[Asterisk-Users] May be OT , but comparing

[ram]

Hi all
This might be OT question

But still i want to ask , if any one have idea about.

Does any one point me to URL SER Vs Asterisk

advantage and disadvantage

where to use SER, and where to use Asterisk

thanks

ram
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Re: [Asterisk-Users] Pickup() h323

[Jeremy McNamara]

Hamid Hashemi wrote:
I did try it again without success. I did check the debug logs and there 
is nothing special there about any errors. Following the logs it says 
that the connection is established but no Voice and no Tone. here is my 
scenario :


I have a SIP phone which make a SIP call to asterisk with G729 Codec. 
The Asterisk then make an H323 call to the external peer with G729 codec 
again and it should make bridge between these 2 calls ( 1 incomming and 
1 outgoing )
I checked it with OH323 with the same scenario and it is working well. 
But for H323 I couldn't make the call. Any idea ?




Without debug there not even a chance that ANYONE can assist you.


Jeremy McNamara
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[Asterisk-Users] AAstra 9133i register double account.. ??

[nik600]

hi

i've got an AAstra 9133i ip phone, when i've bought it, i've set it to
use a SIP/400 account on my asterisk, then, i've changed settings and
i've set set phone to use a SIP/500 account .

now, when i connect the phone to tthe network, it register itself on
asterisk with both accounts!!!

   -- Registered SIP '500' at 192.168.100.188 port 5060 expires 120
   -- Registered SIP '400' at 192.168.100.188 port 5060 expires 120

how can i avoid this? i want to register only SIP 500!!

this is a piece of my sip.conf


[400]
username=400
type=friend
secret=400
record_out=Adhoc
record_in=Adhoc
qualify=no
port=5060
nat=never
[EMAIL PROTECTED]
host=dynamic
dtmfmode=rfc2833
context=from-internal
canreinvite=no
callerid= 400 My user

[500]
username=500
type=friend
secret=500
record_out=Adhoc
record_in=Adhoc
qualify=no
port=5060
nat=never
[EMAIL PROTECTED]
host=dynamic
dtmfmode=rfc2833
context=from-internal
canreinvite=no
callerid= 500 Postazione 500


can you help me please?
thanks
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[Asterisk-Users] Quintum ASM400 FXO configuration

[Abdul Lateef]

Hi All,

This is my first day i brought ASM400 for Calling Card
porpuse, I created AGI script for calling crad, so if
some one is dialing 12345 our Calling Card AGI script
will start to asking PIN,Phone number etc

The Script is working well with SIPURA 3000. But i
wanted to configure in quintum because this model is
already having 4FXO line. So if any once can give me
some usefull link or the idea for FXO configuration i
will be appricate. 

I am looking the following diagram:

PSTN  FXO Line (Quintum)
FXO Line  [EMAIL PROTECTED]

Thats all.

Please help me for this issue. Thank very much in
advance.

Thank You
Abdul

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[Asterisk-Users] Re: gotoif

[Shaun]

What doug said didnt work for me, anybody else having this problem the below 
appears to have resolved it.


exten = s,n,Gotoif($[${menuopt} = ]?1)

-- 

~Shaun





Doug Lytle [EMAIL PROTECTED] wrote in message 
news:[EMAIL PROTECTED]
 Shaun wrote:


 Apr  7 01:32:13 WARNING[24248]: ast_expr2.fl:183 ast_yyerror: 
 ast_yyerror(): syntax error: syntax error, unexpected TOK_EQ, expecting 
 TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN; Input:



 The dial plan works and all, it's just i want those warnings to go away!



 This has been covered a few time in the last 2 months.  You need to 
 initialize the variable:

Set(holdopt=0)

 Before doing any testing with it.

 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] Centrex Question

[Rich Adamson]

Brian Capouch wrote:
I haven't dealt with Centrex for a long time, and one of my customers is 
being courted heavily by a Sprint salesperson.


Am I not correct in assuming that each line of Centrex corresponds to 
an extension in the PBX world?


This site has 2 POTS lines and 5 extensions, and they told me that for 
the same thing they're paying right now (~$40/POTS line) they will be 
getting two Centrex lines that will do the same thing.


The way I understood it, each of those two Centrex lines is an extension.

In general, would they still be paying their POTS fees, too?

Sorry for the noise, but I can't discuss this intelligently with them, 
and that's hurting me.


Historically, there were two forms of Centrex provided by US telcos. 
Centrex, which was based on a shared pbx typically located on the telco 
promises, and, CO Centrex which was based on the Central Office switch 
with added software features. I'd have to guess the majority of the 
current Centrex implementations are actually CO Centrex now, however I 
did run into a recent case (a college) where Qwest was still using a CO 
based pbx.


Both were tariffed by the telcos with rates that were different then 
normal central office business lines, presumably due to shared 
maintenance costs (and features). (Eg, smoke and mirrors.)


Regardless of which implementation Sprint might be using, from the 
central office perspective, a Centrex line is the same physical thing as 
a pots business line. If your customer has been quoted two Centrex 
lines, its two physical connections (or max two simultaneous calls).


It is possible they might also be providing more then two extension 
numbers using something like distinctive ringing, or, some form of 
subscriber carrier system to mux two extensions over a single line 
(doubt that), or mapping five extension numbers at the CO onto two 
physical Centrex lines.


The more likely case is Sprint is simply displacing your customer's on 
site equipment (presumably a key system) with two lines (with different 
numbers or extensions) and five phones. Nothing more, nothing less.


A Centrex line is the same thing as a pots line from a customer's bill 
perspective. In your case, the bill will only have two Centrex line 
charges (no pots charges), plus any features they happen to be selling 
as optional items. (Optional items are typically voicemail services, 
voicemail LED on their phones, possibly custom calling features, etc, etc.)


Without more info, that's about the best guess you're going to get.

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[Asterisk-Users] HELP !!!!!

[hgaillac-sip]

Hello,

I wish to set a sip uri sip:[EMAIL PROTECTED]

I use ser for authorization and authentication
(registrar rtpproxy and outbound proxy)

I use asterisk 1.2.5 with realtime .
the info is used as a hunt group so i add in
extension.conf

[info]
exten = info,1,Answer()
exten = info,n,Dial(Sip/84,10)
exten = info,n,Dial(Sip/85,10)
exten = info,n,Hangup

Ser forward sip:[EMAIL PROTECTED] to asterisk but this one
ask for authentication 407 .

How can I disable this ?

Harry









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Re: [Asterisk-Users] HELP !!!!!

[Jeremy McNamara]

[EMAIL PROTECTED] wrote:

How can I disable this ?




sip.conf:

[general]
context=info



Jeremy McNamara
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Re: [Asterisk-Users] HELP !!!!!

[hgaillac-sip]

Jeremy

I set in sip.conf 
 
[general]
context=sip

and 

[sip]
include = info
include = support


[info]
exten = info,1,Answer()
exten = info,n,Dial(Sip/84,10)
exten = info,n,Dial(Sip/85,10)
exten = info,n,Hangup

where info and support are hunt group



--- Jeremy McNamara [EMAIL PROTECTED] a écrit :

 [EMAIL PROTECTED] wrote:
  How can I disable this ?
 
 
 
 sip.conf:
 
 [general]
 context=info
 
 
 
 Jeremy McNamara
 







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Re: [Asterisk-Users] HELP !!!!!

[Jeremy McNamara]

[EMAIL PROTECTED] wrote:
I set in sip.conf 



And you have reloaded asterisk, right?



Jeremy McNamara
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Re: [Asterisk-Users] HELP !!!!!

[hgaillac-sip]

Yes I reload and restart it 
--- Jeremy McNamara [EMAIL PROTECTED] a écrit :

 [EMAIL PROTECTED] wrote:
  I set in sip.conf 
 
 
 And you have reloaded asterisk, right?
 
 
 
 Jeremy McNamara
 







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[Asterisk-Users] HELP !!!!!

[hgaillac-sip]

Hello,

I wish to set a sip uri sip:[EMAIL PROTECTED]

I use ser for authorization and authentication
(registrar rtpproxy and outbound proxy)

I use asterisk 1.2.5 with realtime .
the info is used as a hunt group so i add in
extension.conf

[info]
exten = info,1,Answer()
exten = info,n,Dial(Sip/84,10)
exten = info,n,Dial(Sip/85,10)
exten = info,n,Hangup

Ser forward sip:[EMAIL PROTECTED] to asterisk but this one
ask for authentication 407 .

How can I disable authentication, is it a bug with
realm ?

Harry














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FW: [Asterisk-Users] CallerID

[Technical Support]

Jay,

I contacted you many times regarding the script, whether you planned to
update it, suggestions for features, etc.  You did not respond to any of my
later emails.  Similarly, there was discussion between list members
regarding whether this script was orphaned after changes to 411.com made the
reverse lookup non-functional - for a long time.  I assumed responsibility
for updating the script as a courtesy to Asterisk users.

Your comments about spelling, resale, etc. are abrasive, unproductive, and
misleading.  Not only is the script available without charge on the web
site, credit to you remains with the script - in fact even the download link
of the web site gives you credit!  And of course, why would I update the
script and then encourage users to download an older version from another
site?

If you have time to dedicate to the cid_rewrite project terrific - I would
rather see one stream benefit all users.  Let's work to integrate changes
going forward.  If you would prefer not to, I would be pleased to rename the
script so that there is no confusion.

Regards,
Michelle


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[Asterisk-Users] Problems getting Asterisk to connect to sipgate - Times out

[Paul A Brown]

Hi All,

Tryinng to get Asterisk to connect to sp gate and keep getting these errors.

Apr  8 12:46:53 NOTICE[5227]: chan_sip.c:5267 sip_reg_timeout:--  
Registration for '[EMAIL PROTECTED]' timed out, trying again (Attempt 
#2)

   -- parse_srv: SRV mapped to host sipgate.co.uk, port 5060

Any ideas?

sip.conf

[general]

register =1234567:[EMAIL PROTECTED]/1234567

[sipgate]
type=peer
fromdomain=sipgate.co.uk
host=sipgate.co.uk
context=sipgate-inbound
diallow=all
allow=ulaw
allow=alaw

extensions.conf


[sipgate-inbound]
exten = 1234567,1,NoOp(Incoming call on sipgate.co.uk)
exten = 1234567,2,Ringing
exten = 1234567,3,Wait(1)
exten = 1234567,4,Dial(SIP/220${CONSOLE},30)
exten = 1234567,5,VoiceMail(u30)
exten = 1234567,6,Hangup

ALSO.

How can I have multiple sipgate accounts register and have then call 
different internal sip extensions?


Thanks

Paul 


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[Asterisk-Users] question about DISA

[Ronaldo Chan]

Lists,
 
  Hi, good day, i was being task to create a DISA access for internal
purpose of the company, i'm having a problem to work with it with
authentication, but i think it's really a straight forward thing to do,
can someone enlight me on this. thanks
 
sample code snippet
 
 exten = 5,Goto(inward,s,1)
 
[inward]
 
   exten = s,1,Disa(1234|outgoing)
   ; DISA apps supposed to ask me a password but it's not
instead it's drop me immedietly to a dial tone
   exten = s,2,Hangup
 
My Workaround.
 
 exten = s,1,Authenticate(1234)
 exten = s,2,Disa(no-password|outgoing)
 
 
Thanks
 
Ronald
 
 
 
 
 
  
 

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Re: [Asterisk-Users] Re: gotoif

[Doug Lytle]

Shaun wrote:
What doug said didnt work for me, anybody else having this problem the below 
appears to have resolved it.



exten = s,n,Gotoif($[${menuopt} = ]?1)
  


In your first post you weren't using the quotes around the option, this 
time you are... I didn't notice that:


exten = s,202,GotoIf($[${holdopt} = 1 ]?4)

Try the same line with the quotes this time.

exten = s,202,GotoIf($[${holdopt} = 1 ]?4)


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] DIALSTATUS for Multiple Dialled Numbers

[Moises Silva]

When you pass several Dial strings only the last exited channel
DIALSTATUS is saved. In the case that 1 of the channels answer, the
status will be ANSWER obviously, but if the second fails because of
CONGESTION and the first because NOANSWER, the last exited channel
dial status will be set.

Regards

On 4/7/06, Alexander Lopez [EMAIL PROTECTED] wrote:
 Without modifications to Dial, I don't think so.

 However,

 Dial(Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED])

 [dialstatus]
 _X.,1,Set(TECH=${CUT(${EXTEN},-,1)})
 _X.,2,Set(DEVICE=${CUT(${EXTEN},-,2)})
 _X.,3,Dial(${TECH}/${DEVICE}||)


 Or something like this...

 I would also create Variable name to track each one.


 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of
 Douglas Garstang
 Sent: Friday, April 07, 2006 2:21 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [Asterisk-Users] DIALSTATUS for Multiple Dialled Numbers
 
 Folks,
 
 When I have a dial string like this:
 
 Dial(SIP/3254101SIP/3254102,20,tr)
 
 and I want to check the ${DIALSTATUS} variable after the
 dial, how do I know which number I am getting the variable for?
 
 And, what about this?
 
 Dial(SIP/3254101SIP/[EMAIL PROTECTED],20,tr)
 
 What happens in that case? How can I get the ${DIALSTATUS}
 variable for EACH NUMBER dialled?
 
 Thanks,
 Doug.
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[Asterisk-Users] 407 proxy authentication

[hgaillac-sip]

Hello,

look at this I can't receive calls from other domains
I wish sip:[EMAIL PROTECTED] are forwarded to asterisk
however this one spend its time to ask 407 proxy
authentication.

asterisk 1.2.5 + realtime


how can i fix this problem what' wrong ?

extension.conf

[info]
exten = info,1,Answer()
exten = info,2,Dial(Sip/84,10)
exten = info,3,Dial(Sip/85,10)
exten = info,4,Hangup

serveur1*CLI sip show user info load
serveur1*CLI

  * Name   : info
  Secret   : Not set
  MD5Secret: Not set
  Context  : info
  Language : fr
  AMA flags: Unknown
  CallingPres  : Presentation Allowed, Not Screened
  Call limit   : 0
  Callgroup:
  Pickupgroup  :
  Callerid :  
  ACL  : No
  Codec Order  : (g729|ilbc|gsm|ulaw|alaw)

harry








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[Asterisk-Users] Re: [OT] Centrex Question

[Levolorman]

Hi 
gang:

Regarding the Centrex, I have digital centrex and joined this group to see 
if I could learn a bit more about VOIP (as my PBX died unexpectedly and I'm 
moving towards a hosted PBX over VOIP).

Here's how Centrex works. You get a number of Centrex lines and then 
run your extensions as you would with a standard POTS system. Each Centrex 
line has its' own number, so in a large business, you would use each line as a 
unique extension giving a direct dial capability to an individual 
extension.

You're describing a small business. If they have 2 Centrex lines, 
they can be on 2 simultaneous calls. Those calls can be incoming, 
outgoing, conference, fax, etc. However, you cannot be on more calls at 
the same time than the number of lines that you have, other than conference or 
switching between calls (as the switching is done central office). 

In my office, we have 3 Centrex voice lines (1 dedicated fax), 6 extensions 
and 5 employees. It's rare that we have 5 employees in at the same time 
and even more unlikely that 3 people are on the phone at the same time, so the 
combination of a PBX with Centrex was to our advantage.

Why Centrex for us? Because when I locked into it, 11 years ago, I 
locked my telephone pricing for 7 years. It was a time of uncertainty due 
to deregulation and I stuck with Verizon (then NY Telephone?) and reduced my 
then current costs by 35%. The features of Centrex were not of much value 
to my employees (as they don't think outside of the box), but the cost savings 
were of instant value.

Some years later, I installed an inexpensive PBX which gave me voicemail 
and better internal call handling. The cost savings from that came a bit 
later. Utility that I discovered, was the ability to make 1 of my 3 
Centrex lines (the last line in the ring pattern) an unlimited outgoing 
line. Having done that, I made that the line that was picked up when 
somebody lifted a receiver to make an outgoing call. Our primary number 
was still advertised for incoming calls. That tactic reduced my local, 
long distance and regional calling costs by approximately $30 per month.

So, what's the advantage of Centrex for a 2 line business? The 
package of call forwarding, call waiting, etc. and perhaps some cost. They 
can transfer between the phones, however, I assume if it's a 2 line business, 
they are too small for that to be of much value.

The utility provided by VOIP PBX is much greater (if they can take 
advantage). Voicemail emailed, transfer to outside phones, ringblasts, 
ext.

Now for the cost implications: My current 4 lines cost me 
$160/mo. Going to VOIP hosted PBX, I will pay approx. $50/mo each for 
unlimited outgoing lines. (That get's me 3 for the same price, however, I 
will still have to maintain a dedicated fax line which will be POTS for the 
moment). What I can't get my head around, is how to increase my total 
quantity of VOIP extensions for the same price as I'm currently paying. 
It's not that I use more than 3 phones at 1 time, it's that I would like to be 
able to pick one up on the front counter as a convenience, or if my installer 
has calls to make, he can make them from his desk, rather than passing a phone 
back and forth.

It appears that your client, may be in a similar quandry without 
understanding the value of the technology. Remember, for a geek it's the 
coolness of the stuff, however for the rest of the world, it's about what will 
it do for me. Not that you can transfer a call to a cell phone and then 
back to the office, but that you can be more accessible to your client which 
will make you more important to them.

Now, if any of you experts have thoughts on how I can make this work better 
for me, I'm all ears. At the moment I have Packet 8 (on a trial basis) but 
am heading towards Aptela because it appears that their system is more flexible 
and business oriented. The one thing I need to get away from is 
maintaining a system myself, which is why I haven't put in a server with 
Asterisk on it. 

Rick

Rick 
SmithJDR Windows IncProviding quality window treatments throughout North 
America914-666-5777 x.13914-666-5796 (fax)

In a message dated 4/8/2006 3:57:43 A.M. Eastern Standard Time, 
[EMAIL PROTECTED] writes:
Message: 
  7Date: Fri, 07 Apr 2006 23:04:09 -0400From: Brian Capouch 
  [EMAIL PROTECTED]Subject: [Asterisk-Users] [OT] Centrex 
  QuestionTo: Asterisk Users Mailing List - Non-Commercial 
  Discussion  
  asterisk-users@lists.digium.comMessage-ID: 
  [EMAIL PROTECTED]Content-Type: text/plain; 
  charset=us-ascii; format=flowedI haven't dealt with Centrex for a long 
  time, and one of my customers is being courted heavily by a Sprint 
  salesperson.Am I not correct in assuming that each "line" of Centrex 
  corresponds to an "extension" in the PBX world?This site has 2 
  POTS lines and 5 extensions, and they told me that for the same thing 
  they're paying right now (~$40/POTS line) they will be getting two Centrex 
  "lines" 

Re: [Asterisk-Users] Call/Contact Center.

[Krzysztof Drewicz]

Erick Perez napisał(a):

What about the Digium TDM2460E (24 analog ports) instead of using a
channel bank?
and the aastra 480e (analog phones)

On the other hand, you ca go voip with he SPA-941 (two line) or the
SPA-942 (four line) and save the channel bank, or the TDM2460E card.


Channel bank gives live upgrade possibility. I could bring another one 
channel bank, do a ztcfg, do asterisk reload and having total downtime 
less than 1 minute, which is less then required level.


While TDM2460E stops at 4 (guessing) cards in one PC box. (and 5th card 
for Telco connection, a card with at less 24*4=96 public access channels).
So i think that TDM is not very good option. While SIP ATA gates give me 
  almost as low as 0 s down time, they require more work on asterisk 
Box (cpu, net).


maybe someone could say which channel bank to choose:

a) Zhone-cheepest
b) Adtran 750 discontinued
c) Rhino, good but expensive.

Anyone wiling to share his experience with above hardware?

kd,

--
Krzysztof Drewicz
Affordable 2/4 span E1/T1 PCI-cards. 100% Asterisk compatible.
See http://4e1.pl

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RE: [Asterisk-Users] OT: HOWTO: Create a 90mbit bonded link 600 m etre s away with Cat 3 or telco wire [long]

[Colin Anderson]

I have to ask, what was wrong with a pair of media converters ($200/pair
new, 
$50/pair on ebay) and some cheap-as-dirt multimode fiber?  Isolated,
100mbit 
and easily, easily gangable. Was the goal simply to get as fast as possible

with regular copper wire, or was there a bigger objective?

I do appreciate the effort put into this, though, and more than anything I 
appreciate your posting it here for others.  I sincerely thank you for
that.

Because of rights-of-way issues that happened after the copper was laid, it
was impossible to lay new fiber. The objective was to create a close to
100mbit as possible, redundant, and prioritizable (sp?) link using the
existing copper. 

The link just functions as a plain jane bridge right now but sticking Linux
in the mix allows for future QoS and routing enhancements when the remote
location grows. This is a large consideration because the company I work for
is experiencing 100% yearly growth with no end in sight. 

Thanks for the comments. It was a cool learning experience for me; I'm
actually suprised and impressed that it works so well. Ping times are
precisely the same as the local switch! When the users moved over, they just
plugged in. 

Yay Linux
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RE: [Asterisk-Users] DIALSTATUS for Multiple Dialled Numbers

[Alexander Lopez]

 Sorry, It was late and I forgot about that SMALL detail!!!

Thanks for the clarification. :-) 


Alex

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Moises Silva
 Sent: Saturday, April 08, 2006 10:22 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] DIALSTATUS for Multiple Dialled Numbers
 
 When you pass several Dial strings only the last exited 
 channel DIALSTATUS is saved. In the case that 1 of the 
 channels answer, the status will be ANSWER obviously, but if 
 the second fails because of CONGESTION and the first because 
 NOANSWER, the last exited channel dial status will be set.
 
 Regards
 
 On 4/7/06, Alexander Lopez [EMAIL PROTECTED] wrote:
  Without modifications to Dial, I don't think so.
 
  However,
 
  Dial(Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED])
 
  [dialstatus]
  _X.,1,Set(TECH=${CUT(${EXTEN},-,1)})
  _X.,2,Set(DEVICE=${CUT(${EXTEN},-,2)})
  _X.,3,Dial(${TECH}/${DEVICE}||)
 
 
  Or something like this...
 
  I would also create Variable name to track each one.
 
 
  -Original Message-
  From: [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED] On Behalf Of 
  Douglas Garstang
  Sent: Friday, April 07, 2006 2:21 PM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: [Asterisk-Users] DIALSTATUS for Multiple Dialled Numbers
  
  Folks,
  
  When I have a dial string like this:
  
  Dial(SIP/3254101SIP/3254102,20,tr)
  
  and I want to check the ${DIALSTATUS} variable after the 
 dial, how 
  do I know which number I am getting the variable for?
  
  And, what about this?
  
  Dial(SIP/3254101SIP/[EMAIL PROTECTED],20,tr)
  
  What happens in that case? How can I get the 
 ${DIALSTATUS} variable 
  for EACH NUMBER dialled?
  
  Thanks,
  Doug.
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 --
 Su nombre es GNU/Linux, no solamente Linux, mas info en 
 http://www.gnu.org;
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[Asterisk-Users] Call parking query

[Alex Brett]

Hi everybody,

I would like to set asterisk up such that to use the call parking 
feature, instead of transferring a call to the extension set up in 
features.conf, you just dial a code (e.g. *3) and this then parks the 
call. The main reason for this is that a number of the phones I use have 
transfer buttons that I can't reprogram to use Asterisk's own transfer 
functions, therefore you don't get the announcement of which extension 
they've been parked onto...


I assumed the way to do this would be the applicationmap in 
features.conf, so I tried various variations on this:


parkcall = 
*3,caller,ParkAndAnnounce,pbx-transfer:PARKED|60|Local/[EMAIL PROTECTED]|internal,${EXTEN},1


The most obvious problem I have had is that ${EXTEN} isn't decoded, I 
couldn't find much documentation on the applicationmap system, so I'm 
guessing there may be some other variable name that would do what I 
want, essentially it wants to be the number of the callee?


If I replaced ${EXTEN} with my extension for testing, it essentially 
worked, pushing *3 would hangup the current connection, and call me back 
then play the extension the user had been parked at, and I could pick 
the call back up etc. However, the timeout feature did not work 
properly, if the call timed out, then in the console I saw an error 
complaining about a default context not existing, and the extension that 
was parked was hung up - I don't know whether this is a problem with the 
ParkAndAnnounce command, or the applicationmap system...


If anybody has any suggestions, or has already implemented something 
similar to this and could tell me how they did it, I'd be very grateful!


Thanks,
Alex Brett
[EMAIL PROTECTED]
http://www.loho.co.uk/

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[Asterisk-Users] quadBRI PCI ISDN on Suse Linux 10

[Colin MacMillan]

Hello, 
I am trying to install the drivers for the Junghanns quadBRI card, but am having a problem.

I have essentially followed the procedure from http://www.voip-manager.net/installation-linux-asterisk.php.

At the stage to install the ISDN card I have completed the following:
1) downloaded the bristuff-0.2.0-RC8q.tar.gz to /usr/src
2) unpacked it in /usr/src - tar xvzf bristuff-0.2.0-RC8q.tar.gz
3) run 'make menuconfig' from /usr/src/linux-2.6.13-15 (my kernel
version) - this is recommended in the Junghanns INSTALL doc included in
bristuff.
4) created a link in /usr/src called linux-2.6 (ln -s /usr/src/linux-2.6.13-15 /usr/src/linux-2.6) as per INSTALL doc
5) run the install.sh script in the bristuff director.

At this stage during the install script, the quadBRI driver section shows the following:


rm -f qozap.o *.ko *.mod.c *.mod.o .*o.cmd *~
rm -rf .tmp_versions
make -C /usr/src/linux-2.6 SUBDIRS=/usr/src/bristuff-0.2.0-RC8q/qozap ZAP=-I/usr/src/bristuff-0.2.0-RC8q/zaptel-1.0.10 modules
make[1]: Entering directory `/usr/src/linux-2.6.13-15'
 CC [M] /usr/src/bristuff-0.2.0-RC8q/qozap/qozap.o
 Building modules, stage 2.
 MODPOST
*** Warning: zt_register [/usr/src/bristuff-0.2.0-RC8q/qozap/qozap.ko] undefined!
*** Warning: zt_receive [/usr/src/bristuff-0.2.0-RC8q/qozap/qozap.ko] undefined!
*** Warning: zt_transmit [/usr/src/bristuff-0.2.0-RC8q/qozap/qozap.ko] undefined!
*** Warning: zt_ec_chunk [/usr/src/bristuff-0.2.0-RC8q/qozap/qozap.ko] undefined!
*** Warning: zt_unregister [/usr/src/bristuff-0.2.0-RC8q/qozap/qozap.ko] undefined!
 CC /usr/src/bristuff-0.2.0-RC8q/qozap/qozap.mod.o
 LD [M] /usr/src/bristuff-0.2.0-RC8q/qozap/qozap.ko
make[1]: Leaving directory `/usr/src/linux-2.6.13-15'
install -D -m 644 qozap.ko /lib/modules/`uname -r`/misc/qozap.ko

quadBRI driver installed.
Press Enter to continue, or CTRL + C to abort.


6) From here I enter the qozap directory. cd qozap
7) now I get the following error -
linux:/usr/src/bristuff-0.2.0-RC8q/qozap # insmod qozap.ko
insmod: error inserting 'qozap.ko': -1 Invalid module format

Any help is greatly appreciated. I'm no expert so sorry if this posting is too 'noob' for some.

Possibly relevant is that my kernel is a 32-bit kernel and my processor is 64-bit.

Thank you,
Colin




Additional info:
linux:/usr/src/bristuff-0.2.0-RC8q/qozap # cat /proc/cpuinfo
processor : 0
vendor_id : GenuineIntel
cpu family : 15
model : 4
model name : Intel(R) Pentium(R) 4 CPU 2.80GHz
stepping : 9
cpu MHz : 2793.552
cache size : 1024 KB
physical id : 0
siblings : 2
core id : 0
cpu cores : 1
fdiv_bug : no
hlt_bug : no
f00f_bug : no
coma_bug : no
fpu : yes
fpu_exception : yes
cpuid level : 5
wp : yes
flags : fpu
vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36
clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe nx lm pni monitor
ds_cpl cid cx16 xtpr lahf_lm
bogomips : 5593.44

processor : 1
vendor_id : GenuineIntel
cpu family : 15
model : 4
model name : Intel(R) Pentium(R) 4 CPU 2.80GHz
stepping : 9
cpu MHz : 2793.552
cache size : 1024 KB
physical id : 0
siblings : 2
core id : 0
cpu cores : 1
fdiv_bug : no
hlt_bug : no
f00f_bug : no
coma_bug : no
fpu : yes
fpu_exception : yes
cpuid level : 5
wp : yes
flags : fpu
vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36
clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe nx lm pni monitor
ds_cpl cid cx16 xtpr lahf_lm
bogomips : 5586.58




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RE: [Asterisk-Users] queueue recording and what to do next

[Anton Krall]

Thank you very much for the tip Matt.

I was wondering, besides recording the queues, I also use mixmonitor on my
dialplans for some extensions, does mixmonitor also use sommix to mix the
call legs are is mixmonitor mixing realtime using inernal asterisk
functions?

I say this because I just to make sure that by replacing sommix I wont brake
anything else.
 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of Matt Roth
|Sent: Thursday, April 06, 2006 12:31 PM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] queueue recording and what to do next
|
|Anton Krall wrote:
|
|Guys, if you define recording on queues.conf and also define a 
|monitor_filename var on your dialplna, you can record a queue 
|call but, 
|isthere a way to do something with the file after the call 
|ends? I need 
|to move the file to some other place but I cant find where to 
|define a 
|command to run after a queue call finishes.
|
|Any hints?
|
|Anton,
|
|In queues.conf set:
|
|monitor-join=yes
|
|for all queues that you are recording.  This will cause soxmix 
|to be executed at the end of the call in order to join the leg 
|files into a single recording. 
|
|Then backup the soxmix binary and replace it with a script 
|that does whatever you want with the leg files.  Your script 
|will be passed 3 arguments (see show application monitor at 
|the CLI for more details):
|
|1) The -in leg filename.
|2) The -out leg filename.
|3) A target mixed filename.
|
|Odds are that you'll only care about the first two arguments.  
|Here is the script that I use to move the leg files from a RAM 
|disk over an NFS mount to a remote machine which handles 
|mixing and archiving the recordings:
|
|[EMAIL PROTECTED] ~]# cat /usr/bin/soxmix
|#!/bin/bash
|
|/bin/nice -n 19 mv --target-directory=/digrec-nfs/ $1 $2 if [ 
|$? -ne 0 ]; then
|echo Failed to mv '$3'  /var/log/asterisk/mvdr_log
|exit 1
|fi
|
|exit 0
|
|As you can see, I'm using the third argument to log any failed moves.  
|So far there haven't been any.
|
|There is another option besides replacing soxmix with a custom 
|program.  
|You can use the dialplan variables MONITOR_EXEC and 
|MONITOR_EXEC_ARGS to tell Monitor() to use another program 
|to mix the leg files.  I have found this method to be 
|unreliable.  Roughly 1% of the time, Monitor() was not calling 
|the program defined by MONITOR_EXEC.  Replacing soxmix has 
|worked for me 100% of the time (we handle 10,000 - 13,000 
|recordings a day), so I recommend it as the preferred solution.
|
|Matthew Roth
|InterMedia Marketing Solutions
|Software Engineer and Systems Developer
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|
|

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Re: [Asterisk-Users] queueue recording and what to do next

[Kevin P. Fleming]

Anton Krall wrote:

 I was wondering, besides recording the queues, I also use mixmonitor on my
 dialplans for some extensions, does mixmonitor also use sommix to mix the
 call legs are is mixmonitor mixing realtime using inernal asterisk
 functions?

MixMonitor mixes the audio internally.
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[Asterisk-Users] unable to enable stutter dialtone

[rkb]

I'm having problems enabling stutter dialtone for users connected to
channel banks.

Half of our users are on iaxy's and the other half are connecting to
channel banks.  The users on ixay's are getting the stutter dialtone
on new voicemails, but the ones on the channel banks are not. 
Currently, all users are in the default context in the voicemail.conf
file.  I've tried the following 3 methods for the entries in the
zapata.conf file, but none have worked.  The first method is what I
use in the iax.conf file and it's working.

mailbox=1234


[EMAIL PROTECTED]


channel = 49
[EMAIL PROTECTED]
callerid=”foo bar” 1234


Does anyone know how to fix this issue?

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[Asterisk-Users] G723 - ulaw codec problem

[Sam Tam]

I don't know whether it is just me but here is my setting.

I have here a voip softswitch  -  asterisk  -  VoIP provider.

I use G723 on softswitch, ulaw for asterisk and then ulaw again for VoIP
provider.

The reason why I don't use 723 on my voip provider is because it does not
support it and my softphone does not support ulaw. + my softswitch can't do
codec convert.

So here is my problem. I experience lag on some occasions when the calls get
pick up and some lags with lose of voice. Then it also have problem bridging
the switch and asterisk server when the call first get pick up.

I have used 729 and find those problems are gone. I am currently using the
free IPP g723 and 729 license and I can't see why it works for 729 and not
723.

If I can get someone to enlighten me then it would be great..

Sam



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Re: [Asterisk-Users] G723 - ulaw codec problem

[Kevin P. Fleming]

Sam Tam wrote:

 I have used 729 and find those problems are gone. I am currently using the
 free IPP g723 and 729 license and I can't see why it works for 729 and not
 723.

You have clearly identified that the problem is with the G.723 codec you
are using, so you should contact the provider of that module.
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[Asterisk-Users] Re: [asterisk-dev] bug or bad chan_sip.c

[hgaillac-sip]

Tzafrir,

How did you set  sip:[EMAIL PROTECTED]

I use serasterisk

look at my sip.conf and extensions.conf

Regards 
Harry 

[general]

context=sip
realm=nxs.yi.org
bindport=5050
bindaddr=nxs.yi.org
srvlookup=yes
tos=lowdelay
maxexpirey=3600
defaultexpirey=1000
allow=all
musicclass=default
language=fr
insecure=very
allowguest=yes

rtptimeout=60
rtpholdtimeout=300
useragent=PBX
dtmfmode = rfc2833
checkmwi=20

promiscredir=no
nat=yes

autodomain=no
domain=nxs.yi.org,sip
allowexternalinvites=yes
rtcachefriends=yes

rtupdate=yes
rtautoclear=yes
ignoreregexpire=yes

and extensions.conf
[general]

static=yes
writeprotect=no
autofallthrough=yes
//

[globals]


[mainmenu]
exten = s,1,Answer()
exten =
s,n,GotoIfTime(09:30-21:00|mon-sun|*|*?day,s,1)
exten =
s,n,GotoIfTime(21:01-09:31|mon-sun|*|*?night,s,1)


[night]
exten = s,1,PlayBack(closed)
exten = s,2,Voicemail(u84)
exten = s,3,Hangup


[day]
exten = s,1,BackGround(annoucement)
exten = s,2,WaitExten(10)

exten = t,1,Playback(no-answer)
exten = t,2,Hangup()

exten = *,1,PlayBack(waiting)
exten = *,2,Queue(info|t||)

exten = 1,1,BackGround(ipbx)
exten = 1,2,Goto(s,1)

exten = 2,1,Playback(informations)
exten = 2,2,Goto(music,600,1)

exten = i,1,PlayBack(key-invalide)
exten = i,2,Goto(s,1)



[sip]
include = info
include = support

exten = info,1,Answer()
exten = info,2,Dial(Sip/84,10)
exten = info,3,Dial(Sip/85,10)
exten = info,4,Hangup

exten = support,1,Answer()
exten = support,2,Queue(support|t||)
exten = support,3,Hangup




[pstn]

exten = s,1,Answer()
exten = s,2,NVFaxDetect()
exten = s,3,Goto(mainmenu,s,1)
exten = s,4,Hangup
exten = fax,1,Dial(Zap/g2)
exten = talk,1,Goto(mainmenu,s,1)

exten = t,1,Hangup()


include = outgoing-pstn


[info]

exten = 84,1,Answer()
exten = 84,2,Dial(Sip/84,30,t)
exten = 84,3,VoiceMail(u84)
exten = 84,103,VoiceMail(b84)

exten = 85,1,Answer()
exten = 85,2,Dial(Sip/85,30,t)
exten = 85,3,VoiceMail(u85)
exten = 85,103,VoiceMail(b85)


include = parkedcalls
include = guest
include = agents
include = pstn
include = music
include = mailbox
include = support
include = aliases

[support]


exten = 86,1,Answer()
exten = 86,2,Dial(Sip/86,30,t)
exten = 86,3,VoiceMail(u86)
exten = 86,103,VoiceMail(b86)

exten = 87,1,Answer()
exten = 87,2,Dial(Sip/87,30,t)
exten = 87,3,VoiceMail(u87)
exten = 87,103,VoiceMail(b87)


include = parkedcalls
include = guest
include = agents
include = pstn
include = music
include = mailbox
include = info


[guest]

exten = 88,1,Answer()
exten = 88,2,Dial(Sip/88,30,t)
exten = 88,3,VoiceMail(u88)
exten = 88,103,VoiceMail(b88)

include = music
include = mailbox

[fax]
exten = fax,1,Dial(Zap/2,40)
exten = fax,2,Congestion
exten = fax,102,Congestion

include = outgoing-pstn



[outgoing-pstn]

ingnorepat = 0
exten = _0,1,ChanIsAvail(Zap/g1, j)
exten = _0,2,Dial(Zap/g1/${EXTEN:1})
exten = _0,102,Playback(busy)
exten = _0,103,Hangup

exten = _0.,1,Dial(Zap/g1/${EXTEN:1})

[mailbox]
exten = 700,1,Answer()
exten = 700,2,VoiceMailMain()

[music]
exten = 600,1,Answer()
exten = 600,2,WaitMusicOnHold(60)
exten = 600,3,Hangup
exten = music,1,Goto(600,1)

[agents]
;Agent Login
exten=
501,1,AgentCallbackLogin(||[EMAIL PROTECTED])
exten=
502,1,AgentCallbackLogin(||[EMAIL PROTECTED])

;Agent Logout
exten= 503,1,AgentCallbackLogin(||l)
exten= 504,1,AgentCallbackLogin(||l)

[aliases]
exten = alice,1,Goto(info,84,1)
exten = bob,1,Goto(support,86,1)
//

--- Tzafrir Cohen [EMAIL PROTECTED] a écrit :

 On Sat, Apr 08, 2006 at 09:31:43PM +0200,
 [EMAIL PROTECTED] wrote:
  Hello,
  
  Anybody could explain me why asterisk spend time
 to
  send back to proxy or sip agent authentication
  messages 407
  
 
 I believe some people tried. At least when I
 happened to be present on
 #asterisk.
 
  nobody can call me from other domains.
 
 Tough. But this is not asterisk-users. Any relevant
 questions you have
 to the developers?
 
  can we disable authentication for none peers or
 users 
 
 You were told how to do something quite similar
 (allowguests). 
 Is that good enough? If not: why not? Still, a
 -users question.
 
  
  Asterisk ask authentication 407 for
  sip:[EMAIL PROTECTED] or sip:[EMAIL PROTECTED]
 
 And after people tell you that this is a matter of
 setting up sip.conf
 properly, you still only bother quoting yor
 dialplan, rather than
 sip.conf.
 
 You also post several thread rather than keeping
 everything in one
 thread. You also post to multiple lists and use
 subject lines suuch as
 HELP !.
 
 Please read a bit about nettique. Your behaviour
 does does not encourge
 people to help you.
 
 -- 
 Tzafrir Cohen  sip:[EMAIL PROTECTED]
 icq#16849755   iax:[EMAIL PROTECTED]
 +972-50-7952406   
 [EMAIL PROTECTED]  http://www.xorcom.com
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[Asterisk-Users] Re: [asterisk-dev] bug or bad chan_sip.c

[hgaillac-sip]

Tzafrir,

How did you set  sip:[EMAIL PROTECTED]

I use serasterisk

look at my sip.conf and extensions.conf

Regards 
Harry 

[general]

context=sip
realm=nxs.yi.org
bindport=5050
bindaddr=nxs.yi.org
srvlookup=yes
tos=lowdelay
maxexpirey=3600
defaultexpirey=1000
allow=all
musicclass=default
language=fr
insecure=very
allowguest=yes

rtptimeout=60
rtpholdtimeout=300
useragent=PBX
dtmfmode = rfc2833
checkmwi=20

promiscredir=no
nat=yes

autodomain=no
domain=nxs.yi.org,sip
allowexternalinvites=yes
rtcachefriends=yes

rtupdate=yes
rtautoclear=yes
ignoreregexpire=yes

and extensions.conf
[general]

static=yes
writeprotect=no
autofallthrough=yes
//

[globals]


[mainmenu]
exten = s,1,Answer()
exten =
s,n,GotoIfTime(09:30-21:00|mon-sun|*|*?day,s,1)
exten =
s,n,GotoIfTime(21:01-09:31|mon-sun|*|*?night,s,1)


[night]
exten = s,1,PlayBack(closed)
exten = s,2,Voicemail(u84)
exten = s,3,Hangup


[day]
exten = s,1,BackGround(annoucement)
exten = s,2,WaitExten(10)

exten = t,1,Playback(no-answer)
exten = t,2,Hangup()

exten = *,1,PlayBack(waiting)
exten = *,2,Queue(info|t||)

exten = 1,1,BackGround(ipbx)
exten = 1,2,Goto(s,1)

exten = 2,1,Playback(informations)
exten = 2,2,Goto(music,600,1)

exten = i,1,PlayBack(key-invalide)
exten = i,2,Goto(s,1)



[sip]
include = info
include = support

exten = info,1,Answer()
exten = info,2,Dial(Sip/84,10)
exten = info,3,Dial(Sip/85,10)
exten = info,4,Hangup

exten = support,1,Answer()
exten = support,2,Queue(support|t||)
exten = support,3,Hangup




[pstn]

exten = s,1,Answer()
exten = s,2,NVFaxDetect()
exten = s,3,Goto(mainmenu,s,1)
exten = s,4,Hangup
exten = fax,1,Dial(Zap/g2)
exten = talk,1,Goto(mainmenu,s,1)

exten = t,1,Hangup()


include = outgoing-pstn


[info]

exten = 84,1,Answer()
exten = 84,2,Dial(Sip/84,30,t)
exten = 84,3,VoiceMail(u84)
exten = 84,103,VoiceMail(b84)

exten = 85,1,Answer()
exten = 85,2,Dial(Sip/85,30,t)
exten = 85,3,VoiceMail(u85)
exten = 85,103,VoiceMail(b85)


include = parkedcalls
include = guest
include = agents
include = pstn
include = music
include = mailbox
include = support
include = aliases

[support]


exten = 86,1,Answer()
exten = 86,2,Dial(Sip/86,30,t)
exten = 86,3,VoiceMail(u86)
exten = 86,103,VoiceMail(b86)

exten = 87,1,Answer()
exten = 87,2,Dial(Sip/87,30,t)
exten = 87,3,VoiceMail(u87)
exten = 87,103,VoiceMail(b87)


include = parkedcalls
include = guest
include = agents
include = pstn
include = music
include = mailbox
include = info


[guest]

exten = 88,1,Answer()
exten = 88,2,Dial(Sip/88,30,t)
exten = 88,3,VoiceMail(u88)
exten = 88,103,VoiceMail(b88)

include = music
include = mailbox

[fax]
exten = fax,1,Dial(Zap/2,40)
exten = fax,2,Congestion
exten = fax,102,Congestion

include = outgoing-pstn



[outgoing-pstn]

ingnorepat = 0
exten = _0,1,ChanIsAvail(Zap/g1, j)
exten = _0,2,Dial(Zap/g1/${EXTEN:1})
exten = _0,102,Playback(busy)
exten = _0,103,Hangup

exten = _0.,1,Dial(Zap/g1/${EXTEN:1})

[mailbox]
exten = 700,1,Answer()
exten = 700,2,VoiceMailMain()

[music]
exten = 600,1,Answer()
exten = 600,2,WaitMusicOnHold(60)
exten = 600,3,Hangup
exten = music,1,Goto(600,1)

[agents]
;Agent Login
exten=
501,1,AgentCallbackLogin(||[EMAIL PROTECTED])
exten=
502,1,AgentCallbackLogin(||[EMAIL PROTECTED])

;Agent Logout
exten= 503,1,AgentCallbackLogin(||l)
exten= 504,1,AgentCallbackLogin(||l)

[aliases]
exten = alice,1,Goto(info,84,1)
exten = bob,1,Goto(support,86,1)
//

--- Tzafrir Cohen [EMAIL PROTECTED] a écrit :

 On Sat, Apr 08, 2006 at 09:31:43PM +0200,
 [EMAIL PROTECTED] wrote:
  Hello,
  
  Anybody could explain me why asterisk spend time
 to
  send back to proxy or sip agent authentication
  messages 407
  
 
 I believe some people tried. At least when I
 happened to be present on
 #asterisk.
 
  nobody can call me from other domains.
 
 Tough. But this is not asterisk-users. Any relevant
 questions you have
 to the developers?
 
  can we disable authentication for none peers or
 users 
 
 You were told how to do something quite similar
 (allowguests). 
 Is that good enough? If not: why not? Still, a
 -users question.
 
  
  Asterisk ask authentication 407 for
  sip:[EMAIL PROTECTED] or sip:[EMAIL PROTECTED]
 
 And after people tell you that this is a matter of
 setting up sip.conf
 properly, you still only bother quoting yor
 dialplan, rather than
 sip.conf.
 
 You also post several thread rather than keeping
 everything in one
 thread. You also post to multiple lists and use
 subject lines suuch as
 HELP !.
 
 Please read a bit about nettique. Your behaviour
 does does not encourge
 people to help you.
 
 -- 
 Tzafrir Cohen  sip:[EMAIL PROTECTED]
 icq#16849755   iax:[EMAIL PROTECTED]
 +972-50-7952406   
 [EMAIL PROTECTED]  http://www.xorcom.com
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Re: [Asterisk-Users] queueue recording and what to do next

[Steve Totaro]

I also use an FTP perl script that runs via cron job every five minutes 
to move the recordings to a NAS device and delete them from the * box.


Anton Krall wrote:

Thank you very much for the tip Matt.

I was wondering, besides recording the queues, I also use mixmonitor on my
dialplans for some extensions, does mixmonitor also use sommix to mix the
call legs are is mixmonitor mixing realtime using inernal asterisk
functions?

I say this because I just to make sure that by replacing sommix I wont brake
anything else.
 


|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of Matt Roth

|Sent: Thursday, April 06, 2006 12:31 PM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] queueue recording and what to do next
|
|Anton Krall wrote:
|
|Guys, if you define recording on queues.conf and also define a 
|monitor_filename var on your dialplna, you can record a queue 
|call but, 
|isthere a way to do something with the file after the call 
|ends? I need 
|to move the file to some other place but I cant find where to 
|define a 
|command to run after a queue call finishes.

|
|Any hints?
|
|Anton,
|
|In queues.conf set:
|
|monitor-join=yes
|
|for all queues that you are recording.  This will cause soxmix 
|to be executed at the end of the call in order to join the leg 
|files into a single recording. 
|
|Then backup the soxmix binary and replace it with a script 
|that does whatever you want with the leg files.  Your script 
|will be passed 3 arguments (see show application monitor at 
|the CLI for more details):

|
|1) The -in leg filename.
|2) The -out leg filename.
|3) A target mixed filename.
|
|Odds are that you'll only care about the first two arguments.  
|Here is the script that I use to move the leg files from a RAM 
|disk over an NFS mount to a remote machine which handles 
|mixing and archiving the recordings:

|
|[EMAIL PROTECTED] ~]# cat /usr/bin/soxmix
|#!/bin/bash
|
|/bin/nice -n 19 mv --target-directory=/digrec-nfs/ $1 $2 if [ 
|$? -ne 0 ]; then

|echo Failed to mv '$3'  /var/log/asterisk/mvdr_log
|exit 1
|fi
|
|exit 0
|
|As you can see, I'm using the third argument to log any failed moves.  
|So far there haven't been any.

|
|There is another option besides replacing soxmix with a custom 
|program.  
|You can use the dialplan variables MONITOR_EXEC and 
|MONITOR_EXEC_ARGS to tell Monitor() to use another program 
|to mix the leg files.  I have found this method to be 
|unreliable.  Roughly 1% of the time, Monitor() was not calling 
|the program defined by MONITOR_EXEC.  Replacing soxmix has 
|worked for me 100% of the time (we handle 10,000 - 13,000 
|recordings a day), so I recommend it as the preferred solution.

|
|Matthew Roth
|InterMedia Marketing Solutions
|Software Engineer and Systems Developer
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|
|

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Re: [Asterisk-Users] unable to enable stutter dialtone

[Eric \ManxPower\ Wieling]

[EMAIL PROTECTED] wrote:

I'm having problems enabling stutter dialtone for users connected to
channel banks.

Half of our users are on iaxy's and the other half are connecting to
channel banks.  The users on ixay's are getting the stutter dialtone
on new voicemails, but the ones on the channel banks are not. 
Currently, all users are in the default context in the voicemail.conf

file.  I've tried the following 3 methods for the entries in the
zapata.conf file, but none have worked.  The first method is what I
use in the iax.conf file and it's working.

mailbox=1234


[EMAIL PROTECTED]


channel = 49
[EMAIL PROTECTED]
callerid=”foo bar” 1234


You set the options BEFORE the channel number.

[EMAIL PROTECTED]
callerid=foo bar 1234
channel = 49
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[Asterisk-Users] ANI on a PRI

[Steve Totaro]

Is there a setting somewhere in * to define whether I am receiving 
callerID or true ANI?  Global Crossing claims they are sending ANI but I 
dont think so.  My understanding of ANI is that it is always sent, 
regardless if callerID is blocked.  If I dial *67 and my DID, I get 
Presentation: Presentation prohibited of network provided number and 
no number.


Before I call GC on Monday to complain, I want to make sure I am correct 
in my understanding of ANI and not missing something on my side. 


 Protocol Discriminator: Q.931 (8)  len=33
 Call Ref: len= 2 (reference 16/0x10) (Originator)
 Message type: SETUP (5)
 [04 03 80 90 a2]
 Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer 
capability: Speech (0)
  Ext: 1  Trans mode/rate: 64kbps, 
circuit-mode (16)

  Ext: 1  User information layer 1: u-Law (34)
 [18 04 e1 81 83 81]
 Channel ID (len= 6) [ Ext: 1  IntID: Explicit, PRI Spare: 0, Preferred 
Dchan: 0

ChanSel: Reserved
   Ext: 1  DS1 Identifier: 1
   Ext: 1  Coding: 0   Number Specified   Channel 
Type: 3

   Ext: 1  Channel: 1 ]
 [6c 02 21 a3]
 Calling Number (len= 4) [ Ext: 0  TON: National Number (2)  NPI: 
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
   Presentation: Presentation prohibited of 
network provided number (35) '' ]

 [70 0b a1 38 30 30 35 36 34 30 38 31 39]
 Called Number (len=13) [ Ext: 1  TON: National Number (2)  NPI: 
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '8005640819' ]

-- Making new call for cr 16
-- Processing Q.931 Call Setup
-- Processing IE 4 (cs0, Bearer Capability)
-- Processing IE 24 (cs0, Channel Identification)
-- Processing IE 108 (cs0, Calling Party Number)
-- Processing IE 112 (cs0, Called Party Number)
 Protocol Discriminator: Q.931 (8)  len=11
 Call Ref: len= 2 (reference 16/0x10) (Terminator)
 Message type: CALL PROCEEDING (2)
 [18 04 e9 81 83 81]
 Channel ID (len= 6) [ Ext: 1  IntID: Explicit, PRI Spare: 0, 
Exclusive Dchan: 0

ChanSel: Reserved
   Ext: 1  DS1 Identifier: 1
   Ext: 1  Coding: 0   Number Specified   Channel 
Type: 3

   Ext: 1  Channel: 1 ]
 Protocol Discriminator: Q.931 (8)  len=15
 Call Ref: len= 2 (reference 16/0x10) (Terminator)
 Message type: CONNECT (7)
 [18 04 e9 81 83 81]
 Channel ID (len= 6) [ Ext: 1  IntID: Explicit, PRI Spare: 0, 
Exclusive Dchan: 0

ChanSel: Reserved
   Ext: 1  DS1 Identifier: 1
   Ext: 1  Coding: 0   Number Specified   Channel 
Type: 3

   Ext: 1  Channel: 1 ]
 [1e 02 81 82]
 Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard 
(0) 0: 0   Location: Private network serving the local user (1)
   Ext: 1  Progress Description: Called 
equipment is non-ISDN. (2) ]

 Protocol Discriminator: Q.931 (8)  len=5
 Call Ref: len= 2 (reference 16/0x10) (Originator)
 Message type: CONNECT ACKNOWLEDGE (15)
 Protocol Discriminator: Q.931 (8)  len=9
 Call Ref: len= 2 (reference 16/0x10) (Originator)
 Message type: DISCONNECT (69)
 [08 02 80 90]
 Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0: 0   
Location: User (0)

  Ext: 1  Cause: Unknown (16), class = Normal Event (1) ]
-- Processing IE 8 (cs0, Cause)
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication, 
peerstate Disconnect Request

 Protocol Discriminator: Q.931 (8)  len=9
 Call Ref: len= 2 (reference 16/0x10) (Terminator)
 Message type: RELEASE (77)
 [08 02 81 90]
 Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0: 0   
Location: Private network serving the local user (1)

  Ext: 1  Cause: Unknown (16), class = Normal Event (1) ]
 Protocol Discriminator: Q.931 (8)  len=5
 Call Ref: len= 2 (reference 16/0x10) (Originator)
 Message type: RELEASE COMPLETE (90)
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null
NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null

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Re: [Asterisk-Users] Voicemail 0 for operator call routing

[broadbandvoice]

Need to add context in the exten files, to differentiate between company A and company B

-- Original message -- From: [EMAIL PROTECTED]  It's the 'o' extension in your context that hits the voicemail.  (thats a lower case o not a zero)   PaulH   - Original Message -  From: "Paul Tinsley" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion"   Sent: Wednesday, February 22, 2006 3:19 AM  Subject: [Asterisk-Users] Voicemail 0 for operator call routing Does anyone know of a way to specify what extension is dialed when 0 is   pressed in the voicemail system. I have a situation where there is more   than one secretary and they want the 0 to redirect to the appropriate   secretary for the two groups of people. 
 t;So an example would be:   555-1234 - voicemail - Secretary 1   555-1235 - voicemail - Secretary 2 Any help would be greatly appreciated.   ___   --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list   To UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users ___  --Bandwidth and Colocation provided by Easynews.com --   Asterisk-Users mailing list  To UNSUBSCRIBE or update options visit:  http://lists.digium.com/mailman/listinfo/asterisk-users 
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[Asterisk-Users] Force codec

[Michael Strelnikov]

Hi, Is it possible to force using codec depends on extension? For example, voice codec is ILBC and with some prefix fax code should be ulaw.Thanks.Best regards,Michael
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Re: [Asterisk-Users] Force codec

[Eric \ManxPower\ Wieling]

Michael Strelnikov wrote:

Hi,

   Is it possible to force using codec depends on extension? For example,
voice codec is ILBC and with some prefix fax code should be ulaw.


[EMAIL PROTECTED] asterisk]# grep CODEC asterisk-1.2/doc/*
asterisk-1.2/doc/README.variables:${SIP_CODEC} Set the SIP codec for a call
[EMAIL PROTECTED] asterisk]#
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Re: [Asterisk-Users] Force codec

[Steve Totaro]

Eric ManxPower Wieling wrote:

Michael Strelnikov wrote:

Hi,

   Is it possible to force using codec depends on extension? For 
example,

voice codec is ILBC and with some prefix fax code should be ulaw.


[EMAIL PROTECTED] asterisk]# grep CODEC asterisk-1.2/doc/*
asterisk-1.2/doc/README.variables:${SIP_CODEC} Set the SIP codec for a 
call

[EMAIL PROTECTED] asterisk]#
___


Great replies Manx.  Not only do we get our answers, we are being 
enabled to answer them ourselves in the future.  I like your style.


Steve Totaro
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Re: [Asterisk-Users] Force codec

[Brian Capouch]

Steve Totaro wrote:

Eric ManxPower Wieling wrote:


Michael Strelnikov wrote:


Hi,

   Is it possible to force using codec depends on extension? For 
example,

voice codec is ILBC and with some prefix fax code should be ulaw.



[EMAIL PROTECTED] asterisk]# grep CODEC asterisk-1.2/doc/*
asterisk-1.2/doc/README.variables:${SIP_CODEC} Set the SIP codec for a 
call

[EMAIL PROTECTED] asterisk]#
___



Great replies Manx.  Not only do we get our answers, we are being 
enabled to answer them ourselves in the future.  I like your style.




Well that trick works sometimes, and sometimes it don't.

I've grepped myself half-blind in the same file areas, except that I'm 
wondering whether the same can be done with a pair of IAX endpoints.


A couple of mails to a person with a pretty good working knowledge of 
the codebase got me the answer, Well that's kind of a grey area. . 


:-)

B.
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Re: [Asterisk-Users] Force codec

[Michael Strelnikov]

Sorry for not mentioning that I'm using IAX.On 4/9/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Michael Strelnikov wrote: Hi,Is it possible to force using codec depends on extension? For example,
 voice codec is ILBC and with some prefix fax code should be ulaw.[EMAIL PROTECTED] asterisk]# grep CODEC asterisk-1.2/doc/*asterisk-1.2/doc/README.variables:${SIP_CODEC} Set the SIP codec for a call[EMAIL PROTECTED]
 asterisk]#___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:
 http://lists.digium.com/mailman/listinfo/asterisk-users-- Best regards,Michael Strelnikov
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[Asterisk-Users] How to set busy

[Miles Scruggs]

For multiline phones how do you set SIP channels to busy.  For instance 
if SIP/101 is on a call then dial would return busy.  Right now it just 
starts ringing on line X, and stacks up from there.


What would be really great is if I could control how many calls by the 
context.  So if a call was routed via


[overload]  Then the ext wouldn't report busy it would just keep ringing 
available lines, but if the call was routed via


[singletrackmind] dial would return busy if the channel already had one 
call.


Thanks

Miles
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Re: [Asterisk-Users] Problems with registering iaxy

[Bartosz Wegrzyn - asterisk]

Anyone knows hot to fix that?

Thanks

 I used to have my iaxy registered to my old version of asterisk.
 I switched to 1.2 ver and now registration fails.

 my config for iax.conf for that client looks like this:

 [user]
 username=user
 type=friend
 context=sip
 auth=plaintext
 secret=password
 host=dynamic
 disallow=all
 allow=ulaw
 trunk=no

 I provisioned my iax with this config:
 [EMAIL PROTECTED] iaxyprov]# cat  iaxy
 ;
 ; IAXY Provisioning description
 ;
 ;dhcp
 ip: 192.168.1.249
 netmask: 255.255.255.0
 gateway: 192.168.1.251
 codec: ulaw
 ;codec: adpcm
 server: 192.168.1.251
 ;altserver: 192.168.0.2
 user: user
 pass: password


 When I do iax2 debug

 I see this:

 IAX2 Debugging Enabled
 Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass:
 REGREQ
Timestamp: 2ms  SCall: 12640  DCall: 0 [192.168.1.249:4569]
USERNAME: user
REFRESH : 60
DEVICE TYPE : iaxy2
SERVICE IDENT   : 0003640005a8
PROVISIONG VER  : 3503263220
 voip*CLI
 Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
 REGAUTH
Timestamp: 00012ms  SCall: 00011  DCall: 12640 [192.168.1.249:4569]
AUTHMETHODS : 1
USERNAME: user
 voip*CLI
 Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass:
 INVAL
Timestamp: 0ms  SCall: 12640  DCall: 00011 [192.168.1.249:4569]
 Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass:
 REGREQ
Timestamp: 2ms  SCall: 08797  DCall: 0 [192.168.1.249:4569]
USERNAME: user
REFRESH : 60
DEVICE TYPE : iaxy2
SERVICE IDENT   : 0003640005a8
PROVISIONG VER  : 3503263220


 Any ideas what is wrong?
 Does new asterisk differs in the iax2 registration?

 Thanks

 Bart


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[Asterisk-Users] MACRO_RESULT=ABORT

[Shaun]

I have a macro that runs off a dial() and gives the callee a bunch of 
options... one of them is to disconnect the caller.  I read that setting 
MACRO_RESULT=ABORT would hang up both legs of the call.  When i set 
MACRO_RESULT=ABORT and return to the context it ends up sending the caller 
to voicemail (the next line in the dialplan/context is voicemail() ).  I 
need it to hangup the call... where am i going wrong?


-- 

~Shaun 



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