personally I'd start with that because then you don't stuck with  
generation old features, and as you are just starting you aren't  
locked into any feature sets or syntax issues, etc.


Of course as it has just been released there are undoubtedly some bugs  
yet to be discovered, 1.4 has been around a while and will probably be  
easier to find support/documentation for.


  

Quote are undoubtedly some bugs  yet to be discovered

Good laugh, look at the BUG reporting site.
1.4 had how many HUNDREDS of bugs reported ?
How many more continue to flow in ?

and  you think someone should go to 1.6 ?

May want to reconsider that.
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[asterisk-users] regcontext

[Nhadie]

hi all,

just wondering what's happening here:

i have a pap2 and an spa941. everytime i call my spa from my pap2 i can 
see it being added dynamically on the regcontext:

[Oct  7 11:59:08] -- Saved useragent Linksys/SPA942-5.2.8 for peer 
100100
[Oct  7 11:59:08] -- Added extension '100100' priority 1 to 
sipregcontext


but from spa to pap2 i dont see it, i looked at the difference on the 
config, under SIP:

On PAP2:
Handle VIA received: yesHandle VIA rport: yes   
Insert VIA received: yesInsert VIA rport: yes   
STUN Enable: yesSTUN Test Enable:

On SPA:
Handle VIA received: no Handle VIA rport: no
Insert VIA received: no Insert VIA rport: no
STUN Enable: no STUN Test Enable:


so i change pap2 to the same config as spa, and now i can see it being 
added on the regcontext when i call it

[Oct  7 11:59:04] -- Saved useragent Linksys/PAP2-3.1.22(LS) for 
peer 100200
[Oct  7 11:59:04] -- Added extension '100200' priority 1 to 
sipregcontext
[Oct  7 11:59:04] -- Saved useragent Linksys/PAP2-3.1.22(LS) for 
peer 100300
[Oct  7 11:59:04] -- Added extension '100300' priority 1 to 
sipregcontext

does that have to do with the STUN? or the Handle VIA thing. just trying 
to understand these behavior.

TIA

Regards,
Nhadie



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Re: [asterisk-users] Vitelity Asterisk configuration help

[Stephen Reese]

 Stephen,   What exactly are you trying to accomplish? If you want basic
 call
 in/out you're just about there. Changes need to be made in your
 extensions.conf. Your phones, by default, are in the [default] context.
 In
 other words when making a call it looks for extensions here. To allow
 outbound calling include your outgoing context within the default. To
 include it, at the bottom of the default context add include =
 outgoing
 either of these should allow outgoing calling. As for incoming, add a
 Goto
 as follows.
 
 [inbound]
 exten = 9045622082,1,Answer
 exten = 9045622082,n,Goto(default,101,1)
 
 That equates to goto the default context, extension 101, at the 1st
 priority which is your Dial command.
 
 Best Regards,Darren Severino

Thanks I am now able to make incoming calls but I'm still unable to call
out. Notice anything else off.

extension.conf

[general]
 static=yes
 writeprotect=yes

[globals]

[default]

exten = 101,1,Dial(SIP/101,20)
exten = 101,2,Voicemail(102)
exten = 101,102,Voicemail(102)

exten=*98,1,VoiceMailMain([EMAIL PROTECTED])   ;This automatically
calls the right mailbox using the ${CALLERIDNUM} variable in the current
context (var ${CONTEXT}).
include = outgoing
include = inbound

[outgoing]
exten = _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED])
exten = _011.,1,Dial(SIP/[EMAIL PROTECTED])

; e911 must be enabled. see DIDs  NPANXXNXXX  Action  e911
exten = _911,1,Dial(SIP/[EMAIL PROTECTED])

[inbound]
exten = 9045622082,1,Goto(default,101,1)


Sip.conf

[general]
register = rsreese:[EMAIL PROTECTED]:5060
context=default ; Default context for incoming calls
realm=ns1.neocipher.net ; Realm for digest authentication
bindport=5060   ; UDP Port to bind to (SIP standard port is
5060)
bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to
all)
srvlookup=yes   ; Enable DNS SRV lookups on outbound calls

domain=neocipher.net; Set default domain for this host

[101]
type=friend ; allows incoming and outgoing calls
username=101
secret=test81
mailbox=101
callerid=\Stephen\ 101
host=dynamic
nat=yes
dtmfmode=rfc2833
canreinvite=no
reinvite=no

musicclass=default  ; Sets the default music on hold class for
all SIP calls
language=en ; Default language setting for all
users/peers

[authentication]

[vitel-inbound] ;(exact format/casing required)
type=friend
host=inbound18.vitelity.net
context=inbound ;(ext-did or from-trunk for [EMAIL PROTECTED])
username=rsreese
secret=key
allow=all
insecure=very
canreinvite=no

[vitel-outbound] ;(exact format/casing required)
type=friend
host=outbound.vitelity.net
context=inbound ;(ext-did or from-trunk for [EMAIL PROTECTED])
username=rsreese
fromuser=rsreese
trustrpid=yes
sendrpid=yes
secret=key
allow=all
canreinvite=no


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Re: [asterisk-users] AEL and swap from macros to contexts

[Pavel Jezek]

Steve Murphy wrote:
 On Mon, 2008-10-06 at 18:25 +0200, Pavel Jezek wrote:
   
 Atis Lezdins wrote:
 
 On Mon, Oct 6, 2008 at 5:21 PM, Pavel Jezek [EMAIL PROTECTED] wrote:
   
   
 Hi, according to discussion on asterisk IRC, where people said, that
 macros will be depracated, I tried to migrate from macros to contexts
 and Gosub
 but if I try to use gosub in extensions.ael, ael compiler complains,
 that I shouln't use Gosub app,
 but I can't find ael keyword, that will be Gosub equivalent, or can I
 ignore this ael warnings? thanks
 PJ


 LOG: lev:3 file:pval.c  line:2521 func: check_pval_item  Warning: file
 /etc/asterisk/extensions.ael, line 36-36: application call to Gosub
 affects flow of control, and needs to be re-written using AEL if, while,
 goto, etc. keywords instead!
 
 
 Hi,

 In definition use:

 macro set_record(A,B) {
   // do something
 }

 And for calling:

 set_record(${CALLERID(NUM)},${EXTEN});

 It will automatically be translated to GoSub in 1.6, but will remain
 as Macro in 1.4.
   
   
 yes, I know, but I hear on IRC, that macros will be deprecated and 
 suggestion was to move to contexts,
 personaly I would like also move away from macros, because macros have 
 some limitations, eg. variable number of arguments isn't possible with 
 classic macros,
 macros also require variable to be defined in macro definition (that is 
 needless, because I'm referecing to ARG1, ARG2 etc. inside macros)
 so I definitively agree with moving from macros to contexts, only one 
 bad thing is compiler warning, when I try to Gosub to context (as macro 
 replacement)
 PJ


 

 Pavel--

 Yes, you can ignore the warnings and go ahead and hardcoded gosub calls
 into your source. I didn't upgrade 1.4 to use gosub-instead-of-macro
 because
 the key element ended up being calling gosub with arguments, which
 didn't
 make it into 1.4.

 Someday, when you upgrade from 1.4 to 1.6, you will have to change
 all your gosub's to use the argument passing feature, if you hardcode
 gosubs now. Or, you can backport the gosub-with-arguments feature to
 1.4,
 and use 1.6 AEL to compile... which will give you some future
 portability
 when you do move to 1.6...

 Sorry to make simple things sound so complicated!

 murf

   
murf, thank you for clear answer,
currently, I'm using asterisk trunk (and 1.6 also),
do you plan to remove quite confusing AEL warnings, that appears, when I 
try to hardcode Gosub with arguments into ael dialplan?
PJ


 Regards,
 Atis



   
   
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