Re: [asterisk-users] Dundi and Dial Arguments

[Michiel van Baak]

On 16:38, Tue 01 Aug 06, Douglas Garstang wrote:
  I suggest you use an AGI for it.
  That gives you way more options
 
 How does AGI help? Your still calling DUNDILOOKUP inside the AGI script, and 
 not matter how many times you call it, your still always going to get the 
 lowest priority path returned.

Yeah, you're right. My bad.
-- 
Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD

Why is it drug addicts and computer afficionados are both called users?

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SV: [asterisk-users] VOIP phone for Receptionist use

[Jon Schøpzinsky]

For the VoIP phone question, I can warmly
recommend the Snom 360.

When using hints in asterisk, this is the
perfect phone for secretary use, as you can also add a side panel with 48 extra
buttons with lights.



When using hints you can see when
extensions a talking, ringing, as well as have up to 12 ingoing lines.

A very good phone, that we recommend to
all of our large customers, with a secretary.



Regards

Jon











Fra:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] På vegne af Jeff Busch
Sendt: 2. august 2006 02:20
Til:
asterisk-users@lists.digium.com
Emne: [asterisk-users] VOIP phone
for Receptionist use







I've searched through the newsgroup and online and haven't
found an answer for my question... maybe I am looking for the wrong terms, I am
not sure...











I have a client that would like a phone that is like a
typical receptionists phone.











Requirements:





- Ability for their3 lines to light-up a
button on the phone when one of them rings in.





- Ability for the phone to ring when the receptionist is on
one call and a second or third call is incoming. (this has been the
biggest frustration up to now. When a second call comes, there is no tone
that heard on the IP500. Perhaps I am missing a setting?)











We are currently using:











Asterisk @ Home 2.1





Polycom IP500/501 phones











Is there a way to do what we need to using the IP500
phones? If so, can anyone give me instructions on how to make it work
with [EMAIL PROTECTED]?











Thanks for your help in advance.











Jeff










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[asterisk-users] Problem with Cisco7970 SIP load / call transfer

[Juha Suhonen]

Hi!


I'm having an interesting problem with Cisco7970 SIP load (8.0(2)SR1) - 
the phone seems to work otherwise fine, but I can't do an assisted 
transfer (and the 7970 phone also doesn't seem to support the BlindXFer 
option that previous models have had). Phones are connected to Asterisk 
1.2.10.



What happens is this: User a calls to my phone. I press Transfer on the 
phone, I then place another call to another extension. When this new call 
is connected, pressing the Transfer -button again sends 2 SIP INVITE 
messages (and asterisk acks them with seemingly appropriate OK 
messages). But.. After getting the acks, phone just says Unable to 
complete transfer and both current calls are placed on hold.


Has anybody else seen this? Any ideas on how to fix? The same 
configuration works with Cisco 7960 (using some pretty ancient SIP load). 
I've also thought about upgrading the phone to 8.0(3) release of the SIP 
load, but atleast voip-info.org wiki states it as a total disaster - can 
anybody confirm if it's really a disaster?



As a related note, I'm also not seeing MWI with the 7970 phone - when 
Asterisk sends the MWI status message to phone, Asterisk immediatetly 
barfs out -- Got SIP response 400 Bad Request back from xxx. Does 
anybody know if this is a bug on the phone and maybe fixed on a later 
image? (and is there any workaround I can enable on asterisk to overcome 
this)



Also, a small UI thing - has anybody found a way to get the # -key to 
directly dial the number which has been inputted and mimic the behaviour 
7960s had? Our users are accustomed to keying in 123# instead of pressing 
123 + dial..





-- juhas
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[asterisk-users] Asterisk config with Analouge Audio Codec model number MP108FXS

[Mr shobhit nirala]

Hi FriendsI want to configure my asterisk server with audio codec model MP-108FXS   my sip.conf has the user name with mohit i want to configure this withanalouge  SHOBHIT NIRALA CONT NO. 9871476403 
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RE : Re: [asterisk-users] SRTP

[harrygaillac-sip]

--- Kai Ober [EMAIL PROTECTED] a écrit :

 Khaled Chehab schrieb:
  Is SRTP available in asterisk?  Or how to
 implement it ? am using trixbox
 

 you asked this question before, and you got answers,
 read your mail, or 
 stay away from this list!
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http://www.e164.org/wiki/AsteriskSRTP

Harry







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[asterisk-users] Slow dialing from PBX via E1

[Gavin Hamill]

Hi :)

I have a 'slow dialing' problem. When I dial 200# for the 
'echo test' application from my PBX extension 1010, I see this
in the console the instant I press the # key:
 
  -- Starting simple switch on 'Zap/65-1'
  -- Accepting overlap call from '1010' to '200' on channel 0/3, span 3
 
so Asterisk has accepted the call setup from the PBX. Then exactly 3
seconds elapses, and finally:
 
  -- Executing Playback(Zap/65-1, demo-echotest) in new stack
  -- Playing 'demo-echotest' (language 'en')
 
at which point Allison announces 'You are about to enter an echo test..'
 
How can I remove this 3 second pause? It's really annoying, and 
it doesn't happen when I dial out from the legacy PBX via an ISDN30 bearer
not connected to Asterisk (nor does it happen with SIP phones on Asterisk).

Even with debug + verbose both at 99, I see no extra information 
 
The extensions.conf is trivial:

[general]
static=yes
writeprotect=yes

[fromaxxess]
exten = 200,1,Playback(demo-echotest)  ; Let them know what's going on
exten = 200,2,Echo ; Do the echo test
exten = 200,3,Playback(demo-echodone)  ; Let them know it's over

This is with Asterisk 1.2.4 and Zaptel 1.2.3, on a Sangoma A104u (Sangoma 
support say their driver does no buffering and can't understand why this
is happening)

As ever, any advice warmly welcomed :)

Cheers,
Gavin.
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SV: [asterisk-users] Help debugging strange asterisk behaviour

[jan.sarin]

I think I'm using native since I don't recall installing anything else (except 
lame codec). How do I check which I am using? I'm unfortunately no asterisk 
expert that's why I need your help! ;)

My musiconhold.conf (I have no musiconhold_additional.conf):
;
; Music on hold class definitions
; This is using the new 1.2 config file format, and will not work with 1.0
; based Asterisk systems
;
[default]
mode=files
directory=/var/lib/asterisk/mohmp3
#include musiconhold_additional.conf

Thanks very much for your time!

Regards,
Jan

-Ursprungligt meddelande-
Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För Mojo with Horan  
Company, LLC
Skickat: den 1 augusti 2006 23:20
Till: Asterisk Users Mailing List - Non-Commercial Discussion
Ämne: Re: [asterisk-users] Help debugging strange asterisk behaviour

Are you using mpg123 for MoH or native?  What's in your musiconhold.conf?



[EMAIL PROTECTED] wrote:
 Hi,
 
 I'm one of those types who want to know what the heck is wrong when 
 something is wrong.
 
 I just installed a new server (see config below) and it all works fine 
 for a few hours. But after 3-5 hours asterisk starts behaving VERY 
 strangely for no apparent reason...
 
 1) MoH stops playing
 2) Some calls are not hung up from Zap-side
 3) Flash Operator Panel starts showing all kind of random letters.
 4) Agents are unable to login/logout.
 
 ..and so on. But the strange thing is that some things seem to work 
 perfectly fine as usual. Inbound calls are getting playbacks() but no 
 MoH when sent to queue, and caller is not sent to an agent. Outgoing 
 sip and zap calls work fine (until all zapchans are filled because of 
 the above hangup problem which is NOT consistent).
 
 I've tried to debug the asterisk log but there are NO ERRORS!
 
 I have asterisk installed on a Dell 2850 server with dual Xeon CPU's.
 I'm running CentOS 4.3 x86_64 and asterisk SVN-branch-1.2-r38611M with
 freepbx-2.1.1 ontop of it all.
 
 I would really appreciate some thoughts on this. Please ask me for 
 furhter info if needed since I'm no debugger. It's a hell of a task 
 to reinstall the whole server so I'd like to know what went wrong this 
 time first.
 
 Regards,
 Jan
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Office Manager, Horan  Company, LLC
(907) 747- x112
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Re: [asterisk-users] Asterisk config with Analouge Audio Codec model number MP108FXS

[ram]

Hi

just creat accounts and configure MP-108FXS
per port basis

its working in my setup

ram
On 8/2/06, Mr shobhit nirala [EMAIL PROTECTED] wrote:


Hi Friends 
 I want to configure my asterisk server with audio codec model MP-108FXS
 my sip.conf has the user name with mohit i want to configure this with 
 analouge



SHOBHIT NIRALA CONT NO. 9871476403



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RE: [asterisk-users] rx_fax problem

[Steve Hanselman]

Rxfax has no ECM, try hylafax and iaxmodem.

Steve


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paradise
Dove
Sent: 01 August 2006 21:28
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] rx_fax problem

hi,
rx_fax fails to get fax on a bit noisy lines
but real fax devices can do that on the same line
with no problem!
what's the problem?

thanks
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SV: [asterisk-users] Help debugging strange asterisk behaviour

[jan.sarin]

I'm thinking this could be a queue problem?

But I still don't understand why the hell it just flips out after a few hours. 
Now it all ran for about 12 hours since last reboot (longest so far). And this 
config worked on my old install of asterisk...

Problem description (one of them):
Incoming call gets answered and processed all the way to exten = 
1,n,Queue(1000|tTn|||300). But it seems like the queue doesn't pick up the 
call, nothing happens. No MoH, no nothing, just silence. Caller is not sent to 
free agent. It just hangs there.

Example of incomming call that gets sent to queue (from extensions_custom.conf):

exten = 1,1,Macro(custom-callerid,${CALLERIDNUM},SPARR)
exten = 1,n,Wait(2)
exten = 1,n,Set(QUEUE_PRIO=10)
exten = 
1,n,Set(MONITOR_FILENAME=/new/monitor/queues/${TIMESTAMP}-${CALLERID(name)}-${CALLERID(num)}-${UNIQUEID})
exten = 1,n,Queue(1000|tTn|||300)
exten = 1,n,Macro(failover-alarm,SPARR,custom-incoming-3000,1,4)

My queues_custom.conf:

[1000]
wrapuptime=10
timeout=600
strategy=leastrecent
retry=15
queue-youarenext=
queue-thereare=
queue-thankyou=queue-thankyou
queue-callswaiting=
music=default
monitor-join=yes
monitor-format=wav
maxlen=0
leavewhenempty=strict
joinempty=strict
context=
announce-holdtime=no
announce-frequency=0
periodic-announce=custom/general_queue_message
periodic-announce-frequency=60

member=Agent/1001 ; Agent 1001


[1001]
wrapuptime=10
timeout=600
strategy=leastrecent
retry=15
queue-youarenext=
queue-thereare=
queue-thankyou=queue-thankyou
queue-callswaiting=
music=default
monitor-join=yes
monitor-format=wav
maxlen=0
leavewhenempty=strict
joinempty=strict
context=
announce-holdtime=no
announce-frequency=0
periodic-announce=custom/general_queue_message
periodic-announce-frequency=60

member=Agent/1001 ; Agent 1001


[1002]
wrapuptime=10
timeout=600
strategy=leastrecent
retry=15
queue-youarenext=
queue-thereare=
queue-thankyou=queue-thankyou
queue-callswaiting=
music=default
monitor-join=yes
monitor-format=wav
maxlen=0
leavewhenempty=strict
joinempty=strict
context=
announce-holdtime=no
announce-frequency=0
periodic-announce=custom/general_queue_message
periodic-announce-frequency=60

member=Agent/1008 ; Agent 1008


-Ursprungligt meddelande-
Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För [EMAIL PROTECTED]
Skickat: den 2 augusti 2006 11:52
Till: asterisk-users@lists.digium.com
Ämne: SV: [asterisk-users] Help debugging strange asterisk behaviour

I think I'm using native since I don't recall installing anything else (except 
lame codec). How do I check which I am using? I'm unfortunately no asterisk 
expert that's why I need your help! ;)

My musiconhold.conf (I have no musiconhold_additional.conf):
;
; Music on hold class definitions
; This is using the new 1.2 config file format, and will not work with 1.0 ; 
based Asterisk systems ; [default] mode=files
directory=/var/lib/asterisk/mohmp3
#include musiconhold_additional.conf

Thanks very much for your time!

Regards,
Jan

-Ursprungligt meddelande-
Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För Mojo with Horan  
Company, LLC
Skickat: den 1 augusti 2006 23:20
Till: Asterisk Users Mailing List - Non-Commercial Discussion
Ämne: Re: [asterisk-users] Help debugging strange asterisk behaviour

Are you using mpg123 for MoH or native?  What's in your musiconhold.conf?



[EMAIL PROTECTED] wrote:
 Hi,
 
 I'm one of those types who want to know what the heck is wrong when 
 something is wrong.
 
 I just installed a new server (see config below) and it all works fine 
 for a few hours. But after 3-5 hours asterisk starts behaving VERY 
 strangely for no apparent reason...
 
 1) MoH stops playing
 2) Some calls are not hung up from Zap-side
 3) Flash Operator Panel starts showing all kind of random letters.
 4) Agents are unable to login/logout.
 
 ..and so on. But the strange thing is that some things seem to work 
 perfectly fine as usual. Inbound calls are getting playbacks() but no 
 MoH when sent to queue, and caller is not sent to an agent. Outgoing 
 sip and zap calls work fine (until all zapchans are filled because of 
 the above hangup problem which is NOT consistent).
 
 I've tried to debug the asterisk log but there are NO ERRORS!
 
 I have asterisk installed on a Dell 2850 server with dual Xeon CPU's.
 I'm running CentOS 4.3 x86_64 and asterisk SVN-branch-1.2-r38611M with
 freepbx-2.1.1 ontop of it all.
 
 I would really appreciate some thoughts on this. Please ask me for 
 furhter info if needed since I'm no debugger. It's a hell of a task 
 to reinstall the whole server so I'd like to know what went wrong this 
 time first.
 
 Regards,
 Jan
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Re: [asterisk-users] Asterisk with VoIP phone

[Bruce Reeves]

Yes this is possible, you just setup the softphones and maybe the provider in sip.conf and write your dialplan :)On 8/1/06, J Rangi 
[EMAIL PROTECTED] wrote:Hello,Is is possible to setup an asterisk server with out buying Digium card.
I mean can we do this type of setup.We all know that X-Lite can be used as a soft phone to have an IPextension.Is it possible to take a service from another VoIP service provider, andget the IP phone number. Make that phone numbe gateway to outside world.
Now all the internal extensions use that phone to receive and make callsto out side world.Has any one done this kind of setup or know anything about this.Thank you,-Jai___
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[asterisk-users] newbie - suggestions on installing Asterisk for SOHO

[Pele Zico]

Hi


can anyone suggest the best way to go about installing Asterisk for a small
business of 20 - ive read all the documentation but the real issue is the
dial plan.  Ive attempted to write my own which is similar to FreePBX's
dailplan.  Ive even attempted to rewrite dialparties.pl simply to increase
my knowlegde of dailplan writing.  The issue i have is all the functions
that come with freebpx are they all necessary.  

If so it look as if i will have to opt for freepbx though i would rather
continue from the bottom up and delevope my own system.  Point is at this
moment is hunt groups is this a feature i should use or can i dismiss. 
Reason for asking this is because i dont yet know perl or php so how can i
iterate thruogh a hunt group checking for CW CF CFB etc.  Thats the point
in a nutshell


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Re: [asterisk-users] Unicall stack, right versions?

[Steve Underwood]

Barzilai Spinak wrote:


Thank you Steve.
About the configs in Asterisk... I confess that I'm new to the code so 
I still need to read more. I didn't know about ast_config()


About the hardcodedness of the countries... that seems to be the 
problem. Everything is too oriented to my country works like this 
with this telephone company.
When in fact, what I'm using is not even to connect it to a the 
telephone company of my country but to some other machine which has an 
old Call Center implementation with some other modification of the MF 
R2 sequence.
It doesn't relate specifically to any country. Yes, they are all 
similar, and being able to specify the number of ANI and DNIS/DID is 
sometimes all you need, that's why I could make it work.


Well that's just a weird system. Config files tend not to get over 
those. See Mexico support in Unicall for an example. :-)




There's some truth in your statement that opening the configuration to 
external files may get some people into trouble.

On the other hand, what I see is a strange mix of:
a) If you're doing telephony stuff you should know what you're doing
b) Most people using Unicall (Asterisk for that matter) have very 
little idea of what they are doing and why (copying and pasting 
configs from here and there).


So, where's the sweet spot? :-)


Most users are in category b. Usable by people who don't know what they 
are doing is paramount. As I said, config files don't get over most 
problems beyond what you can configure right now.




I can spend 1 hour reading the source code and finally knowing how to 
change it to my needs. (For example, adding a new country)

Should I need to? Can people from the (b) set do it?
Is it scalable?  What is more of a support nightmare?

Please take all this as constructive comments. I really appreciate 
your work and if I had to do it from the start it would take me months 
longer!!!



A real question that should go in a different mail, but what the check:

Let's say I have two E1 spans, but one needs to talk 
CountryFooVersion, and the other needs CountryBarVersion (yes, both on 
the same machine and in the same country, maybe different number of 
digits for ANI).


Each channel is individually configured. You could have 30 different 
configurations on a single E1.


How would I go about configing that? 


In a unicall.conf file for chan_unicall? Set a configuration. Define 
some channels. Set a new configuration. Define some more channels. The 
most recent configuration is used as each channel is defined.


Regards,
Steve

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[asterisk-users] Follow ON calling on DISA

[[EMAIL PROTECTED]]

Hello all,I'm a user of the latest version of TrixBox. I would like to know from the users if any one has implemented the follow on calling system on DISA.If you dont understand what i mean, let me make it clear.
When i'm calling from PSTN to my asterisk using DISA and call a trunk that i have configured, i can place a call and after the call i have to hangup and dial back in again to make another call. Instead i would like to use a key combination to hangup and get my dialtone back again to do my second call (eg: ## to hangup the current call).
If anyone knows how to do this please help me out.thanks in advance.Dan
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[asterisk-users] FXO module burn out !?

[Rostislav Bagrov]

Hello list,

Howto be shure if one FXO module on TDM400P is not working because is burn out 
or something like physically demaged. It worked for an almost a year then just 
stopped. The next FXO module on the same card is working like charm.



Thanks.

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RE: [asterisk-users] FXO module burn out !?

[jacobso1]

Hi,

Try to swap both fxo modules.
This way you will notice if the module is out or another problem is present.
The tdm400 could be damaged or there is a configuration problem.

Regards,


-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
Hello list,

Howto be shure if one FXO module on TDM400P is not working because is burn
out or something like physically demaged. It worked for an almost a year
then just stopped. The next FXO module on the same card is working like
charm.



Thanks.


-- 
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Re: [asterisk-users] FXO module burn out !?

[Andrew Kohlsmith]

On Wednesday 02 August 2006 08:52, Rostislav Bagrov wrote:
 Howto be shure if one FXO module on TDM400P is not working because is burn
 out or something like physically demaged. It worked for an almost a year
 then just stopped. The next FXO module on the same card is working like
 charm.

Your message leaves as many questions as it asks.  Typically though unless 
you're familiar with electronic design and troubleshooting there is little 
difference between not working due to burnout and not working.

These modules have a two year warranty from Digium; why not call their 
technical support centre and get the module replaced?  It's very likely under 
warranty and therefore the time spent trying to diagnose it is more or less a 
waste; just replace it and let their warranty department handle the why.

If you feel that the module received a surge or other transient, why not 
attach a surge suppressor and/or telephone line conditioning and protection 
equipment to the line?  I'd recommend that for any equipment connected to 
copper leaving the building.

-A.
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Re: [asterisk-users] Unicall stack, right versions?

[leonimar cape]

Hi Steve,

I need to enable the Unicall channel in my asterisk
box to be able to interconnect to a local telco
provider using MFCR2. I use the unicall release
unicall-0.0.3pre9 and a patch for asterisk 1.2.
Compilation was done with ease. The problem is that I
got an error Unable to read supervisory tone set hk
once I load the chan. I am using 1.2.9.1 version. Is
there any new version that could use.

Regards,

Leonimar
 

--- Steve Underwood [EMAIL PROTECTED] wrote:

 Barzilai Spinak wrote:
 
  Thank you Steve.
  About the configs in Asterisk... I confess that
 I'm new to the code so 
  I still need to read more. I didn't know about
 ast_config()
 
  About the hardcodedness of the countries... that
 seems to be the 
  problem. Everything is too oriented to my
 country works like this 
  with this telephone company.
  When in fact, what I'm using is not even to
 connect it to a the 
  telephone company of my country but to some other
 machine which has an 
  old Call Center implementation with some other
 modification of the MF 
  R2 sequence.
  It doesn't relate specifically to any country.
 Yes, they are all 
  similar, and being able to specify the number of
 ANI and DNIS/DID is 
  sometimes all you need, that's why I could make it
 work.
 
 Well that's just a weird system. Config files tend
 not to get over 
 those. See Mexico support in Unicall for an example.
 :-)
 
 
  There's some truth in your statement that opening
 the configuration to 
  external files may get some people into trouble.
  On the other hand, what I see is a strange mix of:
  a) If you're doing telephony stuff you should know
 what you're doing
  b) Most people using Unicall (Asterisk for that
 matter) have very 
  little idea of what they are doing and why
 (copying and pasting 
  configs from here and there).
 
  So, where's the sweet spot? :-)
 
 Most users are in category b. Usable by people who
 don't know what they 
 are doing is paramount. As I said, config files
 don't get over most 
 problems beyond what you can configure right now.
 
 
  I can spend 1 hour reading the source code and
 finally knowing how to 
  change it to my needs. (For example, adding a new
 country)
  Should I need to? Can people from the (b) set do
 it?
  Is it scalable?  What is more of a support
 nightmare?
 
  Please take all this as constructive comments. I
 really appreciate 
  your work and if I had to do it from the start it
 would take me months 
  longer!!!
 
 
  A real question that should go in a different
 mail, but what the check:
 
  Let's say I have two E1 spans, but one needs to
 talk 
  CountryFooVersion, and the other needs
 CountryBarVersion (yes, both on 
  the same machine and in the same country, maybe
 different number of 
  digits for ANI).
 
 Each channel is individually configured. You could
 have 30 different 
 configurations on a single E1.
 
  How would I go about configing that? 
 
 In a unicall.conf file for chan_unicall? Set a
 configuration. Define 
 some channels. Set a new configuration. Define some
 more channels. The 
 most recent configuration is used as each channel is
 defined.
 
 Regards,
 Steve
 
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[asterisk-users] polycom soundstation 501 crash

[Stas Khromoy]

hey folks

hope some one came across this problem

one of our polycom's just crashed
after reboot it comes up with this error

error loading 0004f204fcc.cfg


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[asterisk-users] Dell Poweredge 1950 / 2950

[Frédéric Marti]

Hi 
all,

Anybody have experience (good/bad) with Dell Poweredge 1950 / 2950 
?
I'll install 
Fedora Core 5 or #PoundKey and Digium Hardware.


Regards
Fred
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[asterisk-users] Call Routing based on Caller-Id

[Matthew Crocker]

Hello,

 How can I build extensions.conf so that Asterisk routes calls based  
on the ANI, not the number dialed.


Example:

All calls coming down a PRI are going to the same number.  I would  
like to route them to a new number based on the Calling-Station-Id.   
I.E. All calls from 413-773- go to 413-773-1234


-Matt


--
Matthew S. Crocker
Vice President
Crocker Communications, Inc.
Internet Division
PO BOX 710
Greenfield, MA 01302-0710
http://www.crocker.com

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[asterisk-users] Playback() does not work

[Camilo Echeverry]

Hi.I've installed Asterisk with a MD3200 modem,zaptel modules recognize the card,when i dial to asterisk, it answers but when I Playback(something) do not receive any audio, only a sound like audio static
but I created in extensions.conf[demo]iclude= defaultand when in the console type the commandCLI dial sthe [default] context (included by [demo]) plays perfectly on the soundcard
Notice that I only modified these files:zapte.confzapata.confextensions.confAny Idea ..?Am I missing something ..?--ThanksCamilo.
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RE: [asterisk-users] polycom soundstation 501 crash

[Alexander Lopez]

Make sure you can access the file on your FPT server. Also make sure
that you did not fry the Ethernet port(s) on the phone.


 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Stas Khromoy
 Sent: Wednesday, August 02, 2006 9:50 AM
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] polycom soundstation 501 crash
 
 hey folks
 
 hope some one came across this problem
 
 one of our polycom's just crashed
 after reboot it comes up with this error
 
 error loading 0004f204fcc.cfg
 
 
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Re: [asterisk-users] polycom soundstation 501 crash

[Stas Khromoy]

i am afraid the second could be the case
since the whole block where the office is lost power yesterday

thanks

PS : FPT = FTP ? :)


 Original Message  
Subject: Re:[asterisk-users] polycom soundstation 501 crash
From: Alexander Lopez [EMAIL PROTECTED]
To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial 
Discussion asterisk-users@lists.digium.com

Date: 8/2/2006 10:13 AM

Make sure you can access the file on your FPT server. Also make sure
that you did not fry the Ethernet port(s) on the phone.


  

-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Stas Khromoy
Sent: Wednesday, August 02, 2006 9:50 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] polycom soundstation 501 crash

hey folks

hope some one came across this problem

one of our polycom's just crashed
after reboot it comes up with this error

error loading 0004f204fcc.cfg


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Re: [asterisk-users] Unicall stack, right versions?

[Moises Silva]

Is possible that you are missing the XML file with the supertones
definitions. Usually is located at /usr/share/spandsp/global-tones.xml
, but it depends on how you configured the spandsp package
(./configure --prefix=/usr/blah). Notice that spandsp and
libsupertone should be configured with the same prefix, so supertone
can look for the file in the correct directory.

Also, hk is the code for Hong Kong tones. You must put in unicall.conf
a parameter called supertones=country code. That country code must
be defined in the XML file i just mentioned. For mexico, supertones=mx

Regards

On 8/2/06, leonimar cape [EMAIL PROTECTED] wrote:

Hi Steve,

I need to enable the Unicall channel in my asterisk
box to be able to interconnect to a local telco
provider using MFCR2. I use the unicall release
unicall-0.0.3pre9 and a patch for asterisk 1.2.
Compilation was done with ease. The problem is that I
got an error Unable to read supervisory tone set hk
once I load the chan. I am using 1.2.9.1 version. Is
there any new version that could use.

Regards,

Leonimar


--- Steve Underwood [EMAIL PROTECTED] wrote:

 Barzilai Spinak wrote:

  Thank you Steve.
  About the configs in Asterisk... I confess that
 I'm new to the code so
  I still need to read more. I didn't know about
 ast_config()
 
  About the hardcodedness of the countries... that
 seems to be the
  problem. Everything is too oriented to my
 country works like this
  with this telephone company.
  When in fact, what I'm using is not even to
 connect it to a the
  telephone company of my country but to some other
 machine which has an
  old Call Center implementation with some other
 modification of the MF
  R2 sequence.
  It doesn't relate specifically to any country.
 Yes, they are all
  similar, and being able to specify the number of
 ANI and DNIS/DID is
  sometimes all you need, that's why I could make it
 work.

 Well that's just a weird system. Config files tend
 not to get over
 those. See Mexico support in Unicall for an example.
 :-)

 
  There's some truth in your statement that opening
 the configuration to
  external files may get some people into trouble.
  On the other hand, what I see is a strange mix of:
  a) If you're doing telephony stuff you should know
 what you're doing
  b) Most people using Unicall (Asterisk for that
 matter) have very
  little idea of what they are doing and why
 (copying and pasting
  configs from here and there).
 
  So, where's the sweet spot? :-)

 Most users are in category b. Usable by people who
 don't know what they
 are doing is paramount. As I said, config files
 don't get over most
 problems beyond what you can configure right now.

 
  I can spend 1 hour reading the source code and
 finally knowing how to
  change it to my needs. (For example, adding a new
 country)
  Should I need to? Can people from the (b) set do
 it?
  Is it scalable?  What is more of a support
 nightmare?
 
  Please take all this as constructive comments. I
 really appreciate
  your work and if I had to do it from the start it
 would take me months
  longer!!!
 
 
  A real question that should go in a different
 mail, but what the check:
 
  Let's say I have two E1 spans, but one needs to
 talk
  CountryFooVersion, and the other needs
 CountryBarVersion (yes, both on
  the same machine and in the same country, maybe
 different number of
  digits for ANI).

 Each channel is individually configured. You could
 have 30 different
 configurations on a single E1.

  How would I go about configing that?

 In a unicall.conf file for chan_unicall? Set a
 configuration. Define
 some channels. Set a new configuration. Define some
 more channels. The
 most recent configuration is used as each channel is
 defined.

 Regards,
 Steve

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--
Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org;
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Re: [asterisk-users] polycom soundstation 501 crash

[Jessee J Holmes]

This means the phone is attempting to load this configuration file and cannot find it from your boot server. The phone at this point must have a boot server with these files. Put this file on a FTP server and point the phone to that server to pickup and download this file. Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products?  Visit our VoIP store at http://voipstore.atacomm.com/ On Aug 2, 2006, at 8:50 AM, Stas Khromoy wrote:hey folkshope some one came across this problemone of our polycom's just crashedafter reboot it comes up with this errorerror loading 0004f204fcc.cfg___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:  http://lists.digium.com/mailman/listinfo/asterisk-users ___
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Re: [asterisk-users] polycom soundstation 501 crash

[Stas Khromoy]

not to sound like an idiot
but where do i get the files ?


these guys ?

http://www.polycom.com/resource_center/0,1454,pw-6812-12612,FF.html
SoundPoint IP/SoundStation IP SIP Software 1.6.7
SoundPoint IP/SoundStation IP BootROM 3.2.1


 Original Message  
Subject: Re:[asterisk-users] polycom soundstation 501 crash
From: Jessee J Holmes [EMAIL PROTECTED]
To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial 
Discussion asterisk-users@lists.digium.com

Date: 8/2/2006 10:20 AM
This means the phone is attempting to load this configuration file and 
cannot find it from your boot server. The phone at this point must 
have a boot server with these files. Put this file on a FTP server and 
point the phone to that server to pickup and download this file.



Jessee Holmes

Atacomm / Ataractic Corporation

www.atacomm.com

V: 1-877-700-VOIP

[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]


Looking for voice over IP products?  Visit our VoIP store at 
http://voipstore.atacomm.com/




On Aug 2, 2006, at 8:50 AM, Stas Khromoy wrote:


hey folks

hope some one came across this problem

one of our polycom's just crashed
after reboot it comes up with this error

error loading 0004f204fcc.cfg


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[asterisk-users] Strange behaviour Panasonic KX-TD1232

[Pablo Mora]

I think still didnt explain me clearly



The problem is when I dial 0, in this case the asterisk
take Zap (connected directly to ext 200 from Panasonic), Panasonic gives tone,
dial another extension (ie 100), the extension rings but when answer the phone asterisk
keeps ringing it doesnt detect when you pick up the phone.










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RE: [asterisk-users] FXO module burn out !?

[asterisk]

Swap the modules and see if fault moves with the module. If it moves then
the module could be faulty, if not, the module is ok

neil

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rostislav
Bagrov
Sent: 02 August 2006 13:52
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] FXO module burn out !?

Hello list,

Howto be shure if one FXO module on TDM400P is not working because is burn
out or something like physically demaged. It worked for an almost a year
then just stopped. The next FXO module on the same card is working like
charm.



Thanks.

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Re: [asterisk-users] polycom soundstation 501 crash

[Sebastian Milioto]

Hi all,

I have a problem sending attached files with voicemail. I have postfix
installed.

When I write attach=no in voicmail.conf the notification is sent with
no problem. But when I change to attach=yes, the notification never
arrives.

Could it be a postfix problem? Anybody could tell me how to configure
it to permit attached files?

I'm using mandriva 2006

Thanks very much in advance

Sebastian
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Re: [asterisk-users] polycom soundstation 501 crash

[Jessee J Holmes]

At first, you would have to get these from your service provider or your reseller. They should have them available. I wish I could find a sample of one of the .cfg files, but I can't seem to locate it at this moment; however, here is a starting sample from Polycom.File Name: .cfg?xml version="1.0" standalone="yes"?!-- Default Master SIP Configuration File--!-- Edit and rename this file to Ethernet-address.cfg for each phone.--!-- $Revision: 1.14 $  $Date: 2005/07/27 18:43:30 $ --APPLICATION APP_FILE_PATH="sip.ld" CONFIG_FILES="phone1.cfg, sip.cfg" MISC_FILES="" LOG_FILE_DIRECTORY="" OVERRIDES_DIRECTORY="" CONTACTS_DIRECTORY=""/Obviously, rename the file to the MAC address of the phone and change the text within the file to match up with your phone and preferred settings.If I can find a full working sample, I'll send it. Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED]Looking for voice over IP products?  Visit our VoIP store at http://voipstore.atacomm.com/ On Aug 2, 2006, at 9:28 AM, Stas Khromoy wrote:not to sound like an idiotbut where do i get the files ?these guys ?http://www.polycom.com/resource_center/0,1454,pw-6812-12612,FF.htmlSoundPoint IP/SoundStation IP SIP Software 1.6.7SoundPoint IP/SoundStation IP BootROM 3.2.1 Original Message  Subject: Re:[asterisk-users] polycom soundstation 501 crashFrom: Jessee J Holmes [EMAIL PROTECTED]To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.comDate: 8/2/2006 10:20 AM This means the phone is attempting to load this configuration file and cannot find it from your boot server. The phone at this point must have a boot server with these files. Put this file on a FTP server and point the phone to that server to pickup and download this file.Jessee HolmesAtacomm / Ataractic Corporationwww.atacomm.comV: 1-877-700-VOIP[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]Looking for voice over IP products?  Visit our VoIP store at http://voipstore.atacomm.com/On Aug 2, 2006, at 8:50 AM, Stas Khromoy wrote: hey folkshope some one came across this problemone of our polycom's just crashedafter reboot it comes up with this errorerror loading 0004f204fcc.cfg___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:  http://lists.digium.com/mailman/listinfo/asterisk-users   ___
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[asterisk-users] Asterisk, Linksys SPA-3000 echo

[Dean @ INKnBITs]

Hi,

I have been looking through all the web sites about echo problems and how to
solve them on the spa-3000, but I still have not managed to fix mine! I'm in
the UK and have setup all the tones, port impedance to 370+620||310nF, had
the echo Canc options on and off, turned down the SPA - PSTN gain until it
would not even dial, still had the echo. It not a little echo, its enough
you cannot really use it. (I have 3 units, and they all do the same)

I'm using the SPA to route PSTN calls to asterisk, then polycom IP501 to
asterisk.

Hardware version: 3.0.0(1178)
Software version: 3.1.3(GWa)

I have tried 3.1.10 and 2.0.13, but still have the same problem, the other
party has a clear call, but I can hear myself as an echo.

Has anybody got these to work ok? If so, please, please, please could I have
the settings you used, it's been driving me crazy for the last two weeks.


Thanks,
Dean.

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[asterisk-users] Arrays ???

[Pele Zico]

Is their a way of implementing arrays in asterisk??  What im trying to do is
allow the user to input numbers (internal and external) from his line thats
gets saved via astdb - (follow me purposes) - i would like to look t each
number individually and check for CF, CFB CFU etc etc . can this be done or
would i have to use AGI.  i prefer not to use agi however

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Re: [asterisk-users] polycom soundstation 501 crash

[Jim Rice]

 error loading 0004f204fcc.cfg 

Missing a digit?


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[asterisk-users] Issue with IAX2 and Real Time configuration

[Facundo Ameal]

Hello everybody,
I'm having a problem trying to dial with an IAX2 extensions. I connect
trough iaxComm and try to dial an extensions, then in asterisk  CLI
appears this:

Aug  3 01:14:29 NOTICE[20915]: chan_iax2.c:7357 socket_read: Rejected
connect attempt from 192.168.1.128, requested/capability 0x2/0x2
incompatible with our capability 0xf90c.

I googled it but nothing appears. I have asterisk, zaptel and libpri
from SVN brach 1.2.

Thanks in advance.

Greets.

--
Facundo Ameal.
famealatgmaildotcom
Linux User #395088

FWD: 741664
MSN: asadoatlamorcilladotcomdotar
ICQ: 74005793


Open your mind, use open source.
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[asterisk-users] [ANN] - Coder Needed for Patch

[Bart Fisher]

I've posted on GAF (Free Lance Site) a request for bids for 
modifications to Asterisk PBX source.
If you are interest in bidding on this, please view it at 
http://www.getafreelancer.com/projects/78138.html


Thanks you for your time.

Bart Fisher
[EMAIL PROTECTED]


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[asterisk-users] Re: Issue with IAX2 and Real Time configuration

[Facundo Ameal]

It's solved. The problem was that the softphone has only one codec
allowed and asterisk was configured to no allow that codec.

On 8/2/06, Facundo Ameal [EMAIL PROTECTED] wrote:

Hello everybody,
I'm having a problem trying to dial with an IAX2 extensions. I connect
trough iaxComm and try to dial an extensions, then in asterisk  CLI
appears this:

Aug  3 01:14:29 NOTICE[20915]: chan_iax2.c:7357 socket_read: Rejected
connect attempt from 192.168.1.128, requested/capability 0x2/0x2
incompatible with our capability 0xf90c.

I googled it but nothing appears. I have asterisk, zaptel and libpri
from SVN brach 1.2.

Thanks in advance.

Greets.

--
Facundo Ameal.
famealatgmaildotcom
Linux User #395088

FWD: 741664
MSN: asadoatlamorcilladotcomdotar
ICQ: 74005793


Open your mind, use open source.




--
Facundo Ameal.
famealatgmaildotcom
Linux User #395088

FWD: 741664
MSN: asadoatlamorcilladotcomdotar
ICQ: 74005793


Open your mind, use open source.
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Re: [asterisk-users] FXO module burn out !?

[Alex Robar]

Rostislav,We just experienced this ourselves. First our second module stopped working, then our fourth. We popped an old Sangoma card in and the lines worked fine, so I figure it must be the modules. I'm going to swap the modules in a tester box soon and see if the suspected faulty ones light up. If they're still not working, I'll call Digium. They're under a two year warranty, and I've heard it's usually not a problem to get replacements.
AlexOn 8/2/06, Rostislav Bagrov 
[EMAIL PROTECTED]
 wrote:
Hello list,Howto be shure if one FXO module on TDM400P is not working because is burn out or something like physically demaged. It worked for an almost a year then just stopped. The next FXO module on the same card is working like charm.
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[EMAIL PROTECTED]


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Re: [asterisk-users] Asterisk, Linksys SPA-3000 echo

[Rich Adamson]

Dean @ INKnBITs wrote:

Hi,

I have been looking through all the web sites about echo problems and how to
solve them on the spa-3000, but I still have not managed to fix mine! I'm in
the UK and have setup all the tones, port impedance to 370+620||310nF, had
the echo Canc options on and off, turned down the SPA - PSTN gain until it
would not even dial, still had the echo. It not a little echo, its enough
you cannot really use it. (I have 3 units, and they all do the same)

I'm using the SPA to route PSTN calls to asterisk, then polycom IP501 to
asterisk.

Hardware version: 3.0.0(1178)
Software version: 3.1.3(GWa)

I have tried 3.1.10 and 2.0.13, but still have the same problem, the other
party has a clear call, but I can hear myself as an echo.

Has anybody got these to work ok? If so, please, please, please could I have
the settings you used, it's been driving me crazy for the last two weeks.


In the US, the spa3k has echo problems with pstn lines that are somewhat 
long, and with pstn lines that are based on some central office remote 
line concentrators.  The spa3k seems to work fairly well on short pstn 
lines, but I've not tried to figure out exactly how long a pstn must be 
before echo becomes a problem.


I know in my case (four pstn lines from two different CO's) the spa3k 
does not do a very good job with echo cancellation, and these lines are 
roughly 7,000 feet. At least one of the lines is via a remote line 
concentrator (from a more distant CO), and it too has issues.


I've spent a significant amount of time with a commercial transmission 
test set to identify the operational parameters around these lines, but 
that only provides a perspective as to length, loss, etc. I gave up on 
the spa3k's as well as the TDM400, and installed a A200d. That has been 
providing excellent audio and echo cancellation for roughly six months.



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Re: [asterisk-users] Arrays ???

[Benchev]

On Wednesday 02 August 2006 18:09, Pele Zico wrote:
 Is their a way of implementing arrays in asterisk??  What im trying to do
 is allow the user to input numbers (internal and external) from his line
 thats gets saved via astdb - (follow me purposes) - i would like to look t
 each number individually and check for CF, CFB CFU etc etc . can this be
 done or would i have to use AGI.  i prefer not to use agi however
*CLI show version
Asterisk SVN-trunk-r37291 
*CLI show function ARRAY

  -= Info about function 'ARRAY' =-

[Syntax]
ARRAY(var1[|var2[...][|varN]])

[Synopsis]
Allows setting multiple variables at once

[Description]
The comma-separated list passed as a value to which the function is set will
be interpreted as a set of values to which the comma-separated list of
variable names in the argument should be set.
Hence, Set(ARRAY(var1|var2)=1\,2) will set var1 to 1 and var2 to 2
Note: remember to either backslash your commas in extensions.conf or quote the
entire argument, since Set can take multiple arguments itself.

So probably with 1.4 will come.

Benchev
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[asterisk-users] SIP_HEADER() read-only

[Vincent Regnard]

Hi,

Having checked the documentation for SIP_HEADER:

pitux-exercice15*CLI
  -= Info about function 'SIP_HEADER' =-

[Syntax]
SIP_HEADER(name)

[Synopsis]
Gets or sets the specified SIP header

I thought I could write some info in SIP_HEADER to retrieve them later.

But when I try to write to it:

exten = s,n,Set(SIP_HEADER(DN)=toto)
exten = s,n,NoOp(Sip DN ${SIP_HEADER(DN)})

Write is refused:

2006-08-02 16:16:09 VERBOSE[5224] logger.c: -- Executing 
Set(SIP/220-aa94, SIP_HEADER(DN)=toto) in new stack
2006-08-02 16:16:09 ERROR[5224] pbx.c: Function SIP_HEADER is read-only, 
it cannot be written to

2006-08-02 16:16:09 DEBUG[5224] pbx.c: Function result is '(null)'


Is this function really read-write ? Is there something I could check or 
modify to achieve my goal (writing to sip header) ?
Is there a switch somewhere to allow to write to it ? Any global or 
channel variable or parameter to set for that ?


Thanks for your help and comments.

I run asterisk 1.2.4.
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[asterisk-users] Limitations of IAX

[Douglas Garstang]

I'm about ready to give up on IAX2. It seems to have some SERIOUS limitations.

Incoming PSTN call comes into to user A on pbx1. We look for user A locally, 
and don't find them. We then do a DUNDi lookup, get a path, and dial user A on 
pbx2 with IAX2. User A picks up the call.

When IAX passes the call from pbx1 to pbx2, it does not pass some of the dial 
plan variables. Namely dnid, which is set to 'unknown'. Consequently (for 
reasons I don't yet understand) when user A on pbx2 tries to do a blind 
transfer, the dnid is not set, and as a result, we _must_ fail the call because 
we have no source number on our network. 

Also, the account code is not picked up on pbx2 as well.

:(


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[asterisk-users] canreinvite=yes and RTP dropping in and out

[Gary Richardson]

Hey guys,I'm having yet another strange problem. I've recently set canreinvite=yes, allowing the RTP streams to avoid our * server. Now, a few people are experience one way audio drops on internal calls. External calls are working fine (they re-invite directly to a Cisco router). Sometimes, if you wait 20 seconds or more, the stream will resume. Flipping the person on and off hold won't resume the stream.
We're using 7960 phones. Enabled_vad is set to 0 (disabled). It doesn't seem to happen all of the time. There are no sip messages being exchanged when the stream stops or restarts.Any suggestions?Thanks.

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[asterisk-users] DTMF intermittent on menu.

[Shane Burrell]

Anyone seen DTMF control lost intermittently inside a menu?
We have had a few problems with one particular menu but both seem identical but
maybe different traffic.



Shane









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Re: [asterisk-users] Strange behaviour Panasonic KX-TD1232

[C F]

Then you have something wrong some other place, if you are using an
FXO card then asterisk is not even giving you the ring, the panasonic
is.

On 8/2/06, Pablo Mora [EMAIL PROTECTED] wrote:





I think still didn't explain me clearly…



The problem is when I dial 0, in this case the asterisk take Zap (connected
directly to ext 200 from Panasonic), Panasonic gives tone, dial another
extension (ie 100), the extension rings but when answer the phone asterisk
keeps ringing… it doesn't detect when you pick up the phone.




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[asterisk-users] asterisk optimizing

[Jack Wei]

Hi,

I'm currently running Asterisk 1.2.10 on a dual Xeon 3.4GHz with 2GB RAM 
and SATA drives, but I'm only able to achieve 120 calls with good audio 
quality (using G.711u).  I'm using realtime for voicemail accounts and 
ODBC for voicemail storage along with one MySQL when dialing out.  The 
max calls I can achieve is 200 simultaneous but audio is really chopping 
due to high jitter.  Does anyone know how to optimize Asterisk and/or 
RedHat Enterprise Linux 4 to increase simultaneous calls? 


Thanks,

Jack
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Re: [asterisk-users] sip phone networking question [possibly OT]

[Mojo with Horan Company, LLC]

I have some ethernet cable splitters I'm not using any more.  They go in 
 pairs, one plugs into the wall socket in the office, the other plugs 
into the other end of the same cable in the server room.  each gives two 
female ethernet sockets that represent two separate network cables, each 
using two of the pairs.  Works great at 100Mb speeds, we needed one of 
the lines to be gigabit and they weren't cutting it for that so we ran 
more cables.


This might be considerably cheaper than extra switches, but your main 
switch has to have enough ports to accommodate the extra stations.


Contact me off-list if you would like to purchase them.  I have 9 pairs, 
which would turn 9 cables into 18.


Moj

T. Shaw wrote:
I have a client that is looking for a least cost solution  of providing 
more SIP phones to an existing asterisk setup.


The Issue is this: He has 7 total data run lines running back to the 
switch/phone room (small company).
However the want to add a total of 5 or more Phones. He doesn't want to foot 
the cost of running more data lines along the middle of a large room all the 
way to the walls and then back to the phone room.


I was wondering if we could uplink small switches to the wall data ports to 
the switch, and connect the additional SIP phones to them to get them 
connectivity to Asterisk?


As long as each phone has seperate IP, i'm not sure if there would be a 
problem.


Anyone care to chime in to point out any pitfalls or potential problems with 
this setup?


Thanks!



Terrelle


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!DSPAM:500,44ce77bb139581298614243!



--
Mojo [EMAIL PROTECTED]
Office Manager, Horan  Company, LLC
(907) 747- x112
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RE: [asterisk-users] cmd DIAL - Who picked up the call?

[Koopmann, Jan-Peter]

On Wednesday, August 02, 2006 6:49 AM Eric ManxPower Wieling wrote:

 Zap/10-43 would indicate that this is the 43rd call (call waiting) on
 channel 10.  Obviously this would have to be removed to do it the way
 you want.  

Obviously. :-)

Or we find another solution for the problem/challange... Ideas?
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Re: [asterisk-users] canreinvite=yes and RTP dropping in and out

[Joshua Colp]

- Original Message -
From: Gary Richardson
[mailto:[EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Non-Commercial Discussion [mailto:[EMAIL PROTECTED]
Sent:
Wed, 02 Aug 2006 13:54:04 -0300
Subject: [asterisk-users] canreinvite=yes
and RTP dropping in and out


 Hey guys,
 
 I'm having yet another strange problem. I've recently set canreinvite=yes,
 allowing the RTP streams to avoid our * server. Now, a few people are
 experience one way audio drops on internal calls. External calls are working
 fine (they re-invite directly to a Cisco router). Sometimes, if you wait 20
 seconds or more, the stream will resume. Flipping the person on and off hold
 won't resume the stream.
 
 We're using 7960 phones. Enabled_vad is set to 0 (disabled). It doesn't seem
 to happen all of the time. There are no sip messages being exchanged when
 the stream stops or restarts.
 
 Any suggestions?

If the audio is going directly there's not too much you can do to examine it. 
There may be software out there to sniff the data on your network and examine 
the RTP stream, maybe even see when it drops out (if it really does drop out, 
ie: stream actually stops). I know there's some Windows software out there 
capable of this as I picked a copy up while at Spring VON but you might need to 
look around. OH - can you also send a sip debug with the reinvites? I'm just 
curious to see the RTP information in the SDP.

 Thanks.

Joshua Colp
Digium
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Re: [asterisk-users] asterisk optimizing

[tracinet]

Out of curiosity - are you running an smp kernel or a uniproc kernel? I am doing some benchmarking as well on a similar system (dual XEON 3Ghz with 4GB RAM and SATA drives in mirrors) and am seeing the uniproc kernel performing better under CentOS 
4.3 (2.6.9-34.0.2.EL) when testing general server functions (file reads/writes/copies/compiler tests/scripts, etc.). Since CentOS is basically RHEL, would be interested in what you are running.
On 8/2/06, Jack Wei [EMAIL PROTECTED] wrote:
Hi,I'm currently running Asterisk 1.2.10 on a dual Xeon 3.4GHz with 2GB RAMand SATA drives, but I'm only able to achieve 120 calls with good audioquality (using G.711u).I'm using realtime for voicemail accounts and
ODBC for voicemail storage along with one MySQL when dialing out.Themax calls I can achieve is 200 simultaneous but audio is really choppingdue to high jitter.Does anyone know how to optimize Asterisk and/or
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RE: [asterisk-users] sip phone networking question [possibly OT]

[Colin Anderson]

 I was wondering if we could uplink small switches to the wall data ports
to 
 the switch, and connect the additional SIP phones to them to get them 
 connectivity to Asterisk?

Yes, we do it and it works fine, as long as you don't cascade more than 3
switches between two devices your latency should be fine, also make sure
they are good switches like a 3com and not a crappy dlink. 
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Re: [asterisk-users] canreinvite=yes and RTP dropping in and out

[Gary Richardson]

My next attempt at this is going to be putting a hub in between the path to the switch. I'm hoping to be able to sniff the packets to see what's going on.Also, using the network status page on the hard phones, the transmit and receive counters for the direction of the channel slows way down as if almost no data is being transmitted. 
How do I send a sip debug?Thanks.On 8/2/06, Joshua Colp [EMAIL PROTECTED] wrote:
- Original Message -From: Gary Richardson[mailto:[EMAIL PROTECTED]]To: Asterisk Users Mailing List -Non-Commercial Discussion [mailto:
asterisk-users@lists.digium.com]Sent:Wed, 02 Aug 2006 13:54:04 -0300Subject: [asterisk-users] canreinvite=yesand RTP dropping in and out Hey guys, I'm having yet another strange problem. I've recently set canreinvite=yes,
 allowing the RTP streams to avoid our * server. Now, a few people are experience one way audio drops on internal calls. External calls are working fine (they re-invite directly to a Cisco router). Sometimes, if you wait 20
 seconds or more, the stream will resume. Flipping the person on and off hold won't resume the stream. We're using 7960 phones. Enabled_vad is set to 0 (disabled). It doesn't seem to happen all of the time. There are no sip messages being exchanged when
 the stream stops or restarts. Any suggestions?If the audio is going directly there's not too much you can do to examine it. There may be software out there to sniff the data on your network and examine the RTP stream, maybe even see when it drops out (if it really does drop out, ie: stream actually stops). I know there's some Windows software out there capable of this as I picked a copy up while at Spring VON but you might need to look around. OH - can you also send a sip debug with the reinvites? I'm just curious to see the RTP information in the SDP.
 Thanks.Joshua ColpDigium___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing list
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Re: [asterisk-users] Limitations of IAX

[Joshua Colp]

- Original Message -
From: Douglas Garstang
[mailto:[EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Non-Commercial Discussion [mailto:[EMAIL PROTECTED]
Sent:
Wed, 02 Aug 2006 13:50:56 -0300
Subject: [asterisk-users] Limitations of
IAX


 I'm about ready to give up on IAX2. It seems to have some SERIOUS
 limitations.

You may have run into these limitations in your deployment but others are using 
IAX2 fine for what it was designed to do. No software out there can anticipate 
everyone's needs. Those who do run into issues end up working around the 
limitations or modifying it to their needs. You will probably need to do the 
same or seek an alternate solution.

 Incoming PSTN call comes into to user A on pbx1. We look for user A locally,
 and don't find them. We then do a DUNDi lookup, get a path, and dial user A
 on pbx2 with IAX2. User A picks up the call.
 
 When IAX passes the call from pbx1 to pbx2, it does not pass some of the
 dial plan variables. Namely dnid, which is set to 'unknown'. Consequently
 (for reasons I don't yet understand) when user A on pbx2 tries to do a blind
 transfer, the dnid is not set, and as a result, we _must_ fail the call
 because we have no source number on our network. 

Have you reported a bug on this DNID not being passed?
 
 Also, the account code is not picked up on pbx2 as well.

As previously pointed out transporting the account code may be a security risk 
as well. If you really need it why don't you encode it into the dialed number 
or something?
 
 :(

Joshua Colp
Digium
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[asterisk-users] Strange behavior with SIP registration/connectivity

[Ronald Lewis]

With Asterisk 1.2.* and TRUNK, I've noticed some odd behavior with SIP registrations and connectivity over the past day. First, I noticed Asterisk REFUSED to register any trunks over SIP, prompting a lot of timeout messages. It also refused to accept registration requests from internal phones, rendering any attempt to place a call pointless. Everything was working flawlessly until yesterday. This morning, I narrowed down the SIP registrations from 5 to only 1-3 active -- they all registered fine. When I included all five, nothing registered, including internal clients.
Strange behavior -- I've never witnessed this before.
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RE: [asterisk-users] cmd DIAL - Who picked up the call?

[Vadim Berezniker]

DIALEDPEERNUMBER contains the exact peer spec for the peer that picked up. 
You can use that.



From: [EMAIL PROTECTED] on behalf of Koopmann, Jan-Peter
Sent: Wed 8/2/2006 1:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] cmd DIAL - Who picked up the call?



On Wednesday, August 02, 2006 6:49 AM Eric ManxPower Wieling wrote:

 Zap/10-43 would indicate that this is the 43rd call (call waiting) on
 channel 10.  Obviously this would have to be removed to do it the way
 you want. 

Obviously. :-)

Or we find another solution for the problem/challange... Ideas?
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winmail.dat___
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Re: [asterisk-users] Strange behaviour Panasonic KX-TD1232

[Jorge Mendoza]

Even if he has r in the dial plan?

Jorge

C F wrote:
 Then you have something wrong some other place, if you are using an
 FXO card then asterisk is not even giving you the ring, the panasonic
 is.

 On 8/2/06, Pablo Mora [EMAIL PROTECTED] wrote:




 I think still didn't explain me clearly…



 The problem is when I dial 0, in this case the asterisk take Zap
 (connected
 directly to ext 200 from Panasonic), Panasonic gives tone, dial another
 extension (ie 100), the extension rings but when answer the phone
 asterisk
 keeps ringing… it doesn't detect when you pick up the phone.




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[asterisk-users] GSM analogue router

[Garth van Sittert]

Hi All

I have a client with 3 analogue gsm routers, one of which is a 'Fusion
100'.  The other 2 routers work perfectly, but the 'Fusion 100' router
refuses to dial.  I can dial from an analogue phone connected to the
router.

From the CLI in debug mode, I can see polarity switches when trying to

initiate the call.  I have tried the answeronpolarityswitch and
hanguponpolarityswitch settings in zapata.conf with no luck.  I have a
feeling it is to do with the timing parameters in zapata.conf.  Does
anyone know where to get more information on these parameters?

Any ideas?

Kind Regards
Garth

--
Garth van Sittert
BSc (Physics  Computer Science)
-
Mobile: +27 (0)83 791 6662
Email:  [EMAIL PROTECTED]
Phone:  08600 BITCO
MSN:[EMAIL PROTECTED]
Web:www.bitco.co.za

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[asterisk-users] Rookie question, trying to learn

[Randy Paries]

Hello, I hired a consultant to setup and asterisk box for me.
I am trying to learn how to maintain some of the things myself,
because the response time on maint requests from this wonderful
consult are brutal (u know once they have their money)

anyways. currently we have it set up so when someone calls our
asterisk they are prompted for a PIN. they enter the PIN and are able
to leave a message to that mailbox.

The problem a number of people are not entering the pin fast enough
,they are not given enough time to enter the PIN( I assume this is a
mailbox number)

looking at all the doc is seems everything is configurable, can some
one point me in the right direction of where to start looking?

Thanks
Randy
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[asterisk-users] Re: Arrays ???

[Pele Zico]

Benchev wrote:

 On Wednesday 02 August 2006 18:09, Pele Zico wrote:
 Is their a way of implementing arrays in asterisk??  What im trying to do
 is allow the user to input numbers (internal and external) from his line
 thats gets saved via astdb - (follow me purposes) - i would like to look
 t each number individually and check for CF, CFB CFU etc etc . can this
 be
 done or would i have to use AGI.  i prefer not to use agi however
 *CLI show version
 Asterisk SVN-trunk-r37291
 *CLI show function ARRAY
 
 -= Info about function 'ARRAY' =-
 
 [Syntax]
 ARRAY(var1[|var2[...][|varN]])
 
 [Synopsis]
 Allows setting multiple variables at once
 
 [Description]
 The comma-separated list passed as a value to which the function is set
 will be interpreted as a set of values to which the comma-separated list
 of variable names in the argument should be set.
 Hence, Set(ARRAY(var1|var2)=1\,2) will set var1 to 1 and var2 to 2
 Note: remember to either backslash your commas in extensions.conf or quote
 the entire argument, since Set can take multiple arguments itself.
 
 So probably with 1.4 will come.
 
 Benchev
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Im aware of ARRAY and would have used it but it only sets variables - you
cant reference the array by name or its elements as is common in other
programming languages.  

Would it be possible to use ASTDB and the while application?  For instance
the user inputs 3 numbers to be used for Follow me and the dial plan
iterates thru them checking for CF CW etc etc changing the numbers if needs
be then dialling.  

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RE: [asterisk-users] Limitations of IAX

[Douglas Garstang]

 -Original Message-
 From: Joshua Colp [mailto:[EMAIL PROTECTED]
 Sent: Wednesday, August 02, 2006 7:42 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Limitations of IAX
 
 
 - Original Message -
 From: Douglas Garstang
 [mailto:[EMAIL PROTECTED]
 To: Asterisk Users Mailing List -
 Non-Commercial Discussion [mailto:[EMAIL PROTECTED]
 Sent:
 Wed, 02 Aug 2006 13:50:56 -0300
 Subject: [asterisk-users] Limitations of
 IAX
 
 
  I'm about ready to give up on IAX2. It seems to have some SERIOUS
  limitations.
 
 You may have run into these limitations in your deployment 
 but others are using IAX2 fine for what it was designed to 
 do. No software out there can anticipate everyone's needs. 
 Those who do run into issues end up working around the 
 limitations or modifying it to their needs. You will probably 
 need to do the same or seek an alternate solution.
How many CLEC's are you aware of that are using Asterisk to provide not just 
enterprise features, but carrier grade features?

 
  Incoming PSTN call comes into to user A on pbx1. We look 
 for user A locally,
  and don't find them. We then do a DUNDi lookup, get a path, 
 and dial user A
  on pbx2 with IAX2. User A picks up the call.
  
  When IAX passes the call from pbx1 to pbx2, it does not 
 pass some of the
  dial plan variables. Namely dnid, which is set to 
 'unknown'. Consequently
  (for reasons I don't yet understand) when user A on pbx2 
 tries to do a blind
  transfer, the dnid is not set, and as a result, we _must_ 
 fail the call
  because we have no source number on our network. 
 
 Have you reported a bug on this DNID not being passed?
No... last time I opened a bug I got told it wasn't a bug, the bug was closed, 
and I was given bad feedback.

  
  Also, the account code is not picked up on pbx2 as well.
 
 As previously pointed out transporting the account code may 
 be a security risk as well. If you really need it why don't 
 you encode it into the dialed number or something?
Yeah well, I know if the account code isn't passed, we can't bill the call. How 
is passing the account code in the dialled string any less insecure than 
passing it somewhere else in the IAX protocol?
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Re: [asterisk-users] canreinvite=yes and RTP dropping in and out

[Joshua Colp]

- Original Message -
From: Gary Richardson
[mailto:[EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Non-Commercial Discussion [mailto:[EMAIL PROTECTED]
Sent:
Wed, 02 Aug 2006 14:34:31 -0300
Subject: Re: [asterisk-users]
canreinvite=yes and RTP dropping in and out


 My next attempt at this is going to be putting a hub in between the path to
 the switch. I'm hoping to be able to sniff the packets to see what's going
 on.
 
 Also, using the network status page on the hard phones, the transmit and
 receive counters for the direction of the channel slows way down as if
 almost no data is being transmitted.
 
 How do I send a sip debug?

Actually since this happens randomly I doubt that will help. Is there any other 
traffic on the network too? Never know... or a faulty switch? Grasping at 
random things but nothing really comes to mind.
 
 Thanks.

Joshua Colp
Digium
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Re: [asterisk-users] Limitations of IAX

[Andrew Kohlsmith]

On Wednesday 02 August 2006 09:42, Joshua Colp wrote:
 As previously pointed out transporting the account code may be a security
 risk as well. If you really need it why don't you encode it into the dialed
 number or something?

Because that's a completely shitty workaround for something that may be a 
security risk.  DISA may be a security risk too but it's there in all its 
glory; why not allow people to shoot themselves in the foot?  Why not have a 
restrictvars=yes in iax.conf?  If I create a patch to do this against svn 
trunk, would such a thing be accepted?

Mr. Garstang is very grating at times but he has brought forward a number of 
shortcomings which if fixed would make the use of Asterisk far more 
intuitive.  And as I am certain you have seen on this very list, some of the 
workarounds people have done in order to make things work are hideous, 
hideous abominations.

-A.
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[asterisk-users] IAX Trunking

[Douglas Garstang]

Ok... it'd be great if someone could explain this to me...

User A on pbx1 wants to dial User B on pbx2. We do a local lookup and don't 
find user B on pbx1, so we do a DUNDi lookup of user B, get a result, and place 
the call to user B on pbx2 with IAX2.

When pbx2 calls the AGI script that dialls user B on pbx2, Asterisk passes a 
'type' of IAX2, eventhough the endpoint for user B is a SIP phone. Why? If user 
B transfers or forwards calls, and Asterisk re-enters the dialplan, and 
subsequently calls the AGI script again, it's still passing a type of IAX2 to 
the script, eventhough, like before this is a SIP call.

This may be part of why I am having a problem with variables. Even when user B 
transfers a call, user B is registered to pbx2, so a new SIP call should be 
initiated with an accountcode set. Because it's IAX, there's no accountcode! 
Why?

It's as if the IAX trunk is overriding or controlling all the calls to/from SIP 
endpoints, which is completely crazy.

Doug.



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RE: [asterisk-users] cmd DIAL - Who picked up the call?

[Koopmann, Jan-Peter]

On Wednesday, August 02, 2006 7:39 PM Vadim Berezniker wrote:

 DIALEDPEERNUMBER contains the exact peer spec for the peer that
 picked up. You can use that.

Consider yourself my hero of the day! That looks VERY promising. It does not 
show the technology so

Dial(SIP/phone_200Zap/g2/13,,M(getchannel))

will return either phone_200 or g2/13 but I will find a solution for that as 
well I suppose!!! Any idea why BRIDGEPEER is empty here all the time?


Kind regards,
  JP


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RE: [asterisk-users] Limitations of IAX

[Joshua Colp]

- Original Message -
From: Douglas Garstang
[mailto:[EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Non-Commercial Discussion [mailto:[EMAIL PROTECTED]
Sent:
Wed, 02 Aug 2006 14:51:10 -0300
Subject: RE: [asterisk-users] Limitations of
IAX


 How many CLEC's are you aware of that are using Asterisk to provide not just
 enterprise features, but carrier grade features?

I don't keep track of everyone who uses Asterisk or what they use it for. I'm 
merely saying that if you run into an issue and you need it to behave a certain 
way then you either have to: work around it, add the ability to do what you 
want, do it differently, or give up and use something different.
 
 No... last time I opened a bug I got told it wasn't a bug, the bug was
 closed, and I was given bad feedback.

This doesn't mean you can't file another bug. I'm extremely easy when it comes 
to people who file bugs who genuinely think it's a bug in Asterisk when it's 
sometimes a configuration issue or just the way it works. Just because you had 
one bad experience doesn't mean you can't go back.
 
 Yeah well, I know if the account code isn't passed, we can't bill the call.
 How is passing the account code in the dialled string any less insecure than
 passing it somewhere else in the IAX protocol?

Writing it directly into the protocol is the dangerous part. You would need 
some control over who can set it and under what circumstances it is allowed. By 
encoding it into the dialed number only the users you control will be able to 
do it and the protocol doesn't need to be altered.

Joshua Colp
Digium
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RE: [asterisk-users] Limitations of IAX

[Douglas Garstang]

 -Original Message-
 From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED]
 Sent: Wednesday, August 02, 2006 11:54 AM
 To: asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] Limitations of IAX
 
 
 On Wednesday 02 August 2006 09:42, Joshua Colp wrote:
  As previously pointed out transporting the account code may 
 be a security
  risk as well. If you really need it why don't you encode it 
 into the dialed
  number or something?
 
 Because that's a completely shitty workaround for something 
 that may be a 
 security risk.  DISA may be a security risk too but it's 
 there in all its 
 glory; why not allow people to shoot themselves in the foot?  
 Why not have a 
 restrictvars=yes in iax.conf?  If I create a patch to do 
 this against svn 
 trunk, would such a thing be accepted?
 
 Mr. Garstang is very grating at times but he has brought 
 forward a number of 
 shortcomings which if fixed would make the use of Asterisk far more 
 intuitive.  And as I am certain you have seen on this very 
 list, some of the 
 workarounds people have done in order to make things work are 
 hideous, 
 hideous abominations.

Thankyou, Mr Kolhsmith (I think...?)

A large part of my frustration with Asterisk is that, without intentionally 
sounding conceited, we are pushing the evelope on what it can do. Most Asterisk 
users are implementing it in an enterprise environment. We are 'trying' to 
implement it in a carrier environment, ie providing a hosted IPT service to 
enterprises, which means that the features and functionality required make my 
head spin sometimes, and push Asterisk beyond what it was designed to do.

Doug.
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RE: [asterisk-users] RemoveQueueMember isn't working.

[Keith Herrington]

Joshua,
Thank you!! I didn't even notice that. I'll fix it and report the bug to
FreePBX.

Keith

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joshua
Colp
Sent: Monday, July 31, 2006 11:13 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] RemoveQueueMember isn't working.

- Original Message -
From: Keith Herrington
[mailto:[EMAIL PROTECTED]
To:
asterisk-users@lists.digium.com
Sent: Mon, 31 Jul 2006 18:51:55
-0300
Subject: [asterisk-users] RemoveQueueMember isn't working.


 Hey guys. I've ran into a queue issue that I'm wondering if anyone has

 seen.
 I am using FreePBX with Asterisk 1.2.9.1 svn rev 34876.
  
 I have setup a queue of 1082. I login to the Queue fine with 1082*, 
 and receive calls via the queue. When I try to logout with 1082**, 
 Alison says I was removed, but when I do a 'show queue 1082' I'm still
listed.
 If I use the CLI and go 'remove queue member Local/1409 from 1082' it 
 works as expected.
 
 Any ideas?
  
 This is CLI output: 
 - Goto (macro-agent-del,s,5)
 -- Executing Set(SIP/1409-60bb, CALLBACKNUM=1409) in new stack
 -- Executing GotoIf(SIP/1409-60bb, 0?2)) in new stack
 -- Executing RemoveQueueMember(SIP/1409-60bb,
 1082|Local/[EMAIL PROTECTED]/n) in new stack
 -- Executing UserEvent(SIP/1409-60bb, RefreshQueue) in new stack
 -- Executing Wait(SIP/1409-60bb, 1) in new stack
 -- Executing Playback(SIP/1409-60bb, agent-loggedoff) in new stack
 -- Playing 'agent-loggedoff' (language 'en')
 -- Executing Hangup(SIP/1409-60bb, ) in new stack asterisk1*CLI 
 show queues
 1082 has 0 calls (max unlimited) in 'ringall' strategy (0s holdtime), 
 W:0, C:0, A:0, SL:0.0% within 0s
 Members: 
 Local/[EMAIL PROTECTED] (dynamic) (Unknown) has taken no calls yet No

 Callers
  
  asterisk1*CLI remove queue member Local/[EMAIL PROTECTED] from 1082

 Removed interface 'Local/[EMAIL PROTECTED]' from queue '1082'
 asterisk1*CLI show queue 1082
 1082 has 0 calls (max unlimited) in 'ringall' strategy (0s holdtime), 
 W:0, C:0, A:0, SL:0.0% within 0s No Members No Callers
 
 Here's a pastebin to my queues.conf and my extensions.conf sections:
 http://www.pastecode.com/2334
 
 Thanks in advance!
 
 Keith

You are removing a queue member that isn't in the queue ;)

You are adding Local/[EMAIL PROTECTED] to the queue while you
are removing Local/[EMAIL PROTECTED]/n

If you remove the /n it should then work.

Joshua Colp
Digium
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RE: [asterisk-users] cmd DIAL - Who picked up the call?

[Vadim Berezniker]

No idea, but DIALEDPEERNAME should contain the same value as BRIDGEPEER. Try 
that.
The only difference is that BRIDGEPEER is set slightly later (when the call is 
bridged). 



From: [EMAIL PROTECTED] on behalf of Koopmann, Jan-Peter
Sent: Wed 8/2/2006 2:05 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] cmd DIAL - Who picked up the call?



On Wednesday, August 02, 2006 7:39 PM Vadim Berezniker wrote:

 DIALEDPEERNUMBER contains the exact peer spec for the peer that
 picked up. You can use that.

Consider yourself my hero of the day! That looks VERY promising. It does not 
show the technology so

Dial(SIP/phone_200Zap/g2/13,,M(getchannel))

will return either phone_200 or g2/13 but I will find a solution for that as 
well I suppose!!! Any idea why BRIDGEPEER is empty here all the time?


Kind regards,
  JP


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[asterisk-users] creidt card processing sripts for asterisk

[Joseph]

Are there any credit card processing scripts for asterisk, that would
allow me to enter credit card number amount and dial my IVR system?

-- 
#Joseph
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Re: [asterisk-users] Limitations of IAX

[Andrew Kohlsmith]

On Wednesday 02 August 2006 10:29, Joshua Colp wrote:
 Writing it directly into the protocol is the dangerous part. You would need
 some control over who can set it and under what circumstances it is
 allowed. By encoding it into the dialed number only the users you control
 will be able to do it and the protocol doesn't need to be altered.

Nonsense; treat it like you'd treat Caller*ID; if you don't want to pass it 
over, explicitly zero it.   In fact, this is just plain old good programming 
common sense; do not assume anything implicitly; if you want a variable at a 
certain value (or not to have a certain value) then explicitly set it to that 
value.

This isn't hyperoptimized SSE2 does-a-billion-cycles superfast tight loop 
stuff; if a var gets set twice... oh well!  The peace of mind it gives is 
worth its weight in L2 cache transistors, and if you actually NEED this data 
transfered over... well it's there for you to use in all its splendor... no 
need to resort to ass-backward hacks like encoding things in the Dial() 
string and trying to parse it out on the other side, all the while worrying 
about escaping awkward values and trying to do so in a programming language 
slightly less functional than BASIC.

... I'm not attacking you, Joshua... and honestly if a patch giving this kind 
of function (with the ability to turn it off) will be accepted into trunk...  
I'll be all over this...  but generally what ends up happening is that no 
solution is accepted and the status quo is left in place, along with all of 
its problems.  I'm trying to help out, just as you are.

-A.
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Re: [asterisk-users] creidt card processing sripts for asterisk

[Jon Pounder]

Quoting Joseph [EMAIL PROTECTED]:


Are there any credit card processing scripts for asterisk, that would
allow me to enter credit card number amount and dial my IVR system?


have a look at www.opayc.com - while not specifically for asterisk, these
drivers use odbc (on unix or windows) to talk to a variety of payment 
gateways.

They are useful where the front end of the system is not a webpage such as
asterisk.




--
#Joseph
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Jon Pounder



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


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[asterisk-users] DUNDi with SIP

[Douglas Garstang]

I've trying to use DUNDi with SIP to see if it works around some limitations of 
IAX2.

I do a DUNDi lookup, get my SIP path, and try to dial it. Asterisk immediately 
says 'No such host', eventhough that's the path is just returned!

[Aug  2 13:07:05]   == Spawn extension (global_vmdeposit, u9220371, 3) exited 
non-zero on 'SIP/3254101-eb7d'
[Aug  2 13:07:13] -- Executing NoOp(SIP/3254101-6373, *** OnNet 
originated call Chocolate Chip 3254101 - 9220371) in new stack
[Aug  2 13:07:13] -- Executing AGI(SIP/3254101-6373, ipt/originator.py) 
in new stack
[Aug  2 13:07:13] -- Launched AGI Script 
/var/lib/asterisk/agi-bin/ipt/originator.py
[Aug  2 13:07:13] -- AGI Script Executing Application: (SetAccount) 
Options: (9220371)
[Aug  2 13:07:13] -- AGI Script Executing Application: (ChanIsAvail) 
Options: (SIP/9220371)
[Aug  2 13:07:14] -- AGI Script Executing Application: (Dial) Options: 
(SIP/dundisip:[EMAIL PROTECTED]/9220371)
[Aug  2 13:07:14] WARNING[5429]: chan_sip.c:1980 create_addr: No such host: 
xxx.yyy.142.163/9220371
[Aug  2 13:07:14] NOTICE[5429]: app_dial.c:1040 dial_exec_full: Unable to 
create channel of type 'SIP' (cause 3 - No route to destination)
[Aug  2 13:07:14]   == Everyone is busy/congested at this time (1:0/0/1)

Not sure what is going on. I can see the query at the other end, but it doesn't 
look like it ever receives the call.

Doug.


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Re: [asterisk-users] creidt card processing sripts for asterisk

[Joseph]

I don't need a gateway; I was looking to find a script what would let me
dial into our IVR system, provide merchant number + device number +
credit card + exp. date + amount
Merchant #, device # are constant so it can be build into the script;
credit card #, exp. date and amount are variable so it could be pulled
out of the database or a file.

-- 
#Joseph

On Wed, 2006-08-02 at 15:03 -0400, Jon Pounder wrote:
 Quoting Joseph [EMAIL PROTECTED]:
 
  Are there any credit card processing scripts for asterisk, that would
  allow me to enter credit card number amount and dial my IVR system?
 
 have a look at www.opayc.com - while not specifically for asterisk, these
 drivers use odbc (on unix or windows) to talk to a variety of payment 
 gateways.
 They are useful where the front end of the system is not a webpage such as
 asterisk.


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Re: [asterisk-users] DUNDi with SIP

[Aaron Daniel]

Using the SECRET variable for sip doesn't work.

On Wed, 2006-08-02 at 13:11 -0600, Douglas Garstang wrote:
 I've trying to use DUNDi with SIP to see if it works around some limitations 
 of IAX2.
 
 I do a DUNDi lookup, get my SIP path, and try to dial it. Asterisk 
 immediately says 'No such host', eventhough that's the path is just returned!
 
 [Aug  2 13:07:05]   == Spawn extension (global_vmdeposit, u9220371, 3) exited 
 non-zero on 'SIP/3254101-eb7d'
 [Aug  2 13:07:13] -- Executing NoOp(SIP/3254101-6373, *** OnNet 
 originated call Chocolate Chip 3254101 - 9220371) in new stack
 [Aug  2 13:07:13] -- Executing AGI(SIP/3254101-6373, 
 ipt/originator.py) in new stack
 [Aug  2 13:07:13] -- Launched AGI Script 
 /var/lib/asterisk/agi-bin/ipt/originator.py
 [Aug  2 13:07:13] -- AGI Script Executing Application: (SetAccount) 
 Options: (9220371)
 [Aug  2 13:07:13] -- AGI Script Executing Application: (ChanIsAvail) 
 Options: (SIP/9220371)
 [Aug  2 13:07:14] -- AGI Script Executing Application: (Dial) Options: 
 (SIP/dundisip:[EMAIL PROTECTED]/9220371)
 [Aug  2 13:07:14] WARNING[5429]: chan_sip.c:1980 create_addr: No such host: 
 xxx.yyy.142.163/9220371
 [Aug  2 13:07:14] NOTICE[5429]: app_dial.c:1040 dial_exec_full: Unable to 
 create channel of type 'SIP' (cause 3 - No route to destination)
 [Aug  2 13:07:14]   == Everyone is busy/congested at this time (1:0/0/1)
 
 Not sure what is going on. I can see the query at the other end, but it 
 doesn't look like it ever receives the call.
 
 Doug.
 
 
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-- 
Aaron Daniel
Computer Systems Technician
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198
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RE: [asterisk-users] Asterisk with VoIP phone

[shadowym]

You need to carefully consider outside VoIPproviders 
IMHO. I would look for providers who are very upfront about their network 
architecture and how they connect to the PSTN (the public telephone 
network). As a minimum, I would ask for IP addresses to some of their SIP 
servers and check ping times. I would look for consistent ping times at 
different times of the day with round trips below 50ms ideally but definitely no 
more than 100ms. A lot of this depends on geography. It is also 
important to know how they connect to the public phone network IMHO. 
Ideally, they have direct "TDM" connections (digital telephone connections) as 
opposed to IP connections to who knows where. 

There are other considerations as well but I think that is 
a good start IMHO.

  
  
  From: Bruce Reeves 
  [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 02, 
  2006 4:58 AMTo: Asterisk Users Mailing List - Non-Commercial 
  DiscussionSubject: Re: [asterisk-users] Asterisk with VoIP 
  phone
  Yes this is possible, you just setup the softphones and maybe the 
  provider in sip.conf and write your dialplan :)
  On 8/1/06, J Rangi 
   [EMAIL PROTECTED] 
wrote:
  Hello,Is 
is possible to setup an asterisk server with out buying Digium card. I 
mean can we do this type of setup.We all know that X-Lite can be used as 
a soft phone to have an IPextension.Is it possible to take a service 
from another VoIP service provider, andget the IP phone number. Make 
that phone numbe gateway to outside world. Now all the internal 
extensions use that phone to receive and make callsto out side 
world.Has any one done this kind of setup or know anything about 
this.Thank 
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RE: [asterisk-users] VOIP phone for Receptionist use

[shadowym]

Join the club,

I would like shared line appearance ability along with MANY 
other people too. That feature is only now on the rader screen for 
Asterisk but it probably won't be available for some time. Months if not 
years (probably not till v1.6 which is after the next version which will be 
1.4).

There are alternatives which some people here may 
suggest. Unfortunately they require a change in usage and thinking by the 
receptionist. You would definitely want to be upfront and give some sort of demo 
beforehand IMHO to avoid problems with the client 
relationship.

My 2 cents.

  
  
  From: Jeff Busch 
  [mailto:[EMAIL PROTECTED] Sent: Tuesday, August 01, 2006 
  5:20 PMTo: asterisk-users@lists.digium.comSubject: 
  [asterisk-users] VOIP phone for Receptionist use
  
  I've searched 
  through the newsgroup and online and haven't found an answer for my 
  question... maybe I am looking for the wrong terms, I am not 
  sure...
  
  I have a client 
  that would like a phone that is like a "typical" receptionists 
  phone.
  
  Requirements:
  - Ability for 
  their3 lines to "light-up" a button on the phone when one of them rings 
  in.
  - Ability for the 
  phone to ring when the receptionist is on one call and a second or third call 
  is incoming. (this has been the biggest frustration up to now. 
  When a second call comes, there is no tone that heard on the IP500. 
  Perhaps I am missing a setting?)
  
  We are currently 
  using:
  
  Asterisk @ Home 
  2.1
  Polycom IP500/501 
  phones
  
  Is there a way to 
  do what we need to using the IP500 phones? If so, can anyone give me 
  instructions on how to make it work with [EMAIL PROTECTED]?
  
  Thanks for your 
  help in advance.
  
  Jeff
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Re: [asterisk-users] creidt card processing sripts for asterisk

[Jon Pounder]

Quoting Joseph [EMAIL PROTECTED]:


I don't need a gateway; I was looking to find a script what would let me
dial into our IVR system, provide merchant number + device number +
credit card + exp. date + amount
Merchant #, device # are constant so it can be build into the script;
credit card #, exp. date and amount are variable so it could be pulled
out of the database or a file.



you need to make it clearer what your actual application is. do you mean speak
that information to the caller, collect and store it from the caller ? other ?





--
#Joseph

On Wed, 2006-08-02 at 15:03 -0400, Jon Pounder wrote:

Quoting Joseph [EMAIL PROTECTED]:

 Are there any credit card processing scripts for asterisk, that would
 allow me to enter credit card number amount and dial my IVR system?

have a look at www.opayc.com - while not specifically for asterisk, these
drivers use odbc (on unix or windows) to talk to a variety of payment
gateways.
They are useful where the front end of the system is not a webpage such as
asterisk.



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Jon Pounder

  _/_/_/  _/_/  _/   _/_/_/  _/_/  _/_/_/_/
   _/_/_/  _/  _/ _/_/_/  _/  _/_/
  _/_/  _/_/  _/ _/_/  _/_/  _/
_/_/_/  _/_/  _/_/_/_/ _/_/_/  _/_/  _/_/_/_/


Inline Internet Systems Inc.
Thorold, Ontario, Canada

Tools to Power Your e-Business Solutions
www.inline.net
www.ihtml.com
www.ihtmlmerchant.com
www.opayc.com


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


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Re: [asterisk-users] creidt card processing sripts for asterisk

[Mojo with Horan Company, LLC]

I think what the poster meant was that this script would conceivably 
need to be designed to work with the API of a SPECIFIC gateway and might 
not be easily generic.


Joseph wrote:

I don't need a gateway; I was looking to find a script what would let me
dial into our IVR system, provide merchant number + device number +
credit card + exp. date + amount
Merchant #, device # are constant so it can be build into the script;
credit card #, exp. date and amount are variable so it could be pulled
out of the database or a file.



--
Mojo [EMAIL PROTECTED]
Office Manager, Horan  Company, LLC
(907) 747- x112
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RE: [asterisk-users] DUNDi with SIP

[Douglas Garstang]

Secret? Do you mean sbsecret in sip.conf?

 -Original Message-
 From: Aaron Daniel [mailto:[EMAIL PROTECTED]
 Sent: Wednesday, August 02, 2006 1:33 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] DUNDi with SIP
 
 
 Using the SECRET variable for sip doesn't work.
 
 On Wed, 2006-08-02 at 13:11 -0600, Douglas Garstang wrote:
  I've trying to use DUNDi with SIP to see if it works around 
 some limitations of IAX2.
  
  I do a DUNDi lookup, get my SIP path, and try to dial it. 
 Asterisk immediately says 'No such host', eventhough that's 
 the path is just returned!
  
  [Aug  2 13:07:05]   == Spawn extension (global_vmdeposit, 
 u9220371, 3) exited non-zero on 'SIP/3254101-eb7d'
  [Aug  2 13:07:13] -- Executing NoOp(SIP/3254101-6373, 
 *** OnNet originated call Chocolate Chip 3254101 - 
 9220371) in new stack
  [Aug  2 13:07:13] -- Executing AGI(SIP/3254101-6373, 
 ipt/originator.py) in new stack
  [Aug  2 13:07:13] -- Launched AGI Script 
 /var/lib/asterisk/agi-bin/ipt/originator.py
  [Aug  2 13:07:13] -- AGI Script Executing Application: 
 (SetAccount) Options: (9220371)
  [Aug  2 13:07:13] -- AGI Script Executing Application: 
 (ChanIsAvail) Options: (SIP/9220371)
  [Aug  2 13:07:14] -- AGI Script Executing Application: 
 (Dial) Options: 
 (SIP/dundisip:[EMAIL PROTECTED]/9220371)
  [Aug  2 13:07:14] WARNING[5429]: chan_sip.c:1980 
 create_addr: No such host: xxx.yyy.142.163/9220371
  [Aug  2 13:07:14] NOTICE[5429]: app_dial.c:1040 
 dial_exec_full: Unable to create channel of type 'SIP' (cause 
 3 - No route to destination)
  [Aug  2 13:07:14]   == Everyone is busy/congested at this 
 time (1:0/0/1)
  
  Not sure what is going on. I can see the query at the other 
 end, but it doesn't look like it ever receives the call.
  
  Doug.
  
  
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Re: [asterisk-users] VOIP phone for Receptionist use

[Dr. Michael J. Chudobiak]

- Ability for the phone to ring when the receptionist is on one call
and a second or third call is incoming.  (this has been the biggest
frustration up to now.  When a second call comes, there is no tone
that heard on the IP500.  Perhaps I am missing a setting?)


The Snom 360 can certainly do this - you can have a muted ringer, or 
just visual indication, or you can turn it off entirely.


- Mike

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[asterisk-users] Re: DUNDi with SIP

[Tony Mountifield]

In article [EMAIL PROTECTED],
Douglas Garstang [EMAIL PROTECTED] wrote:
 I've trying to use DUNDi with SIP to see if it works around some limitations 
 of IAX2.
 
 I do a DUNDi lookup, get my SIP path, and try to dial it. Asterisk 
 immediately says 'No such
 host', eventhough that's the path is just returned!
 
 [Aug  2 13:07:05]   == Spawn extension (global_vmdeposit, u9220371, 3) exited 
 non-zero on
 'SIP/3254101-eb7d'
 [Aug  2 13:07:13] -- Executing NoOp(SIP/3254101-6373, *** OnNet 
 originated call
 Chocolate Chip 3254101 - 9220371) in new stack
 [Aug  2 13:07:13] -- Executing AGI(SIP/3254101-6373, 
 ipt/originator.py) in new stack
 [Aug  2 13:07:13] -- Launched AGI Script 
 /var/lib/asterisk/agi-bin/ipt/originator.py
 [Aug  2 13:07:13] -- AGI Script Executing Application: (SetAccount) 
 Options: (9220371)
 [Aug  2 13:07:13] -- AGI Script Executing Application: (ChanIsAvail) 
 Options: (SIP/9220371)
 [Aug  2 13:07:14] -- AGI Script Executing Application: (Dial) Options:
 (SIP/dundisip:[EMAIL PROTECTED]/9220371)
 [Aug  2 13:07:14] WARNING[5429]: chan_sip.c:1980 create_addr: No such host: 
 xxx.yyy.142.163/9220371

Try specifying the SIP argument as:

SIP/dundisip:[EMAIL PROTECTED]@xxx.yyy.142.163

See the following line in the sample extensions.conf as an example:

;exten = _42X.,1,Dial(SIP/user:[EMAIL PROTECTED]:[EMAIL PROTECTED],30,rT)

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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RE: [asterisk-users] DUNDi with SIP

[Aaron Daniel]

I'm talking about the rotating DUNDi secret that is stored in dbsecret
in iax.conf.  It doesn't exist in the SIP channel.

On Wed, 2006-08-02 at 13:43 -0600, Douglas Garstang wrote:
 Secret? Do you mean sbsecret in sip.conf?
 
  -Original Message-
  From: Aaron Daniel [mailto:[EMAIL PROTECTED]
  Sent: Wednesday, August 02, 2006 1:33 PM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: Re: [asterisk-users] DUNDi with SIP
  
  
  Using the SECRET variable for sip doesn't work.
  
  On Wed, 2006-08-02 at 13:11 -0600, Douglas Garstang wrote:
   I've trying to use DUNDi with SIP to see if it works around 
  some limitations of IAX2.
   
   I do a DUNDi lookup, get my SIP path, and try to dial it. 
  Asterisk immediately says 'No such host', eventhough that's 
  the path is just returned!
   
   [Aug  2 13:07:05]   == Spawn extension (global_vmdeposit, 
  u9220371, 3) exited non-zero on 'SIP/3254101-eb7d'
   [Aug  2 13:07:13] -- Executing NoOp(SIP/3254101-6373, 
  *** OnNet originated call Chocolate Chip 3254101 - 
  9220371) in new stack
   [Aug  2 13:07:13] -- Executing AGI(SIP/3254101-6373, 
  ipt/originator.py) in new stack
   [Aug  2 13:07:13] -- Launched AGI Script 
  /var/lib/asterisk/agi-bin/ipt/originator.py
   [Aug  2 13:07:13] -- AGI Script Executing Application: 
  (SetAccount) Options: (9220371)
   [Aug  2 13:07:13] -- AGI Script Executing Application: 
  (ChanIsAvail) Options: (SIP/9220371)
   [Aug  2 13:07:14] -- AGI Script Executing Application: 
  (Dial) Options: 
  (SIP/dundisip:[EMAIL PROTECTED]/9220371)
   [Aug  2 13:07:14] WARNING[5429]: chan_sip.c:1980 
  create_addr: No such host: xxx.yyy.142.163/9220371
   [Aug  2 13:07:14] NOTICE[5429]: app_dial.c:1040 
  dial_exec_full: Unable to create channel of type 'SIP' (cause 
  3 - No route to destination)
   [Aug  2 13:07:14]   == Everyone is busy/congested at this 
  time (1:0/0/1)
   
   Not sure what is going on. I can see the query at the other 
  end, but it doesn't look like it ever receives the call.
   
   Doug.
   
   
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  Aaron Daniel
  Computer Systems Technician
  Sam Houston State University
  [EMAIL PROTECTED]
  (936) 294-4198
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Aaron Daniel
Computer Systems Technician
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198
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[asterisk-users] unsubscribe

[Keith Herrington]

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RE: [asterisk-users] Re: DUNDi with SIP

[Douglas Garstang]

 -Original Message-
 From: Tony Mountifield [mailto:[EMAIL PROTECTED]
 Sent: Wednesday, August 02, 2006 2:01 PM
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] Re: DUNDi with SIP
 
 
 In article 
 [EMAIL PROTECTED],
 Douglas Garstang [EMAIL PROTECTED] wrote:
  I've trying to use DUNDi with SIP to see if it works around 
 some limitations of IAX2.
  
  I do a DUNDi lookup, get my SIP path, and try to dial it. 
 Asterisk immediately says 'No such
  host', eventhough that's the path is just returned!
  
  [Aug  2 13:07:05]   == Spawn extension (global_vmdeposit, 
 u9220371, 3) exited non-zero on
  'SIP/3254101-eb7d'
  [Aug  2 13:07:13] -- Executing NoOp(SIP/3254101-6373, 
 *** OnNet originated call
  Chocolate Chip 3254101 - 9220371) in new stack
  [Aug  2 13:07:13] -- Executing AGI(SIP/3254101-6373, 
 ipt/originator.py) in new stack
  [Aug  2 13:07:13] -- Launched AGI Script 
 /var/lib/asterisk/agi-bin/ipt/originator.py
  [Aug  2 13:07:13] -- AGI Script Executing Application: 
 (SetAccount) Options: (9220371)
  [Aug  2 13:07:13] -- AGI Script Executing Application: 
 (ChanIsAvail) Options: (SIP/9220371)
  [Aug  2 13:07:14] -- AGI Script Executing Application: 
 (Dial) Options:
  (SIP/dundisip:[EMAIL PROTECTED]/9220371)
  [Aug  2 13:07:14] WARNING[5429]: chan_sip.c:1980 
 create_addr: No such host: xxx.yyy.142.163/9220371
 
 Try specifying the SIP argument as:
 
 SIP/dundisip:[EMAIL PROTECTED]@xxx.yyy.142.163
 
 See the following line in the sample extensions.conf as an example:
 
 ;exten = 
 _42X.,1,Dial(SIP/user:[EMAIL PROTECTED]:[EMAIL PROTECTED],30,rT)

Tony... it's DUNDi
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RE: [asterisk-users] DUNDi with SIP

[Douglas Garstang]

So what are the options? 

 -Original Message-
 From: Aaron Daniel [mailto:[EMAIL PROTECTED]
 Sent: Wednesday, August 02, 2006 2:03 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [asterisk-users] DUNDi with SIP
 
 
 I'm talking about the rotating DUNDi secret that is stored in dbsecret
 in iax.conf.  It doesn't exist in the SIP channel.
 
 On Wed, 2006-08-02 at 13:43 -0600, Douglas Garstang wrote:
  Secret? Do you mean sbsecret in sip.conf?
  
   -Original Message-
   From: Aaron Daniel [mailto:[EMAIL PROTECTED]
   Sent: Wednesday, August 02, 2006 1:33 PM
   To: Asterisk Users Mailing List - Non-Commercial Discussion
   Subject: Re: [asterisk-users] DUNDi with SIP
   
   
   Using the SECRET variable for sip doesn't work.
   
   On Wed, 2006-08-02 at 13:11 -0600, Douglas Garstang wrote:
I've trying to use DUNDi with SIP to see if it works around 
   some limitations of IAX2.

I do a DUNDi lookup, get my SIP path, and try to dial it. 
   Asterisk immediately says 'No such host', eventhough that's 
   the path is just returned!

[Aug  2 13:07:05]   == Spawn extension (global_vmdeposit, 
   u9220371, 3) exited non-zero on 'SIP/3254101-eb7d'
[Aug  2 13:07:13] -- Executing NoOp(SIP/3254101-6373, 
   *** OnNet originated call Chocolate Chip 3254101 - 
   9220371) in new stack
[Aug  2 13:07:13] -- Executing AGI(SIP/3254101-6373, 
   ipt/originator.py) in new stack
[Aug  2 13:07:13] -- Launched AGI Script 
   /var/lib/asterisk/agi-bin/ipt/originator.py
[Aug  2 13:07:13] -- AGI Script Executing Application: 
   (SetAccount) Options: (9220371)
[Aug  2 13:07:13] -- AGI Script Executing Application: 
   (ChanIsAvail) Options: (SIP/9220371)
[Aug  2 13:07:14] -- AGI Script Executing Application: 
   (Dial) Options: 
   (SIP/dundisip:[EMAIL PROTECTED]/9220371)
[Aug  2 13:07:14] WARNING[5429]: chan_sip.c:1980 
   create_addr: No such host: xxx.yyy.142.163/9220371
[Aug  2 13:07:14] NOTICE[5429]: app_dial.c:1040 
   dial_exec_full: Unable to create channel of type 'SIP' (cause 
   3 - No route to destination)
[Aug  2 13:07:14]   == Everyone is busy/congested at this 
   time (1:0/0/1)

Not sure what is going on. I can see the query at the other 
   end, but it doesn't look like it ever receives the call.

Doug.


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   Sam Houston State University
   [EMAIL PROTECTED]
   (936) 294-4198
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 -- 
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 Sam Houston State University
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[asterisk-users] Ateus Easy gate call progress

[Jan Fousek]

Hi all, 
 has anybody any experience with Ateus Easy Gate connected via Digium card to 
asterisk? It works fine for me except it doesn't pass the caller id and the 
hangup detection is quite slow. Are there some tips how to shorten the hangup 
delay?
Thanks.
Jan Fousek

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Re: [asterisk-users] DUNDi with SIP

[Alex Robar]

You can use an unchanging password. It's not as secure, but it will provide functionality.AlexOn 8/2/06, Douglas Garstang 
[EMAIL PROTECTED] wrote:So what are the options? -Original Message-
 From: Aaron Daniel [mailto:[EMAIL PROTECTED]] Sent: Wednesday, August 02, 2006 2:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] DUNDi with SIP
 I'm talking about the rotating DUNDi secret that is stored in dbsecret in iax.conf.It doesn't exist in the SIP channel. On Wed, 2006-08-02 at 13:43 -0600, Douglas Garstang wrote:
  Secret? Do you mean sbsecret in sip.conf?-Original Message-   From: Aaron Daniel [mailto:[EMAIL PROTECTED]]   Sent: Wednesday, August 02, 2006 1:33 PM
   To: Asterisk Users Mailing List - Non-Commercial Discussion   Subject: Re: [asterisk-users] DUNDi with SIP   Using the SECRET variable for sip doesn't work.
 On Wed, 2006-08-02 at 13:11 -0600, Douglas Garstang wrote:I've trying to use DUNDi with SIP to see if it works around   some limitations of IAX2.
   I do a DUNDi lookup, get my SIP path, and try to dial it.   Asterisk immediately says 'No such host', eventhough that's   the path is just returned!
   [Aug2 13:07:05] == Spawn extension (global_vmdeposit,   u9220371, 3) exited non-zero on 'SIP/3254101-eb7d'[Aug2 13:07:13] -- Executing NoOp(SIP/3254101-6373,
   *** OnNet originated call Chocolate Chip 3254101 -   9220371) in new stack[Aug2 13:07:13] -- Executing AGI(SIP/3254101-6373,
   ipt/originator.py) in new stack[Aug2 13:07:13] -- Launched AGI Script   /var/lib/asterisk/agi-bin/ipt/originator.py[Aug2 13:07:13] -- AGI Script Executing Application:
   (SetAccount) Options: (9220371)[Aug2 13:07:13] -- AGI Script Executing Application:   (ChanIsAvail) Options: (SIP/9220371)[Aug2 13:07:14] -- AGI Script Executing Application:
   (Dial) Options:   (SIP/dundisip:[EMAIL PROTECTED]/9220371)[Aug2 13:07:14] WARNING[5429]: chan_sip.c:1980   create_addr: No such host: 
xxx.yyy.142.163/9220371[Aug2 13:07:14] NOTICE[5429]: app_dial.c:1040   dial_exec_full: Unable to create channel of type 'SIP' (cause   3 - No route to destination)
[Aug2 13:07:14] == Everyone is busy/congested at this   time (1:0/0/1)   Not sure what is going on. I can see the query at the other
   end, but it doesn't look like it ever receives the call.   Doug.  ___
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   http://lists.digium.com/mailman/listinfo/asterisk-users   --   Aaron Daniel   Computer Systems Technician
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 http://lists.digium.com/mailman/listinfo/asterisk-users-- Alex Robar
[EMAIL PROTECTED]
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RE: [asterisk-users] DUNDi with SIP

[Douglas Garstang]

I've tried doing it without a username/password as described at:
http://www.voip-info.org/wiki/view/DUNDi+Enterprise+Configuration+SIP+with+no+passwords

but then authentication to the INVITE fails. I'm authenticating on the from: 
field, ie the individual user, which I don't think is right.

I've also tried it with this in dundi.conf:
180netsip = global_dundi_local,1,SIP,dundisip:[EMAIL 
PROTECTED]/${NUMBER},nopartial

and this in sip.conf:
[dundisip]
type=user
context=global_dundi_local
secret=password

and I still get the 'create_addr: No such host: xxx.yyy.142.163/9220371' 
messages on the client side.

Doug.



 -Original Message-
 From: Aaron Daniel [mailto:[EMAIL PROTECTED]
 Sent: Wednesday, August 02, 2006 2:03 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [asterisk-users] DUNDi with SIP
 
 
 I'm talking about the rotating DUNDi secret that is stored in dbsecret
 in iax.conf.  It doesn't exist in the SIP channel.
 
 On Wed, 2006-08-02 at 13:43 -0600, Douglas Garstang wrote:
  Secret? Do you mean sbsecret in sip.conf?
  
   -Original Message-
   From: Aaron Daniel [mailto:[EMAIL PROTECTED]
   Sent: Wednesday, August 02, 2006 1:33 PM
   To: Asterisk Users Mailing List - Non-Commercial Discussion
   Subject: Re: [asterisk-users] DUNDi with SIP
   
   
   Using the SECRET variable for sip doesn't work.
   
   On Wed, 2006-08-02 at 13:11 -0600, Douglas Garstang wrote:
I've trying to use DUNDi with SIP to see if it works around 
   some limitations of IAX2.

I do a DUNDi lookup, get my SIP path, and try to dial it. 
   Asterisk immediately says 'No such host', eventhough that's 
   the path is just returned!

[Aug  2 13:07:05]   == Spawn extension (global_vmdeposit, 
   u9220371, 3) exited non-zero on 'SIP/3254101-eb7d'
[Aug  2 13:07:13] -- Executing NoOp(SIP/3254101-6373, 
   *** OnNet originated call Chocolate Chip 3254101 - 
   9220371) in new stack
[Aug  2 13:07:13] -- Executing AGI(SIP/3254101-6373, 
   ipt/originator.py) in new stack
[Aug  2 13:07:13] -- Launched AGI Script 
   /var/lib/asterisk/agi-bin/ipt/originator.py
[Aug  2 13:07:13] -- AGI Script Executing Application: 
   (SetAccount) Options: (9220371)
[Aug  2 13:07:13] -- AGI Script Executing Application: 
   (ChanIsAvail) Options: (SIP/9220371)
[Aug  2 13:07:14] -- AGI Script Executing Application: 
   (Dial) Options: 
   (SIP/dundisip:[EMAIL PROTECTED]/9220371)
[Aug  2 13:07:14] WARNING[5429]: chan_sip.c:1980 
   create_addr: No such host: xxx.yyy.142.163/9220371
[Aug  2 13:07:14] NOTICE[5429]: app_dial.c:1040 
   dial_exec_full: Unable to create channel of type 'SIP' (cause 
   3 - No route to destination)
[Aug  2 13:07:14]   == Everyone is busy/congested at this 
   time (1:0/0/1)

Not sure what is going on. I can see the query at the other 
   end, but it doesn't look like it ever receives the call.

Doug.


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   -- 
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   Computer Systems Technician
   Sam Houston State University
   [EMAIL PROTECTED]
   (936) 294-4198
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 -- 
 Aaron Daniel
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 Sam Houston State University
 [EMAIL PROTECTED]
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RE: [asterisk-users] DUNDi with SIP

[Douglas Garstang]

Alex,

Thanks... I haven't had any luck with it yet.

My 
dundi.conf has:

180netsip = 
global_dundi_local,1,SIP,dundisip:[EMAIL PROTECTED]/${NUMBER},nopartial

and my 
sip.conf has:

[dundisip]type=usercontext=global_dundi_localsecret=password

A 
DUNDI lookup on the console returns a SIP path:

*CLI dundi lookup [EMAIL PROTECTED] 
1. 1 SIP/dundisip:[EMAIL PROTECTED]/9220370 
(EXISTS) from 00:14:22:1e:2a:d0, expires in 0 
sDUNDi lookup completed in 129 ms

However, when I try to connect, I get a 'No such host' 
error...

*CLI [Aug 2 14:18:43] -- Executing 
NoOp("SIP/3254101-a8d9", "*** OnNet originated call "Chocolate Chip" 
3254101 - 9220371") in new stack[Aug 2 
14:18:43] -- Executing AGI("SIP/3254101-a8d9", 
"ipt/originator.py") in new stack[Aug 2 
14:18:43] -- Launched AGI Script 
/var/lib/asterisk/agi-bin/ipt/originator.py[Aug 2 
14:18:43] -- AGI Script Executing Application: 
(SetAccount) Options: (9220371)[Aug 2 
14:18:43] -- AGI Script Executing Application: 
(ChanIsAvail) Options: (SIP/9220371)[Aug 2 
14:18:43] -- AGI Script Executing Application: (Dial) 
Options: (SIP/dundisip:[EMAIL PROTECTED]/9220371)[Aug 2 
14:18:43] WARNING[7842]: chan_sip.c:1980 create_addr: No such host: 
xxx.yyy.142.163/9220371[Aug 2 14:18:43] NOTICE[7842]: app_dial.c:1040 
dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to 
destination)[Aug 2 14:18:43] == Everyone is busy/congested 
at this time (1:0/0/1)

Doug.


  -Original Message-From: Alex Robar 
  [mailto:[EMAIL PROTECTED]Sent: Wednesday, August 02, 2006 2:17 
  PMTo: Asterisk Users Mailing List - Non-Commercial 
  DiscussionSubject: Re: [asterisk-users] DUNDi with 
  SIPYou can use an unchanging password. It's not as 
  secure, but it will provide functionality.Alex
  On 8/2/06, Douglas 
  Garstang  
  [EMAIL PROTECTED] wrote:
  So 
what are the options? -Original Message-  From: 
Aaron Daniel [mailto:[EMAIL PROTECTED]] Sent: 
Wednesday, August 02, 2006 2:03 PM To: Asterisk Users Mailing List - 
Non-Commercial Discussion Subject: RE: [asterisk-users] DUNDi with 
SIP  I'm talking about the rotating DUNDi secret 
that is stored in dbsecret in iax.conf.It doesn't exist 
in the SIP channel. On Wed, 2006-08-02 at 13:43 -0600, 
Douglas Garstang wrote:   Secret? Do you mean sbsecret in 
sip.conf?-Original Message- 
  From: Aaron Daniel [mailto:[EMAIL PROTECTED]]   Sent: 
Wednesday, August 02, 2006 1:33 PMTo: Asterisk Users 
Mailing List - Non-Commercial Discussion   Subject: Re: 
[asterisk-users] DUNDi with SIP 
  Using the SECRET variable for sip doesn't work.   
   On Wed, 2006-08-02 at 13:11 -0600, Douglas Garstang 
wrote:I've trying to use DUNDi with SIP to see if it 
works around   some limitations of IAX2.
I do a DUNDi lookup, get my SIP path, and try to 
dial it.   Asterisk immediately says 'No such host', 
eventhough that's   the path is just returned!   
 [Aug2 13:07:05] == 
Spawn extension (global_vmdeposit,   u9220371, 3) exited 
non-zero on 'SIP/3254101-eb7d'[Aug2 
13:07:13] -- Executing NoOp("SIP/3254101-6373", 
   "*** OnNet originated call "Chocolate Chip" 
3254101 -   9220371") in new stack  
  [Aug2 13:07:13] -- Executing 
AGI("SIP/3254101-6373","ipt/originator.py") in new 
stack[Aug2 
13:07:13] -- Launched AGI Script   
/var/lib/asterisk/agi-bin/ipt/originator.py
[Aug2 13:07:13] -- AGI Script Executing 
Application:(SetAccount) Options: (9220371)  
  [Aug2 13:07:13] -- AGI Script 
Executing Application:   (ChanIsAvail) Options: 
(SIP/9220371)[Aug2 
13:07:14] -- AGI Script Executing Application: 
   (Dial) Options:   
(SIP/dundisip:[EMAIL PROTECTED]/9220371)  
  [Aug2 13:07:14] WARNING[5429]: chan_sip.c:1980 
  create_addr: No such host: xxx.yyy.142.163/9220371  
  [Aug2 13:07:14] NOTICE[5429]: app_dial.c:1040 
  dial_exec_full: Unable to create channel of type 'SIP' 
(cause   3 - No route to destination)
 [Aug2 13:07:14] == Everyone is busy/congested 
at this   time (1:0/0/1) 
  Not sure what is going on. I can see the query at the 
other   end, but it doesn't look like it ever receives the 
call.   Doug.   
   
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Systems TechnicianSam Houston State University 
  [EMAIL PROTECTED] 
  (936) 294-4198   
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Re: [asterisk-users] creidt card processing sripts for asterisk

[Joseph]

On Wed, 2006-08-02 at 15:33 -0400, Jon Pounder wrote:
 Quoting Joseph [EMAIL PROTECTED]:
 
  I don't need a gateway; I was looking to find a script what would let me
  dial into our IVR system, provide merchant number + device number +
  credit card + exp. date + amount
  Merchant #, device # are constant so it can be build into the script;
  credit card #, exp. date and amount are variable so it could be pulled
  out of the database or a file.
 
 
 you need to make it clearer what your actual application is. do you mean speak
 that information to the caller, collect and store it from the caller ? other ?

I think I wasn't very clear.
Right now we process the credit card using our bank IVR system. Dial-In
and provide the information by pressing the number on a touch tone phone
especially credit card numbers and amount, all other information
(merchant number etc) are stored in a phone memory so are easy to
access.
I was playing with the dial plan trying to accomplish this in dial-plan
using all kind of delay ex:
;exten = _51,1,Dial(SIP/[EMAIL PROTECTED],30,D(w1))
but it didn't work.  
When we call our IVR there is a recording and I need to enter a delay
before entering other numbers. 
Maybe I'm approaching it wrong.

-- 
#Joseph
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Re: [asterisk-users] DUNDi with SIP

[Alex Robar]

Doug,Two things: If you try to place that call manually (either via dialling it from a phone that supports SP URIs or by making an ext. for it in your dialplan and calling that extension), does it work properly? Are you able to place the call? If not, is the CLI output the same as when you try it via DUNDi?
Second, are your keys generated properly, with public keys shared between the two boxes OK? I had a lot of DUNDi problems initially, and found that my keys were the problem.Alex
On 8/2/06, Douglas Garstang [EMAIL PROTECTED] wrote:







Alex,

Thanks... I haven't had any luck with it yet.

My 
dundi.conf has:

180netsip = 
global_dundi_local,1,SIP,dundisip:[EMAIL PROTECTED]/${NUMBER},nopartial

and my 
sip.conf has:

[dundisip]type=usercontext=global_dundi_localsecret=password

A 
DUNDI lookup on the console returns a SIP path:

*CLI dundi lookup [EMAIL PROTECTED] 
1. 1 SIP/dundisip:[EMAIL PROTECTED]/9220370 
(EXISTS) from 00:14:22:1e:2a:d0, expires in 0 
sDUNDi lookup completed in 129 ms

However, when I try to connect, I get a 'No such host' 
error...

*CLI [Aug 2 14:18:43] -- Executing 
NoOp(SIP/3254101-a8d9, *** OnNet originated call Chocolate Chip 
3254101 - 9220371) in new stack[Aug 2 
14:18:43] -- Executing AGI(SIP/3254101-a8d9, 
ipt/originator.py) in new stack[Aug 2 
14:18:43] -- Launched AGI Script 
/var/lib/asterisk/agi-bin/ipt/originator.py[Aug 2 
14:18:43] -- AGI Script Executing Application: 
(SetAccount) Options: (9220371)[Aug 2 
14:18:43] -- AGI Script Executing Application: 
(ChanIsAvail) Options: (SIP/9220371)[Aug 2 
14:18:43] -- AGI Script Executing Application: (Dial) 
Options: (SIP/dundisip:[EMAIL PROTECTED]/9220371)[Aug 2 
14:18:43] WARNING[7842]: chan_sip.c:1980 create_addr: No such host: 
xxx.yyy.142.163/9220371[Aug 2 14:18:43] NOTICE[7842]: app_dial.c:1040 
dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to 
destination)[Aug 2 14:18:43] == Everyone is busy/congested 
at this time (1:0/0/1)

Doug.


  -Original Message-From: Alex Robar 
  [mailto:[EMAIL PROTECTED]]Sent: Wednesday, August 02, 2006 2:17 
  PMTo: Asterisk Users Mailing List - Non-Commercial 
  DiscussionSubject: Re: [asterisk-users] DUNDi with 
  SIPYou can use an unchanging password. It's not as 
  secure, but it will provide functionality.Alex
  On 8/2/06, Douglas 
  Garstang  
  [EMAIL PROTECTED] wrote:
  So 
what are the options? -Original Message-  From: 
Aaron Daniel [mailto:[EMAIL PROTECTED]] Sent: 
Wednesday, August 02, 2006 2:03 PM To: Asterisk Users Mailing List - 
Non-Commercial Discussion Subject: RE: [asterisk-users] DUNDi with 
SIP  I'm talking about the rotating DUNDi secret 
that is stored in dbsecret in iax.conf.It doesn't exist 
in the SIP channel. On Wed, 2006-08-02 at 13:43 -0600, 
Douglas Garstang wrote:   Secret? Do you mean sbsecret in 
sip.conf?-Original Message- 
  From: Aaron Daniel [mailto:[EMAIL PROTECTED]]   Sent: 
Wednesday, August 02, 2006 1:33 PMTo: Asterisk Users 
Mailing List - Non-Commercial Discussion   Subject: Re: 
[asterisk-users] DUNDi with SIP 
  Using the SECRET variable for sip doesn't work.   
   On Wed, 2006-08-02 at 13:11 -0600, Douglas Garstang 
wrote:I've trying to use DUNDi with SIP to see if it 
works around   some limitations of IAX2.
I do a DUNDi lookup, get my SIP path, and try to 
dial it.   Asterisk immediately says 'No such host', 
eventhough that's   the path is just returned!   
 [Aug2 13:07:05] == 
Spawn extension (global_vmdeposit,   u9220371, 3) exited 
non-zero on 'SIP/3254101-eb7d'[Aug2 
13:07:13] -- Executing NoOp(SIP/3254101-6373, 
   *** OnNet originated call Chocolate Chip 
3254101 -   9220371) in new stack  
  [Aug2 13:07:13] -- Executing 
AGI(SIP/3254101-6373,ipt/originator.py) in new 
stack[Aug2 
13:07:13] -- Launched AGI Script   
/var/lib/asterisk/agi-bin/ipt/originator.py
[Aug2 13:07:13] -- AGI Script Executing 
Application:(SetAccount) Options: (9220371)  
  [Aug2 13:07:13] -- AGI Script 
Executing Application:   (ChanIsAvail) Options: 
(SIP/9220371)[Aug2 
13:07:14] -- AGI Script Executing Application: 
   (Dial) Options:   
(SIP/dundisip:[EMAIL PROTECTED]/9220371)  
  [Aug2 13:07:14] WARNING[5429]: chan_sip.c:1980 
  create_addr: No such host: xxx.yyy.142.163/9220371  
  [Aug2 13:07:14] NOTICE[5429]: app_dial.c:1040 
  dial_exec_full: Unable to create channel of type 'SIP' 
(cause   3 - No route to destination)
 [Aug2 13:07:14] == Everyone is busy/congested 
at this   time (1:0/0/1) 
  Not sure what is going on. I can see the query at the 
other   end, but it doesn't look like it ever receives the 
call.   Doug.   
   
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RE: [asterisk-users] DUNDi with SIP

[Douglas Garstang]

Alex,

Yep, I 
can dial 9220370 directly. I have two extensions on pbx1 and two on pbx2. I can 
place calls from 9220371 to 9220370 which goes through pbx2 only, and all is ok. 
9220370 and 9220371 are registered on pbx2.

I had 
this all working with IAX. I didn't change the keys... so I would assume that 
they would all still be ok. I haven't modified the keys or the key definitions 
in dundi.conf.

Doug

-Original Message-From: 
Alex Robar [mailto:[EMAIL PROTECTED]Sent: Wednesday, August 02, 
2006 2:48 PMTo: Asterisk Users Mailing List - Non-Commercial 
DiscussionSubject: Re: [asterisk-users] DUNDi with 
SIP
Doug,Two 
  things: If you try to place that call manually (either via dialling it from a 
  phone that supports SP URIs or by making an ext. for it in your dialplan and 
  calling that extension), does it work properly? Are you able to place the 
  call? If not, is the CLI output the same as when you try it via DUNDi? 
  Second, are your keys generated properly, with public keys shared 
  between the two boxes OK? I had a lot of DUNDi problems initially, and found 
  that my keys were the problem.Alex
  On 8/2/06, Douglas 
  Garstang [EMAIL PROTECTED] 
  wrote:
  


Alex,

Thanks... I haven't had any 
luck with it yet.

My dundi.conf 
has:


180netsip = 
global_dundi_local,1,SIP,dundisip:[EMAIL PROTECTED]/${NUMBER},nopartial


and my sip.conf 
has:


[dundisip]type=usercontext=global_dundi_localsecret=password


A DUNDI lookup on the 
console returns a SIP path:

*CLI dundi lookup [EMAIL PROTECTED] 1. 1 
SIP/dundisip:[EMAIL PROTECTED]/9220370 
(EXISTS) from 00:14:22:1e:2a:d0, expires in 0 
sDUNDi lookup completed in 129 ms

However, when I try to 
connect, I get a 'No such host' error...

*CLI [Aug 2 
14:18:43] -- Executing NoOp("SIP/3254101-a8d9", "*** 
OnNet originated call "Chocolate Chip" 3254101 - 9220371") in 
new stack[Aug 2 14:18:43] -- Executing 
AGI("SIP/3254101-a8d9", "ipt/originator.py") in new stack[Aug 2 
14:18:43] -- Launched AGI Script 
/var/lib/asterisk/agi-bin/ipt/originator.py[Aug 2 
14:18:43] -- AGI Script Executing Application: 
(SetAccount) Options: (9220371)[Aug 2 
14:18:43] -- AGI Script Executing Application: 
(ChanIsAvail) Options: (SIP/9220371)[Aug 2 
14:18:43] -- AGI Script Executing Application: 
(Dial) Options: 
(SIP/dundisip:[EMAIL PROTECTED]/9220371)[Aug 2 14:18:43] 
WARNING[7842]: chan_sip.c:1980 create_addr: No such host: 
xxx.yyy.142.163/9220371[Aug 2 14:18:43] NOTICE[7842]: 
app_dial.c:1040 dial_exec_full: Unable to create channel of type 'SIP' 
(cause 3 - No route to destination)[Aug 2 14:18:43] == 
Everyone is busy/congested at this time (1:0/0/1)

Doug.



-Original 
Message-From: Alex Robar [mailto:[EMAIL PROTECTED]]Sent: Wednesday, August 
02, 2006 2:17 PMTo: Asterisk Users Mailing List - Non-Commercial 
DiscussionSubject: Re: [asterisk-users] DUNDi with 
SIP
You can use an unchanging password. It's not as secure, 
but it will provide functionality.Alex


On 8/2/06, Douglas Garstang  
[EMAIL PROTECTED] wrote: 


So what are the options?
 -Original Message- 
 From: Aaron Daniel [mailto:[EMAIL PROTECTED]] 
Sent: Wednesday, August 02, 2006 2:03 PM To: Asterisk Users Mailing 
List - Non-Commercial Discussion Subject: RE: [asterisk-users] DUNDi 
with SIP  I'm talking about the rotating DUNDi 
secret that is stored in dbsecret in iax.conf.It doesn't 
exist in the SIP channel. On Wed, 2006-08-02 at 13:43 -0600, 
Douglas Garstang wrote:   Secret? Do you mean sbsecret in 
sip.conf?-Original Message- 
  From: Aaron Daniel [mailto:[EMAIL PROTECTED]] 
  Sent: Wednesday, August 02, 2006 1:33 PMTo: 
Asterisk Users Mailing List - Non-Commercial Discussion   
Subject: Re: [asterisk-users] DUNDi with SIP
   Using the SECRET variable for sip doesn't work. 
 On Wed, 2006-08-02 at 13:11 -0600, 
Douglas Garstang wrote:I've trying to use DUNDi with 
SIP to see if it works around   some limitations of IAX2. 
   I do a DUNDi lookup, get my 
SIP path, and try to dial it.   Asterisk immediately says 
'No such host', eventhough that's   the path is just 
returned![Aug2 
13:07:05] == Spawn extension (global_vmdeposit,  
 u9220371, 3) exited non-zero on 'SIP/3254101-eb7d'   
 [Aug2 13:07:13] -- Executing 
NoOp("SIP/3254101-6373","*** OnNet originated call 
"Chocolate Chip" 3254101 -   9220371") in new 
stack[Aug2 
13:07:13] -- Executing AGI("SIP/3254101-6373", 
   "ipt/originator.py") in new stack
[Aug2 13:07:13] -- Launched AGI 
Script   /var/lib/asterisk/agi-bin/ipt/originator.py 
   [Aug2 13:07:13] -- 

[asterisk-users] chan_zap.c: Failed to read gains: Invalid argument

[jan.sarin]

Hi,

I'm recieving the following error in my asterisk log (when starting *):
chan_zap.c: Failed to read gains: Invalid argument

Why? Attaching my zapata.conf and zaptel.conf. Using TE405P.

Thanks!

zaptel.conf:

span=1,1,0,ccs,hdb3
span=2,0,0,ccs,hdb3
span=3,0,0,ccs,hdb3
span=4,0,0,ccs,hdb3

bchan=1-15,17-31
dchan=16
bchan=32-46,48-62
dchan=47
bchan=63-77,79-93
dchan=78
bchan=94-108,110-124
dchan=109

loadzone=se
defaultzone=se

Zapata.conf

[channels] 
language=se
context=from-pstn 
switchtype=euroisdn 
pridialplan=unknown 
prilocaldialplan=unknown 
signalling=pri_cpe 
usecallerid=yes 
hidecallerid=no
callwaiting=yes 
callerid=asreceived
usecallingpres=yes
callwaitingcallerid=yes 
threewaycalling=yes 
transfer=yes 
cancallforward=yes 
callreturn=yes 
echocancel=yes 
echocancelwhenbridged=no 
echotraining=400
rxgain=-1.0
txgain=-1.5
group=0 
callgroup=1 
pickupgroup=1 
immediate=no 
overlapdial=no
channel = 1-15,17-31,32-46,48-62

group=1
channel = 94-108,110-124

group=2
context=from-internal
signalling=pri_net
channel = 63-77,79-93

Regards,
Jan
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Re: [asterisk-users] Re: need a pointer regarding scripting asterisk

[Andy Kuo]

Hi,

Can you give a quick example on how to query an EXTERNAL database?

Thank you.
Andy

On 7/29/06, Tzafrir Cohen [EMAIL PROTECTED] wrote:

On Fri, Jul 28, 2006 at 04:08:19PM -0500, shawn bright wrote:
 i would use a dial plan, but we are monitoring about 1200 units in the
 field, i thought a dial plan would be a little long or complex for that. I
 suppose that i could use a dial plan and set guys up by editing the
 extensions.conf file for each one ? I just thought it might be easier to
 script it somehow.

You can always generate part of extensions.conf automatically and
#include it. It will be updated by, e.g., 'extensions reload'.

Maybe you'll also find a smart way to do that using wildcards or
whatever. You can also query the internal asteriskdb or an external
dataase from the dialplan.

--
Tzafrir Cohen sip:[EMAIL PROTECTED]
icq#16849755  iax:[EMAIL PROTECTED]
+972-50-7952406  jabber:[EMAIL PROTECTED]
[EMAIL PROTECTED] http://www.xorcom.com
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[asterisk-users] RE: Asterisk with VoIP phone (shadowym)

[J Rangi]

Thanks for the information.
I have been a paying a lot to make international calls. After reading 
through this mailing list I have a feeling that a system can be setup 
easily where the telecommunication can be affordable. Still remember 
those days when I had to wait for two weeks to talk to my family and it 
used to cost me over $15 just for 30 minutes.


Here is what I am planning. I want to take say 50 channel PRI or T1 or 
DID number for Country A and 50 Channel DID number in another country B. 
Now any user from country A (with PSTN phone) will call in Asterisk 
server using local telephone number and dial the number to make calls to 
country B. Now the asterisk server will route the call to the 
destination using country B's local number which will be local to the 
destination number.

This way no one will be paying long distance fees.
I am not sure what all is involved in this specially at macro level.
I will appreciate if some one has already done this and would like to 
share. Or even if some one can point me in the right direction.

Dont want to reinvent the wheel.

Thank you,
-






You need to carefully consider outside VoIP providers IMHO.  I would look
for providers who are very upfront about their network architecture and how
they connect to the PSTN (the public telephone network).  As a minimum, I
would ask for IP addresses to some of their SIP servers and check ping
times.  I would look for consistent ping times at different times of the day
with round trips below 50ms ideally but definitely no more than 100ms.  A
lot of this depends on geography.  It is also important to know how they
connect to the public phone network IMHO.   Ideally, they have direct TDM
connections (digital telephone connections) as opposed to IP connections to
who knows where.  


There are other considerations as well but I think that is a good start
IMHO.


 _  

From: Bruce Reeves [mailto:[EMAIL PROTECTED] 
Sent: Wednesday, August 02, 2006 4:58 AM

To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk with VoIP phone


Yes this is possible, you just setup the softphones and maybe the provider
in sip.conf and write your dialplan  :) 



On 8/1/06, J Rangi  [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]  wrote: 


Hello,

Is is possible to setup an asterisk server with out buying Digium card. 
I mean can we do this type of setup.

We all know that X-Lite can be used as a soft phone to have an IP
extension.
Is it possible to take a service from another VoIP service provider, and
get the IP phone number. Make that phone numbe gateway to outside world. 
Now all the internal extensions use that phone to receive and make calls

to out side world.
Has any one done this kind of setup or know anything about this.

Thank you,
-Jai





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[asterisk-users] Re: DUNDi with SIP

[Tony Mountifield]

In article [EMAIL PROTECTED],
Douglas Garstang [EMAIL PROTECTED] wrote:
  -Original Message-
  From: Tony Mountifield [mailto:[EMAIL PROTECTED]
  Sent: Wednesday, August 02, 2006 2:01 PM
  To: asterisk-users@lists.digium.com
  Subject: [asterisk-users] Re: DUNDi with SIP
  
  
  In article 
  [EMAIL PROTECTED],
  Douglas Garstang [EMAIL PROTECTED] wrote:
   I've trying to use DUNDi with SIP to see if it works around 
  some limitations of IAX2.
   
   I do a DUNDi lookup, get my SIP path, and try to dial it. 
  Asterisk immediately says 'No such
   host', eventhough that's the path is just returned!
   
   [Aug  2 13:07:05]   == Spawn extension (global_vmdeposit, 
  u9220371, 3) exited non-zero on
   'SIP/3254101-eb7d'
   [Aug  2 13:07:13] -- Executing NoOp(SIP/3254101-6373, 
  *** OnNet originated call
   Chocolate Chip 3254101 - 9220371) in new stack
   [Aug  2 13:07:13] -- Executing AGI(SIP/3254101-6373, 
  ipt/originator.py) in new stack
   [Aug  2 13:07:13] -- Launched AGI Script 
  /var/lib/asterisk/agi-bin/ipt/originator.py
   [Aug  2 13:07:13] -- AGI Script Executing Application: 
  (SetAccount) Options: (9220371)
   [Aug  2 13:07:13] -- AGI Script Executing Application: 
  (ChanIsAvail) Options: (SIP/9220371)
   [Aug  2 13:07:14] -- AGI Script Executing Application: 
  (Dial) Options:
   (SIP/dundisip:[EMAIL PROTECTED]/9220371)
   [Aug  2 13:07:14] WARNING[5429]: chan_sip.c:1980 
  create_addr: No such host: xxx.yyy.142.163/9220371
  
  Try specifying the SIP argument as:
  
  SIP/dundisip:[EMAIL PROTECTED]@xxx.yyy.142.163
  
  See the following line in the sample extensions.conf as an example:
  
  ;exten = 
  _42X.,1,Dial(SIP/user:[EMAIL PROTECTED]:[EMAIL PROTECTED],30,rT)
 
 Tony... it's DUNDi

OK, I know nothing about DUNDi. I was only highlighting what appeared
to be invalid or at least ambiguous syntax in the SIP channel requested.
SIP appears not to like SIP/user:[EMAIL PROTECTED]/number, but instead wants
SIP/user:[EMAIL PROTECTED]@host

Unless you can set up a sip.conf friend entry [] and then use
SIP//number

Hope this helps. If not, oh well.

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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RE: [asterisk-users] Re: DUNDi with SIP

[Douglas Garstang]

 -Original Message-
 From: Tony Mountifield [mailto:[EMAIL PROTECTED]
 Sent: Wednesday, August 02, 2006 3:49 PM
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] Re: DUNDi with SIP
 
 
 In article 
 [EMAIL PROTECTED],
 Douglas Garstang [EMAIL PROTECTED] wrote:
   -Original Message-
   From: Tony Mountifield [mailto:[EMAIL PROTECTED]
   Sent: Wednesday, August 02, 2006 2:01 PM
   To: asterisk-users@lists.digium.com
   Subject: [asterisk-users] Re: DUNDi with SIP
   
   
   In article 
   [EMAIL PROTECTED],
   Douglas Garstang [EMAIL PROTECTED] wrote:
I've trying to use DUNDi with SIP to see if it works around 
   some limitations of IAX2.

I do a DUNDi lookup, get my SIP path, and try to dial it. 
   Asterisk immediately says 'No such
host', eventhough that's the path is just returned!

[Aug  2 13:07:05]   == Spawn extension (global_vmdeposit, 
   u9220371, 3) exited non-zero on
'SIP/3254101-eb7d'
[Aug  2 13:07:13] -- Executing NoOp(SIP/3254101-6373, 
   *** OnNet originated call
Chocolate Chip 3254101 - 9220371) in new stack
[Aug  2 13:07:13] -- Executing AGI(SIP/3254101-6373, 
   ipt/originator.py) in new stack
[Aug  2 13:07:13] -- Launched AGI Script 
   /var/lib/asterisk/agi-bin/ipt/originator.py
[Aug  2 13:07:13] -- AGI Script Executing Application: 
   (SetAccount) Options: (9220371)
[Aug  2 13:07:13] -- AGI Script Executing Application: 
   (ChanIsAvail) Options: (SIP/9220371)
[Aug  2 13:07:14] -- AGI Script Executing Application: 
   (Dial) Options:
(SIP/dundisip:[EMAIL PROTECTED]/9220371)
[Aug  2 13:07:14] WARNING[5429]: chan_sip.c:1980 
   create_addr: No such host: xxx.yyy.142.163/9220371
   
   Try specifying the SIP argument as:
   
   SIP/dundisip:[EMAIL PROTECTED]@xxx.yyy.142.163
   
   See the following line in the sample extensions.conf as 
 an example:
   
   ;exten = 
   _42X.,1,Dial(SIP/user:[EMAIL PROTECTED]:[EMAIL PROTECTED],30,rT)
  
  Tony... it's DUNDi
 
 OK, I know nothing about DUNDi. I was only highlighting what appeared
 to be invalid or at least ambiguous syntax in the SIP channel 
 requested.
 SIP appears not to like SIP/user:[EMAIL PROTECTED]/number, but instead wants
 SIP/user:[EMAIL PROTECTED]@host
 
 Unless you can set up a sip.conf friend entry [] and then use
 SIP//number
 
 Hope this helps. If not, oh well.

Well yes, it looked dubious to me too, although I can't find the syntaxt 
documented anywhere.
However, that's what DUNDis giving me as a path to the phone!

Something is screwed with DUNDi and SIP. Has ANYONE actually implemnted it?
I can't find it documented anywhere

Doug.
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RE: [asterisk-users] Re: DUNDi with SIP

[Douglas Garstang]

 -Original Message-
 From: Tony Mountifield [mailto:[EMAIL PROTECTED]
 Sent: Wednesday, August 02, 2006 3:49 PM
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] Re: DUNDi with SIP
 
 
 In article 
 [EMAIL PROTECTED],
 Douglas Garstang [EMAIL PROTECTED] wrote:
   -Original Message-
   From: Tony Mountifield [mailto:[EMAIL PROTECTED]
   Sent: Wednesday, August 02, 2006 2:01 PM
   To: asterisk-users@lists.digium.com
   Subject: [asterisk-users] Re: DUNDi with SIP
   
   
   In article 
   [EMAIL PROTECTED],
   Douglas Garstang [EMAIL PROTECTED] wrote:
I've trying to use DUNDi with SIP to see if it works around 
   some limitations of IAX2.

I do a DUNDi lookup, get my SIP path, and try to dial it. 
   Asterisk immediately says 'No such
host', eventhough that's the path is just returned!

[Aug  2 13:07:05]   == Spawn extension (global_vmdeposit, 
   u9220371, 3) exited non-zero on
'SIP/3254101-eb7d'
[Aug  2 13:07:13] -- Executing NoOp(SIP/3254101-6373, 
   *** OnNet originated call
Chocolate Chip 3254101 - 9220371) in new stack
[Aug  2 13:07:13] -- Executing AGI(SIP/3254101-6373, 
   ipt/originator.py) in new stack
[Aug  2 13:07:13] -- Launched AGI Script 
   /var/lib/asterisk/agi-bin/ipt/originator.py
[Aug  2 13:07:13] -- AGI Script Executing Application: 
   (SetAccount) Options: (9220371)
[Aug  2 13:07:13] -- AGI Script Executing Application: 
   (ChanIsAvail) Options: (SIP/9220371)
[Aug  2 13:07:14] -- AGI Script Executing Application: 
   (Dial) Options:
(SIP/dundisip:[EMAIL PROTECTED]/9220371)
[Aug  2 13:07:14] WARNING[5429]: chan_sip.c:1980 
   create_addr: No such host: xxx.yyy.142.163/9220371
   
   Try specifying the SIP argument as:
   
   SIP/dundisip:[EMAIL PROTECTED]@xxx.yyy.142.163
   
   See the following line in the sample extensions.conf as 
 an example:
   
   ;exten = 
   _42X.,1,Dial(SIP/user:[EMAIL PROTECTED]:[EMAIL PROTECTED],30,rT)
  
  Tony... it's DUNDi
 
 OK, I know nothing about DUNDi. I was only highlighting what appeared
 to be invalid or at least ambiguous syntax in the SIP channel 
 requested.
 SIP appears not to like SIP/user:[EMAIL PROTECTED]/number, but instead wants
 SIP/user:[EMAIL PROTECTED]@host
 
 Unless you can set up a sip.conf friend entry [] and then use
 SIP//number
 
 Hope this helps. If not, oh well.

Tony, I was able to fiddle with dundi.conf, and am now getting a SIP path in 
the format SIP/user:[EMAIL PROTECTED]@host:

*CLI dundi lookup [EMAIL PROTECTED]
  1. 1 SIP/dundisip:[EMAIL PROTECTED]@xxx.yyy.142.163 (EXISTS)
 from 00:14:22:1e:2a:d0, expires in 0 s
DUNDi lookup completed in 171 ms

However, when I try to dial this, I am still getting:

[Aug  2 15:57:35] WARNING[9916]: chan_sip.c:1980 create_addr: No such host: 
[EMAIL PROTECTED]
[Aug  2 15:57:35] NOTICE[9916]: app_dial.c:1040 dial_exec_full: Unable to 
create channel of type 'SIP' (cause 3 - No route to destination)

Arrgh


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[asterisk-users] need dialout help in python script

[shawn bright]

Lo there, i have an app that needs to initiate a phone call on a zap channel.i have been able to test it out ok with the method of dropping a call fileinto the /var/spool/asterisk/outgoing and specifing the phone number in the call file.
 what i need to do, however, is initiate a phone call from a python script.i need to pass asterisk the phone number and then a couple of files to play.if anyone can tell me how to pull this off, or could post a link to some good doc or how-to,
i would greatly appreciate it.thanks- shawn
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RE: [asterisk-users] Re: DUNDi with SIP

[Watkins, Bradley]

The way to make this work is to define a sip user/peer with the IP
address in it, then have your dundi.conf entry look like:

180netsip = global_dundi_local,1,SIP/peername/${NUMBER},nopartial

As far as I can tell from the code, this is the only way to make it work
properly based on the way the string sent to the channel driver is being
parsed.

- Brad

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Wednesday, August 02, 2006 5:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Re: DUNDi with SIP

 -Original Message-
 From: Tony Mountifield [mailto:[EMAIL PROTECTED]
 Sent: Wednesday, August 02, 2006 3:49 PM
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] Re: DUNDi with SIP
 
 
 In article
 [EMAIL PROTECTED],
 Douglas Garstang [EMAIL PROTECTED] wrote:
   -Original Message-
   From: Tony Mountifield [mailto:[EMAIL PROTECTED]
   Sent: Wednesday, August 02, 2006 2:01 PM
   To: asterisk-users@lists.digium.com
   Subject: [asterisk-users] Re: DUNDi with SIP
   
   
   In article
   [EMAIL PROTECTED],
   Douglas Garstang [EMAIL PROTECTED] wrote:
I've trying to use DUNDi with SIP to see if it works around
   some limitations of IAX2.

I do a DUNDi lookup, get my SIP path, and try to dial it. 
   Asterisk immediately says 'No such
host', eventhough that's the path is just returned!

[Aug  2 13:07:05]   == Spawn extension (global_vmdeposit, 
   u9220371, 3) exited non-zero on
'SIP/3254101-eb7d'
[Aug  2 13:07:13] -- Executing NoOp(SIP/3254101-6373, 
   *** OnNet originated call
Chocolate Chip 3254101 - 9220371) in new stack
[Aug  2 13:07:13] -- Executing AGI(SIP/3254101-6373, 
   ipt/originator.py) in new stack
[Aug  2 13:07:13] -- Launched AGI Script 
   /var/lib/asterisk/agi-bin/ipt/originator.py
[Aug  2 13:07:13] -- AGI Script Executing Application: 
   (SetAccount) Options: (9220371)
[Aug  2 13:07:13] -- AGI Script Executing Application: 
   (ChanIsAvail) Options: (SIP/9220371)
[Aug  2 13:07:14] -- AGI Script Executing Application: 
   (Dial) Options:
(SIP/dundisip:[EMAIL PROTECTED]/9220371)
[Aug  2 13:07:14] WARNING[5429]: chan_sip.c:1980
   create_addr: No such host: xxx.yyy.142.163/9220371
   
   Try specifying the SIP argument as:
   
   SIP/dundisip:[EMAIL PROTECTED]@xxx.yyy.142.163
   
   See the following line in the sample extensions.conf as
 an example:
   
   ;exten =
   _42X.,1,Dial(SIP/user:[EMAIL PROTECTED]:[EMAIL PROTECTED],30,rT)
  
  Tony... it's DUNDi
 
 OK, I know nothing about DUNDi. I was only highlighting what appeared 
 to be invalid or at least ambiguous syntax in the SIP channel 
 requested.
 SIP appears not to like SIP/user:[EMAIL PROTECTED]/number, but instead wants 
 SIP/user:[EMAIL PROTECTED]@host
 
 Unless you can set up a sip.conf friend entry [] and then use 
 SIP//number
 
 Hope this helps. If not, oh well.

Tony, I was able to fiddle with dundi.conf, and am now getting a SIP
path in the format SIP/user:[EMAIL PROTECTED]@host:

*CLI dundi lookup [EMAIL PROTECTED]
  1. 1 SIP/dundisip:[EMAIL PROTECTED]@xxx.yyy.142.163 (EXISTS)
 from 00:14:22:1e:2a:d0, expires in 0 s DUNDi lookup completed in
171 ms

However, when I try to dial this, I am still getting:

[Aug  2 15:57:35] WARNING[9916]: chan_sip.c:1980 create_addr: No such
host: [EMAIL PROTECTED] [Aug  2 15:57:35] NOTICE[9916]:
app_dial.c:1040 dial_exec_full: Unable to create channel of type 'SIP'
(cause 3 - No route to destination)

Arrgh


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