date:20051019

I am asked to consider deploying asterisk servers as soft-switches on a
large scale, but wanted to preserve TDM properties of a call, especially
for modem applications which some of the end users may want.  I was
thinking TDMoE may work well for this, at least on the surafce but had
specific questions regarding modem data on the call.

As most of you are aware a TDM network virtually guarantees that the
data that enters the network comes out at the same cadence that it went
in.  Modems like this near exact timing.  IP networks have no such
guarantee so modems tend to not want to work well when VoIP protocols
are used.  Compression methods (codecs) used in VoIP can also distort
the data for a modem call, as such they are undesirable.

The usage that I am considering would be to have soft switches placed in
stragetic locations throughout a large geographic area but be able to
provide service to customers, which can include modem usage (think large
phone company selling arbitrary phone lines to be used however the
customer sees fit).  As such I need modems to be able to work over this
network.

I had considered linking all the remote sites together via TDMoE
(private network primarily using dark fiber).  Does TDMoE provide
effectively the same capacity to preserve modem data (upto and including
56k speeds) as a T1 would?  Or would I need to actually transmit voice
channels on T1/DS3/whatever framed circuits using the Zap interface?

Has anyone tried TDMoE on longer runs, or at all with modem data?


-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378


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Re: [Asterisk-Users] DID setup from goiax.com

On Wed, 2005-10-19 at 14:24 +0800, Ronald Wiplinger wrote:
 Can anybody post a step by step setup guide, please?

Its like anything else once you have signed up ...

in iax.conf
register = PHONENUMBER:[EMAIL PROTECTED]/goiax-in

[goiax]
type= peer
host= server1.goiax.com
context = default
secret  = PASSWORD
allow   = gsm
;allow  = ulaw
;disallow   = all
notransfer  = yes
qualify = yes
auth= md5
username= PHONENUMBER


replace PHONENUMBER with the 8782 number you were issued.  Replace
PASSWORD with your password from you account signup.

Then in extensions.conf
; for outbound
exten = _1NX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN},90,R)
exten = _1NX,2,Busy
exten = _1NX,102,Congestion
exten = _1NX,202,playback(tt-weasels)

; for inbound
exten = goiax-in,1,DO WHATEVER HERE

asterisk -rx reload

you should be set.  


-- 
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UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378


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Re: [Asterisk-Users] SER and Asterisk

On Wed, 2005-10-19 at 10:55 +0200, Yair Hakak wrote:
 hello,
  trace the SIP packets and see if they are actually addressed to 5062.
 if you post the ngrep or ethereal dump we'll see whats actually going
 on. I do this with SER on 5060 and asterisk on 5070 and there are no
 problems -  my extensions point to 5060 and my DID's point to 5070 so
 asterisk serves as the gateway to the PSTN. 
  
 -yair
 
  
also look for dns packets and see if htey are pulling the server info.
Some sip clients look for specific server type dns records to see where
they should go.

5060 is the default, wouldnt it make more sense to have the default port
be what you want the devices to goto and have that proxy to the device
you dont want direct connectivity to?  Or am I missing something in that

 
-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378


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[Asterisk-Users] How to suppres leading zeros in zapata.conf?

2005-10-19 Thread Klaus P. Pieper


Hi,

my asterisk is running behind a Siemens HiCom, connected via ISDN. 
Connection to that Hicom equipment is madewith an AVM Fritz! card. I use 
a HFC card for the local trunk (NT mode with zaphfc). My problem: 
something adds an additional leading 0 to all inbound calls (except 
those coming from other local HiCom users).


Originally, it added the leading 0 also to the local calls (from other 
users at the HiCom device). But I can suppress this leading 0 by setting 
prilocaldialplan=unknown in zapata.conf. However, this setting (and a 
pridialplan=unknown as well) seems to have no effect on any inbound call 
coming from the outside through the HiCom to my asterisk.


Not sure if this gives enough backgound to answer the question how to 
avoid this additional leading 0 but any idea is welcome.


Klaus


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[Asterisk-Users] Persistant connection for MYSQL command

2005-10-19 Thread Ed Greenberg


When doing mysql commands, such as:

exten = _X.,1,MYSQL(Connect connid localhost dbuser dbpass dbname)
exten = _X.,2,MYSQL(Query resultid ${connid} SELECT\ scriptname\ from\ 
mac2pin\ where\ userid=${CALLERIDNAME})

exten = _X.,3,MYSQL(Fetch fetchid ${resultid} AGIScript)
exten = _X.,4,GotoIf($[${AGIScript} = NULL]?5:7)
exten = _X.,5,AGI(${DefaultAGIScript},${EXTEN})
exten = _X.,6,Goto(_X.,8)
exten = _X.,7,AGI(${AGIScript},${EXTEN})
exten = _X.,8,MYSQL(Clear ${resultid})
exten = _X.,9,MYSQL(Disconnect ${connid})
exten = _X.,10,Hangup

over and over again, is there any way to cache the connection so we can 
reuse it, to save overhead?


/edg
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[Asterisk-Users] Problem with select correct network interface (oh323)

2005-10-19 Thread Oleh Mukha

i build asterisk on pc with 3 network inerface
eth0 (yyy.yyy.yyy.yyy) main public ip 
eth1 (xxx.xxx.xxx.xxx) seconf public ip used only for voip connection
eth2 (zzz.zzz.zzz.zzz) local ip 
i config oh323 to bind eth1 interface 
i try make call 
from my local network - Asterisk - provider h323

when i try to call from ata 186 throught my astersik oh323 module
asetrisk resive calls from ata but send it to my oh323 providet not from eth1 
(with ip xxx.xxx.xxx.xxx) or from eth0 (ip yyy.yyy.yyy.yyy) 

how can i tel asterisk send data from me to my provider from eth1 (ip  
xxx.xxx.xxx.xxx)


Oleh Mukha
IClub
380322722738
www.ic.lviv.ua
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Re: [Asterisk-Users] fax - conversion problem

2005-10-19 Thread Brian May

 asterisk == asterisk  [EMAIL PROTECTED] writes:

asterisk The problem is in the tiff2ps, not in the ps2pdf.  I
asterisk found that if I remove the -h and -w parameter
asterisk everything is OK

My computer has a tiff2pdf command (from the libtiff-tools Debian
package), so I can do everything in one command:


--- /usr/local/sbin/mailfax ---
#!/bin/sh -e

FAXFILE=$1
RECIPIENT=$2
FAXSENDER=$3
REMOTESTATIONID=$4
FAXPAGES=$5
FAXRESOLUTION=$6

if [ ! -f $FAXFILE ]
then
echo Fax $FAXFILE not found 2
exit 1
fi

tiff2pdf -pA4 $FAXFILE |
  mime-construct --to $RECIPIENT --subject Fax from $FAXSENDER \
 --attachment fax.pdf --type application/pdf --file -
--- cut ---

I am not sure of the -pA4 option, but I don't know enough about FAX
standards to change it - it might be better to set the resolution with
-r depending on the FAXRESOLUTION parameter. I did a web search on the
resolution for the different modes, and got numerous different answers
for the same thing :-(.

Oh, and this is called with:

[fax]
exten = s,1,Macro(faxreceive)
exten = h,1,system(/usr/local/sbin/mailfax ${FAXFILE} ${EMAILADDR} 
${CALLERIDNUM} ${REMOTESTATIONID} ${FAXPAGES} ${FAXRESOLUTION})

[macro-faxreceive]
exten = s,1,SetVar(FAXFILE=/var/spool/asterisk-fax/${UNIQUEID}.tif)
exten = s,2,SetVar([EMAIL PROTECTED])
exten = s,3,rxfax(${FAXFILE})
-- 
Brian May [EMAIL PROTECTED]
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Re: [Asterisk-Users] SER and Asterisk

2005-10-19 Thread Yair Hakak


On 10/19/05, Yair Hakak [EMAIL PROTECTED] wrote:

i do it this way because i want all the dialplan logic and CDR having to do with PSTN in asterisk, not SER. 
so, calls from the outside are adressed to [EMAIL PROTECTED]:5070 and hit asterisk. asterisk either sends them along to 5060, or handles them internally (IVR, voicemail, etc) based on the dialplan. 

clients on the inside are registered to the SER at 5060 and the SER automatically forwards them to asterisk. if they are PSTN asterisk serves as PSTN gateway, if they are internal, asterisk native bridges and drops out, but still keeps the CDR (i have full SIP addresses in my dial statements instead of asterisk SIP peers) 

the reason i do this is i found that if the endpoints are scattered on the internet, SER+rtpproxy is much more stable than asterisk as a SIP server (asterisk kept dropping endpoints). This way SER serves as a completely dumb SIP server, and just sends everything along. there is a minimal increase in overhead (i could handle internal calls just with SER) but it's worth it to have all the dialplan logic and CDR's in one place. 


also, obviously, if i use an IAX provider for outgoing, asterisk has to be in the middle.

i agree though, it makes more sense to have SER on 5060 and asterisk somewhere else.

hope i'm making some sense, please point out if i'm doing something really stupid.
-yair

On 10/19/05, trixter aka Bret McDanel 
[EMAIL PROTECTED] wrote: 

On Wed, 2005-10-19 at 10:55 +0200, Yair Hakak wrote: hello,trace the SIP packets and see if they are actually addressed to 5062.  if you post the ngrep or ethereal dump we'll see whats actually going
 on. I do this with SER on 5060 and asterisk on 5070 and there are no problems -my extensions point to 5060 and my DID's point to 5070 so  asterisk serves as the gateway to the PSTN. -yair
also look for dns packets and see if htey are pulling the server info.Some sip clients look for specific server type dns records to see where they should go.5060 is the default, wouldnt it make more sense to have the default port
be what you want the devices to goto and have that proxy to the deviceyou dont want direct connectivity to?Or am I missing something in that --Trixter 
http://www.0xdecafbad.com Bret McDanelUK +44 870 340 4605 Germany +49 801 777 555 3402US +1 360 207 0479 or +1 516 687 5200FreeWorldDialup: 635378 -BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.1 (GNU/Linux)iD8DBQBDVg5f+1olxlzQw5cRAhl5AJ91lwjqMb2EPcDSXH69dOELBOq0IQCgvr8m4NqQAGLmWLokUXjl7Bi7SbI==thAz-END PGP SIGNATURE-

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Re: [Asterisk-Users] TDMoE question

2005-10-19 Thread Appan KH


You can use MPLS which takes care all the point you had mentioned.

appan kh

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

Sent: Wednesday, October 19, 2005 9:54 AM
Subject: [Asterisk-Users] TDMoE question



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[Asterisk-Users] Call queuing question

2005-10-19 Thread Peter Spikings

Hi,

Could I have clarification on the logic in app_queue which treats no
answer as needing a retry? What I want to do is have all calls firstly
always go to phone A, then if there is no answer make it call B or C in
a round robin fashion. The obvious thing to do is put a penalty on B  C
but then if phone A doesn't pick up it just keeps retrying which isn't
what I want as the person with phone A on their desk may be absent for a
couple of minutes. Could I ask why no answer is treated as needing a
retry rather than moving up to the next penalty group?

Thanks,

Peter Spikings.

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[Asterisk-Users] what hw/OS to choose [please help]

2005-10-19 Thread extreme2000

Hi,
First of all, I would like to say hello to everybody, it's my first post on the 
list.

I'm building a pbx for a client and I need help/suggestions on what hardware 
and os to choose. I've read all I could find on the net, but still can't decide 
myself. Appart from signal switching, the main concern here is reliability.
The config will stand as follows: 15 sip phone terminals, 4 POTS France telecom 
lines, 1 ISDN line, 4 ip-providers lines, all this will run on on a france 
telecom (argh) dsl line (20M/1M)

In the begining there will be quite a lot of load on this network, but in the 
future the client wishes to connect 30 WAN sip terminals to the asterisk server 
and add 8-10 ip-pstn lines. From what I've heard Asterisk is quite hungry on 
ressources, what kind of hardware can you suggest me to use? Is it worth to buy 
a server mainboard?

And then will the T410P with 4 FXO work together with a T1 (I've heard it was 
not recommended to use 2 digiums on the same M-board).

The second point is about OS, I thought about some free BSD or Solaris and also 
Debian, the first two for quality and Debian because it's well documented and I 
like it, but I don't have any serious opinion on that neither.

Thanks,

Jays

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Re: [Asterisk-Users] sip rfc bye violated?

2005-10-19 Thread Olle E. Johansson

Matt Hess wrote:
 Attached is a pcap of sip packets that pertain to another call similar
 to the history shown.. it's hard to nail these down as it takes a lot of
 time, patience and sifting through dumps.
 
Well, a pcap does not tell me how Asterisk reacts, sorry. That was what
I wanted to see - the logging from the SIP channel. From the pcap, I can
only see the same as in the history, Asterisk sends a re-invite, at the
same time as the other end sends a BYE.  We acknowledge the bye and the
other end sends a trying after it sent a BYE, which is interesting...

/Olle
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Re: [Asterisk-Users] TDMoE question

On Wed, 2005-10-19 at 10:43 +0100, Appan KH wrote:
 You can use MPLS which takes care all the point you had mentioned.
 
 appan kh

Not entirely, at least not as I understand MPLS.  MPLS will add a little
bit of data which is used to route the traffic, it doesnt deal with
encapsulating TDM data (say from a T1 or DS3 from a telco) and allowing
that to cross a data link.  So that still leaves the question of TDMoE
or not given that I need to optionally (and unknown beforehand) be able
to traffic modem data reliably.  

Unless you are talknig about using MPLS with TDMoE which doesnt answer
the actual question I had about has anyone tried it, does it work
reliably even at the faster modem speeds, etc.


-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378


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Re: [Asterisk-Users] Menu/IVR and transfering to an extension after pressing an option number.

 
 Here's what I'm trying to accomplish:
 
 Press 1 to transfer to extension
 Press 2 for Directory
 Press 0 for Operator
 
 Got directory and operator working. My problem is with transfering to an 
 extension after pressing 1. Asterisk keeps adding the 1 to the extension 
 that I need to transfer to (extension 500 would be 1500 to asterisk 
 rathern than 500). I've tried ${EXTEN}, ${EXTEN:1},2,3,4 etc..No luck.
 
 This is what I have currently:
 
 [MainMenu]
 include = extensions
 
 exten = s,1,Answer
 exten = s,2,SetMusicOnHold(default)
 exten = s,3,Set(TIMEOUT(digit)=5)
 exten = s,4,Set(TIMEOUT(response)=10)
 exten = s,5,Background(danner) ; Play Welcome to companyname greeting
 exten = i,1,Playback(invalid)
 exten = i,2,GoTo(MainMenu,s,1)
 
 
 ; Dial an extension
 exten = 1,1,Dial(SIP/${EXTEN:1})
 
 
 ; Go to the sales department
 exten = 2,1,Directory(extensions) ;
 
 ; Leave; a voicemail for the operator
 exten = 0,1,Dial(${P100}${P103},30,t) ;
 
 
 Can anybody be of any assistance?

The above should work with a couple of small changes (I'm using basically
the same thing).

Change your greating message to say something like ... or if you know
the extension, you may dial it at any time.  Then, dumpt the press 1
stuff.

Change your logic to something like this:
[bus-ivr-main]  
exten = s,1,Wait,1
exten = s,2,Answer 
exten = s,3,Set(TIMEOUT(digit)=5)  
exten = s,4,Set(TIMEOUT(response)=15)

exten = s,5,Background(npi-greeting)  ; Thanks for calling press 1 for   
exten = s,6,WaitExten   
exten = s,7,Goto(bus-ivr-main|s|3)
include = local-calls
include = local-extns

exten = 1,1,Goto(local-extns|3026|1) ; Sales
exten = 8,1,Goto(abclist|s|1); Company directory list
exten = 0,1,Goto(local-extns|3000|1) ; Operator
exten = *,1,Goto(vm|s|1)   ; go to voicemail menu
exten = #,1,Background(vm-goodbye)

The above essentially allows the caller to press 1, 8, 0, *, or # (as an 
example only); and if they know the extension, the include = local-extns
send other key-presses through the local-extns context.

The key statement about is the WaitExten, allowing asterisk to pick up
one or more key presses.


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[Asterisk-Users] Url dialing

2005-10-19 Thread Alessio Focardi


jsss My suggestion would be the one-line eyeBeam phone under
jsss development. Check out support.xten.com.


I checked a multiline versionof eyebeam: no url opening within the phone call,
using this syntax:

Dial(sip/399|||http://www.google.it)

Could it be that only IAX2 supports this ?

Tnx!


-- 
Best regards,
 Alessiomailto:[EMAIL PROTECTED]

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[Asterisk-Users] my SIPURA ATA does not make calls thru teliax

2005-10-19 Thread Kumara Jayaweera

Greetings!

Dear List,

I had been making calls thru teliax for more than two month using my SIPURA
ATA. but one day when my a/c balance fell around $ 18.00 (one month ago) and
after that it could not make any calls thru teliax. until now I can not make
calls. Mr. David said that their side is ok and check my side. But I did not
change any setting in my device when it beginning to happen. before I use
g729 (only) but now I have activated all the codecs in both sides. but
unfortunately, I can not make calls yet. it gives an engaging tone. I can
see my SIPURA ATA is always registered to the provider (teliax).

Does anyone have any idea about this matter?

Thanks in advance

Mohan Kumara





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[Asterisk-Users] Asterisk on Slackware ...

2005-10-19 Thread Support




Does anybody installed Asterisk on 
Slackware?
It seems the installation went ok. 
But which config files do I have to look and edit 
first for the testing on two internal peers.
Which free version of VoIP softphone is the best to 
use with asterisk?

Thanks ..

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Re: [Asterisk-Users] Asterisk on Slackware ...

2005-10-19 Thread Matt Florell

Hello,

We have 12 servers in production that run Asterisk on Slackware Linux they run beautifully. 

If you are starting out in Asterisk I suggest reading the recently completed Asterisk book(it's free in electronic format):
http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11

It's the best place to start.

As for softphones, what operating system will your users be on? will the softphones need to be SIP or IAX?

MATT---
On 10/19/05, Support [EMAIL PROTECTED] wrote:







Does anybody installed Asterisk on 
Slackware?
It seems the installation went ok. 
But which config files do I have to look and edit 
first for the testing on two internal peers.
Which free version of VoIP softphone is the best to 
use with asterisk?

Thanks ..


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Re: [Asterisk-Users] more dids added to goiax.com




 I made two attempts this morning to send some comments off list, and
 both got returned due to some sort of spam filter, so would hope that
 any future controls will not suffer from that inability to communicate.

Maybe if you posted what the returned email said then he could remove or
alter that filter.  both got returned due to some sort of spam filter,
doesn't help anyone.


 Those of us who have signed up early, and not abused your generosity
 perhaps ought to be on the top of the invite list?
 Can GOIAX be configured to show the assigned DID and registered name,
 rather than the present 202-556- that shows up when outbound calls
 could be made?
 A limit on number of calls per minute,hour,day,week or month might be a
 place to start
 Low DID usage and reassignment might discourage use. If one gives out a
 DID and it is then moved xx days later for low usage, then who will even
 publish it?

Or even use it?  What is the point of having a DID that changes all the
time?

 Perhaps expand the registration process to include name, address and
 another phone number?

And this would accomplish what?  I have many aliases that I use on websites
if I suspect SPAM or just dont feel like giving my details.

 If your business plan is to evolve into a pay service of some sort, you
 will want that information anyway

A business must make money otherwise it is a hobby.  Giving away DIDs and
outbound calling could quickly become a very expensive hobby.

If you had a PayPal Donate button, I for one would probably stick a few
bucks in the tip jar.

 Don't make everyones life too complicated simply to discourage a (
 hopefully ) small number of abusers.

If you think your life will be too complicated because you have to jump
through a couple hoops to use a FREE service, just think what happens when
someone files a complaint with the authorities because the owner's circuit
was used to call in threats.  Better yet, a terror threat and Homeland
Security swoops in with all the powers of the Patriot Act.

Mathew, don't be too generous here.  Protect yourself FIRST.  In all
reality, if I wanted to contact a sleeper cell in the US and not be worried
about Echelon or Carnivore, I would use your service through a proxy or a
pay cash at an internet cafe.  The number of abusers is not the issue, it
only takes one to cause some real nightmares.

Maybe you should even consider recording calls for your own protection.
Just play message to both the caller and the callee to inform them of the
recording.

 The service worked great, though call setup time did increase the last
 several days, probably due to the rising abuse

 John Novack



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[Asterisk-Users] IAX termination/DID provider in Panama?

2005-10-19 Thread Frank Tarczynski

Does anyone know of a IAX termination/DID provider in Panama?  (507 
country code).



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Re: [Asterisk-Users] IAX only speech one way

Have you tried 'iax debug' and/or using ethereal to see what (if anything)
either system might be spewing?



 Thanks for your suggestion.
 
 Unfortunately, it didnt change anything, A can still not hear B, but B
 can hear A, strange..
 
 Michael
 
 2005/10/19, Rich Adamson [EMAIL PROTECTED]:
 
   I have two Asterisk's connected via IAX, they are sitting on the same
   network, via a VPN, so there should be no problems with firewalls.
  
   My problem is that when a person calls from A to B, A will not hear B
   speak. B hears A fine.
  
   I doesn't matter who initiates the call.
  
   One of the Asterisk'ses is a new installation, just installed, but
   with the Conf-files from an earlier setup, that worked fine.
  
   Asterisk version on computer A is Asterisk CVS-v1-0-12/09/04-08:58:31
   Asterisk version on computer B is Asterisk 
   CVS-D2005.05.28.22.00.00-10/17/05
  
   Two different versions, but I dont think it should matter?
 
  Not sure this applies, but I was having the same problem with teliax.com
  and turning off the jitterbuffer in iax.conf fixed the problem. Kind
  of looks like we are running two different versions of asterisk as
  well, but I'd suspect that teliax has modified their system for
  other business purposes.
 
  Try jitterbuffer=no and see if it helps.
 
  Rich
 
 
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Re: [Asterisk-Users] Priority jump in AEL

2005-10-19 Thread Sergey Okhapkin





There is no way in AEL to specify the priority explicitly. To solve the problem use DB_EXISTS function. Here is an example from my dialplan:

 if(${DB_EXISTS(Provider/${prov}/used)}) 
 Set(MINUTES_USED=${DB_RESULT}); 


On Tue, 2005-10-18 at 21:16 +, Kris Edwards wrote:


-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

I use this macro for call screening:

[macro-screen]
exten = s,1,Wait(1)
exten = s,2,DBget(SCREENFILE=callerid/${CALLERIDNUM})
exten =
s,3,ParkAndAnnounce(beep:beep:callfrom:${SCREENFILE}:holdingonexten:PARKED:beep:beep:${SCREENFILE}:isholdingonext:PARKED|180|${ARG1}|${ARG2})
exten = s,103,Set(SCREENFILE=/var/lib/asterisk/sounds/names/${UNIQUEID})
exten = s,104,Playback(unknownid)
exten = s,105,Record(${SCREENFILE}:gsm|3)
exten = s,106,System(/usr/bin/normalize -g 6db ${SCREENFILE})
exten = s,107,DBput(callerid/${CALLERIDNUM}=${SCREENFILE})
exten =
s,108,ParkAndAnnounce(beep:beep:callfrom:${SCREENFILE}:holdingonexten:PARKED:beep:beep:${SCREENFILE}:isholdingonext:PARKED|40|${ARG1}|${ARG2})


I'm trying to convert my dialplan to ael, but I don't get how to handle
the jump if there is no entry in the database for the caller.  I'm
guessing it's an if statement, but what does the db return if there is
no entry? 0, null??  If somebody could get me started with what that
staement should be (at 103) then I should be good to go.
(If this is a stupid question or explained elsewhere, feel free to let
me know)

Thanks,

Kris
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Version: GnuPG v1.4.1 (GNU/Linux)

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jVxDtsvzMnjdjtj0EwMqevk=
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[Asterisk-Users] chan_capi.so: undefined symbol: ast_smoother_feed

2005-10-19 Thread asterisk

Hello,

i have installed Asterisk Version 1.2 from source on Sarge with a 2.6.8
Kernel. Then i did a apt-get install libcapi20-2.

When i start asterisk, i get this error:

[ Booting...Oct 19 13:13:47 NOTICE[18363]: cdr.c:1160 do_reload: CDR
simple logging enabled.
...Oct 19 13:13:47 WARNING[18363]: res_musiconhold.c:813 moh_register:
Unable to open pseudo channel for timing...  Sound may be choppy.
...Oct 19 13:13:48 WARNING[18363]: chan_iax2.c:9355 load_module:
Unable to open IAX timing interface: No such device or address
...Oct 19 13:13:48 WARNING[18363]: loader.c:314 __load_resource:
/usr/lib/asterisk/modules/chan_capi.so: undefined symbol:
ast_smoother_feed
Oct 19 13:13:48 WARNING[18363]: loader.c:543 load_modules: Loading module
chan_capi.so failed!



Is my capi module too old?

Thanks, Mario

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Re: [Asterisk-Users] Problems Calling PSTN PSTN FROM ASTERISK


 I terminated a call through SIP to a landphone i have the following 
 problems.
 
 1.) asterisk gives a fake riming tone, it does not give the real tone from 
 the phone company.
 
 2.) when I put the call on hold the on hold music is not very clear.
 but when I talk the call quality is very clear.
 
 if any of you guys have come across this please let me know what I did 
 wrong.

You probably should do a little bit of research via google and the wiki.

To get rid of the fake ring tone, do not use the r option in your
Dial statement.

For the music on hold problem, look around in your phone config (not
asterisk) for an option to disable silence suppression. Asterisk wants
a continuous flow of rtp traffic and the suppress silence does not
provide that. That's why things sound clear when you are talking.


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RE: [Asterisk-Users] Terrible echo with Te110P and Adit 600


 Yupgot one running at home thanks to your WIKI.  But for clients
 moving forward, I need something a bit more mainstream.  I'm
 disappointed that the TE110P + adit 600 has been an issue on multiple
 systems now, and that the software echo canceller has been a major
 failure.  
 
 It makes that solution WAY to expensive with the echo
 cancellerthat's well into the 2k range, and a good FXO - SIP gateway
 with echo canceling is significantly less than that.  
 
I don't have any specific suggestions for you other then to say that
_lots_ of other implementors have that specific combination of T1
card and Adit 600 working just fine, therefore there has to be something
messed up with your config, options, code, or something.

I don't have a 600 to suggestion options, but I'm sure others on this
list might be able to help.


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[Asterisk-Users] Asterisk management portal

2005-10-19 Thread Tomislav Parčina

Does anybody have detailed instruction how to Install AMP? I have tried to 
install it using Installation Guide on their pages but I'm unable to satisfy 
AMP's PERL module dependencies.

Thank you for your time.


Tomislav
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Re: [Asterisk-Users] Terrible echo with Te110P and Adit 600

2005-10-19 Thread steve



On Tue, 18 Oct 2005, Matt wrote:

 try sangoma card which has a very good echo cancel solution.


Huh?  I believe that the Sangoma uses the same zaptel echo cancellers that 
are used with the Digium cards.

Steve

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Re: [Asterisk-Users] more dids added to goiax.com

On Wed, 2005-10-19 at 07:09 -0400, Steve Totaro wrote:
 Or even use it?  What is the point of having a DID that changes all the
 time?
 

When I was younger this type of thing would have been just what my
friends would have wanted.  The ability to have a temporary disposable
anonymous number for call details pulled from various, albiet stupid,
telephone companies.

Now ebay scams are a bigger threat for this type of DID.  Someone gets
one to look like a legit seller, and even is 'tracable' as a seller.
Most people think that a person is 100% tracable if they have their
phone number, they only have to call the cops and give the number to
them.  goiax proves that wrong, as do prepaid mobile phones where cash
was the only thing used.  I know about the patent case where prepaid
mobile service is patented in the US and now illegal to do without
paying the $0.0025/min tax to the patent holder, but there has to be a
way around that and given the financial risks cingular (go phone),
tmobile, sprint-nextel (boost) and others are facing they will find it
rather than give up prepaid all together.

Scammers will use any means possible to get what they want, and
disposable DIDs are a tool they can use and just walk away from leaving
virtually no trace of who they really are (think war driving coupled
with illegal access to open networks found coupled with disposable free
DIDs).

  Perhaps expand the registration process to include name, address and
  another phone number?
 
 And this would accomplish what?  I have many aliases that I use on websites
 if I suspect SPAM or just dont feel like giving my details.
 

Not to mention the cost of verification of all of that information.
That would make goiax a headache for anyone who operates it (and right
now I think its just matthew).  Information is useless unless verified.

  Don't make everyones life too complicated simply to discourage a (
  hopefully ) small number of abusers.
 
 If you think your life will be too complicated because you have to jump
 through a couple hoops to use a FREE service, just think what happens when
 someone files a complaint with the authorities because the owner's circuit
 was used to call in threats.  Better yet, a terror threat and Homeland
 Security swoops in with all the powers of the Patriot Act.
 

Look at the pranks that have occured to PSAPs in the past few years.
There have been many news stories about swat teams being deployed
because of a prank to the PSAP (usually through the direct dial not
911).  


 Mathew, don't be too generous here.  Protect yourself FIRST.  In all
 reality, if I wanted to contact a sleeper cell in the US and not be worried
 about Echelon or Carnivore, I would use your service through a proxy or a
 pay cash at an internet cafe.  The number of abusers is not the issue, it
 only takes one to cause some real nightmares.
 

You could never stop something like that.  If it were to coordinate an
attack you wouldnt always know who was going to place the call
beforehand.  Those types are typically better funded anyway, they could
at least spring for a $5 prepaid calling card and a public phone, or a
mobile or ...  quite often they seem to prefer mobiles as it gives them
the ability to choose where they speak from, with voip you are limited
to places where there is internet.  No phone provider, free or otherwise
has ever been charged because, unknown to them, someone used the system
for illegal purposes, it wont start now.

I would be more concerned with someone using it for telemarketing
purposes or for resale.  That would seem to drive the minute usage up
faster and create more havok.


 Maybe you should even consider recording calls for your own protection.
 Just play message to both the caller and the callee to inform them of the
 recording.
 

The storage capacity needed, extra cpu requirements, and fact that it
would drive people away from using it would be a problem there.  Phone
companies arent required to record 'just in case', and I wouldnt want
the legal liability of having those recordings.  What happens when there
is a divorce or something?  The recordings get subpoenaed.  There would
be a large volume of people asking for those recordings, and having them
creates a liability to produce them (often at your own cost).


-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378


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[Asterisk-Users] Help with Dial Plan

2005-10-19 Thread Dave Morrow

Title: Help with Dial Plan

Hi all. So far this list is proving it's worth, even on my first day using it!

I hope that someone might know an easy solution to this one.

I would like to create a dial plan which will allow me to have all extensions 6XXX cause a dial-out of my T1 interface (TE110P) to a local number, wait for an answer, wait 2 seconds and then enter the extension. Can I do this in a dial plan somehow? This will allow me to pseudo-integrate a legacy telephone switch (whose extensions are all 6XXX) to my Asterisk system for direct extension dialing.

David A. Morrow

Technical Systems Lead

Autodata Solutions Company

[EMAIL PROTECTED]

http://www.autodata.net

Tel: (519) 951-6079

Fax: (519) 451-6615

Poor planning on your part does not necessarily constitute an emergency on my part!

This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]

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Re: [Asterisk-Users] Call queuing question

2005-10-19 Thread Lenz


Hello,
if you use a mechanism like agents, * will know that there is nobody at  
the first level of penalty and route the call to the other level. A  
different approach could be to have a queue ring A for say 20 second,  
timeout, route the call to a second queue where B and C are. This should  
fix yoiur problem.

Bye
l.



On Wed, 19 Oct 2005 11:49:16 +0200, Peter Spikings  
[EMAIL PROTECTED] wrote:



Hi,

Could I have clarification on the logic in app_queue which treats no
answer as needing a retry? What I want to do is have all calls firstly
always go to phone A, then if there is no answer make it call B or C in
a round robin fashion. The obvious thing to do is put a penalty on B  C
but then if phone A doesn't pick up it just keeps retrying which isn't
what I want as the person with phone A on their desk may be absent for a
couple of minutes. Could I ask why no answer is treated as needing a
retry rather than moving up to the next penalty group?

Thanks,

Peter Spikings.

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[Asterisk-Users] Re: PRI echo issues: solvable?

2005-10-19 Thread Doug Meredith

Andrew Kohlsmith [EMAIL PROTECTED] wrote:

On Tuesday 18 October 2005 12:18, Doug Meredith wrote:
 Andrew Kohlsmith [EMAIL PROTECTED] wrote:
 I've never seen that, it's always when we call out.  Certain numbers will
 always trigger it.  888-737-4787 (IPC Resistors, it dumps into an IVR so
  it's safe to call) is one such number, but we have local numbers that hit
  other

 I just tried this number, and it was answered by a person.

It's IVR most of time time.  :-)  Did you hear echo?

No, no echo.  But I have an analog PSTN connection, not PRI.

Doug
-- 
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SystemGuard - Oracle remote support
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Re: [Asterisk-Users] Asterisk on Slackware ...

2005-10-19 Thread Support




Dear Matt,

Thanks a lot for the asterisk book link. I am 
currently reading through the book.
I'm also wondering that will it be able to use the 
FXO and FXS ports from Cisco 1760 router or is there some integrations to use 
with Cisco routers for Voice?

Regarding to the softphones, I will be using on 
Windows and I am new to both of SIP and IAX. Which is the best and easier to 
deploy?

Thanks again for your kind help and 
suggestions.

Arnold.


  - Original Message - 
  From: 
  Matt Florell 
  
  To: Asterisk Users Mailing List - 
  Non-Commercial Discussion 
  Sent: Wednesday, October 19, 2005 5:37 
  PM
  Subject: Re: [Asterisk-Users] Asterisk on 
  Slackware ...
  Hello,We have 12 servers in production that run 
  Asterisk on Slackware Linux they run beautifully. If you are starting 
  out in Asterisk I suggest reading the recently completed Asterisk book(it's 
  free in electronic format):http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11It's 
  the best place to start.As for softphones, what operating system will 
  your users be on? will the softphones need to be SIP or 
  IAX?MATT---
  On 10/19/05, Support [EMAIL PROTECTED] wrote:
  
Does anybody installed Asterisk on 
Slackware?
It seems the installation went ok. 

But which config files do I have to look and 
edit first for the testing on two internal peers.
Which free version of VoIP softphone is the 
best to use with asterisk?

Thanks ..
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Re: [Asterisk-Users] Terrible echo with Te110P and Adit 600

2005-10-19 Thread Patrick

On Wed, 2005-10-19 at 13:32 +0200, [EMAIL PROTECTED] wrote:
 
 On Tue, 18 Oct 2005, Matt wrote:
 
  try sangoma card which has a very good echo cancel solution.
 
 
 Huh?  I believe that the Sangoma uses the same zaptel echo cancellers that 
 are used with the Digium cards.

Unless you have the Sangoma card with the hardware echo can on board.

Regards,
Patrick
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Re: [Asterisk-Users] Terrible echo with Te110P and Adit 600

2005-10-19 Thread steve



On Wed, 19 Oct 2005, Patrick wrote:

 
 Unless you have the Sangoma card with the hardware echo can on board.
 

So am I right in saying that the normal Sangoma uses the standard Zaptel
software echo canceller - the same one that the Digium board uses?

That's been my understanding, but people seem to keep popping up on the
list giving a different impression, leading me to think that I must have
missed something.

Tell me, are the Sangoma with hardware echo cancellation shipping?  They 
are not yet shown on the website.

Regards,
Steve
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Re: [Asterisk-Users] Call queuing question

2005-10-19 Thread Peter Spikings

Hi,

Using agents would involve the user having to remember to login again
every time they leave their desk as it would only be useful if they were
auto-logged off ;)

I've tried playing with the timeouts and have found that the timeout
parameter to queue causes it to return to the dialplan after that long
(as if the n option was set, which it wasn't). The timeout parameter on
the queue moves onto the next phone at the same penalty and never
advances to the next penalty. It's the latter behaviour that I find
puzzling, surely a timeout when the strategy is ringall or all other
phones at the current penalty have also timed out should make it advance
to the next penalty 

Cheers,

Peter.

On Wed, 2005-10-19 at 13:50 +0200, Lenz wrote:
 Hello,
 if you use a mechanism like agents, * will know that there is nobody at  
 the first level of penalty and route the call to the other level. A  
 different approach could be to have a queue ring A for say 20 second,  
 timeout, route the call to a second queue where B and C are. This should  
 fix yoiur problem.
 Bye
 l.
 
 
 
 On Wed, 19 Oct 2005 11:49:16 +0200, Peter Spikings  
 [EMAIL PROTECTED] wrote:
 
  Hi,
 
  Could I have clarification on the logic in app_queue which treats no
  answer as needing a retry? What I want to do is have all calls firstly
  always go to phone A, then if there is no answer make it call B or C in
  a round robin fashion. The obvious thing to do is put a penalty on B  C
  but then if phone A doesn't pick up it just keeps retrying which isn't
  what I want as the person with phone A on their desk may be absent for a
  couple of minutes. Could I ask why no answer is treated as needing a
  retry rather than moving up to the next penalty group?
 
  Thanks,
 
  Peter Spikings.
 
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Re: [Asterisk-Users] Terrible echo with Te110P and Adit 600


  Unless you have the Sangoma card with the hardware echo can on board.
  
 
 So am I right in saying that the normal Sangoma uses the standard Zaptel
 software echo canceller - the same one that the Digium board uses?
 
 That's been my understanding, but people seem to keep popping up on the
 list giving a different impression, leading me to think that I must have
 missed something.
 
 Tell me, are the Sangoma with hardware echo cancellation shipping?  They 
 are not yet shown on the website.

If you go back through some of the recent -user list postings, you'll
find comments relative to Sangoma's T1 card having an onboard echo can
(either currently or near ready for shipment).

Same with a TDM work-a-like analog card that supposedly will support up
to something like eight analog lines using multiple pci slots. It kind
of sounded like one reseller began discussing/advertising it before
Sangoma was ready to release it for general knowledge.


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Re: [Asterisk-Users] Help with Dial Plan

2005-10-19 Thread steve



On Wed, 19 Oct 2005, Dave Morrow wrote:

 Hi all. So far this list is proving it's worth, even on my first day
 using it!  I hope that someone might know an easy solution to this one.  
 I would like to create a dial plan which will allow me to have all
 extensions 6XXX cause a dial-out of my T1 interface (TE110P) to a local
 number, wait for an answer, wait 2 seconds and then enter the extension.
 Can I do this in a dial plan somehow? This will allow me to
 pseudo-integrate a legacy telephone switch (whose extensions are all
 6XXX) to my Asterisk system for direct extension dialing.

exten = _6XXX,1,Dial(Zap/gX/1234567890,60,D(,,${EXTEN}))

where:
gX needs to become the group of the channels of your T1,
1234567890 is the number of your legacy system.
60 is the dial timeout

You may need to adjust the number of commas to get the right delay.

Hope that helps,
Steve

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Re: [Asterisk-Users] Terrible echo with Te110P and Adit 600

2005-10-19 Thread Patrick

On Wed, 2005-10-19 at 14:06 +0200, [EMAIL PROTECTED] wrote:
 
 On Wed, 19 Oct 2005, Patrick wrote:
 
  
  Unless you have the Sangoma card with the hardware echo can on board.
  
 
 So am I right in saying that the normal Sangoma uses the standard Zaptel
 software echo canceller - the same one that the Digium board uses?
 
 That's been my understanding, but people seem to keep popping up on the
 list giving a different impression, leading me to think that I must have
 missed something.
 
 Tell me, are the Sangoma with hardware echo cancellation shipping?  They 
 are not yet shown on the website.

I never heard of Sangoma using a different software echo can can but
maybe I was not paying enough attention :) Perhaps they stuck something
in their drivers? 

Wrt the echo can board rumours are October 24th.

Regards,
Patrick
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[Asterisk-Users] How can I signal a flash to PABX ...

2005-10-19 Thread Mauro Zanin

Hi everybody,
I have an Asterisk box connected locally to a PABX via analog and BRI
extensions. Some remote VOIP phones are remotelly connected to this box,
which acts as an IP gateway or better a remote PABX (analog extension 1 is
connected to VOIP phone 1, extension 2 to VOIP 2, and so on.)
The problem is: how can I signal flash to the PABX, so I can forward the
call via PABX to another inner extension? This is why I need ACD suppport
from PABX and it must be aware of every change in connections.

Best regards and thanks!
Mauro Zanin
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[Asterisk-Users] Fw: asterisk shutting down...

2005-10-19 Thread Dov Bigio




Hi,

Got the following messages log tonight... and 
Asterisk was down until I manually restarted it...

Any ideas?

Thank you
Dov

Oct 19 03:40:18 WARNING[28005]: Avoided deadlock 
for 'SIP/raphael.pavanelli-f40b', 10 retries!Oct 19 03:40:28 NOTICE[28005]: 
Still have a call...Oct 19 03:40:50 NOTICE[28005]: PRI got event: HDLC Bad 
FCS (8) on Primary D-channel of span 1Oct 19 03:40:50 NOTICE[28005]: PRI got 
event: HDLC Bad FCS (8) on Primary D-channel of span 1Oct 19 03:44:21 
NOTICE[28005]: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 
1Oct 19 03:44:59 WARNING[28005]: Avoided initial deadlock for 
'SIP/marcelo.araujo-0241', 10 retries!Oct 19 03:45:31 NOTICE[28005]: PRI got 
event: HDLC Bad FCS (8) on Primary D-channel of span 1Oct 19 03:46:41 
NOTICE[28005]: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 
1Oct 19 03:46:51 NOTICE[28005]: PRI got event: HDLC Bad FCS (8) on Primary 
D-channel of span 1Oct 19 03:47:46 WARNING[28005]: Maximum retries exceeded 
on call [EMAIL PROTECTED] 
for seqno 102 (Non-critical Request)Oct 19 03:47:49 WARNING[28005]: Maximum 
retries exceeded on call [EMAIL PROTECTED] 
for seqno 102 (Non-critical Request)Oct 19 03:47:57 WARNING[28005]: Maximum 
retries exceeded on call [EMAIL PROTECTED] 
for seqno 102 (Non-critical Request)Oct 19 03:48:00 WARNING[28005]: Maximum 
retries exceeded on call [EMAIL PROTECTED] 
for seqno 102 (Non-critical Request)Oct 19 03:48:01 NOTICE[28005]: PRI got 
event: HDLC Bad FCS (8) on Primary D-channel of span 1Oct 19 03:49:28 
NOTICE[28005]: Still have a call...Oct 19 03:50:12 NOTICE[28005]: PRI got 
event: HDLC Bad FCS (8) on Primary D-channel of span 1Oct 19 03:50:12 
NOTICE[28005]: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 
1Oct 19 03:50:32 NOTICE[28005]: PRI got event: HDLC Bad FCS (8) on Primary 
D-channel of span 1Oct 19 03:53:53 NOTICE[28005]: PRI got event: HDLC Bad 
FCS (8) on Primary D-channel of span 1Oct 19 03:53:56 WARNING[28005]: 
Avoided deadlock for 'SIP/raphael.pavanelli-f40b', 10 retries!Oct 19 
03:54:54 WARNING[28005]: Avoided deadlock for 'SIP/alexandre.catao-d9b5', 10 
retries!Oct 19 03:55:03 NOTICE[28005]: PRI got event: HDLC Bad FCS (8) on 
Primary D-channel of span 1Oct 19 03:55:03 NOTICE[28005]: PRI got event: 
HDLC Bad FCS (8) on Primary D-channel of span 1Oct 19 03:56:13 
NOTICE[28005]: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 
1Oct 19 03:56:13 NOTICE[28005]: PRI got event: HDLC Bad FCS (8) on Primary 
D-channel of span 1
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[Asterisk-Users] SIP to IAX

2005-10-19 Thread Frank Kostin

Hello everybody,
Is it possible to route any incoming SIP call
(without authentication - register) from an Asterisk A
to a remote Asterisk B(throught IAX2), transparently ?
Otherwise said, I would like to pass any incoming SIP
call from Asterisk A to Asterisk B without SIP need to
be registered, like a phone call in zap.
I would apreciate any hint,
Thanks,
Frank




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Re: [Asterisk-Users] SIP to IAX

YES

- Original Message - 
From: Frank Kostin [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, October 19, 2005 8:58 AM
Subject: [Asterisk-Users] SIP to IAX 


Hello everybody,
Is it possible to route any incoming SIP call
(without authentication - register) from an Asterisk A
to a remote Asterisk B(throught IAX2), transparently ?
Otherwise said, I would like to pass any incoming SIP
call from Asterisk A to Asterisk B without SIP need to
be registered, like a phone call in zap.
I would apreciate any hint,
Thanks,
Frank




__ 
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Re: [Asterisk-Users] Terrible echo with Te110P and Adit 600

2005-10-19 Thread Darren Nickerson


[EMAIL PROTECTED] wrote:


Tell me, are the Sangoma with hardware echo cancellation shipping?  They
are not yet shown on the website.


The A104D begins shipping Monday and authorized Sangoma resellers are 
already accepting orders. See:


http://shop.ifax.com/product_info.php?cPath=32_33products_id=124

-Darren

--
Darren Nickerson
Senior Sales  Support Engineer
iFAX Solutions, Inc. www.ifax.com
[EMAIL PROTECTED]
+1.215.438.4638 x8106
+1.215.243.8335 (fax) 


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[Asterisk-Users] SIP CallerID

2005-10-19 Thread Dave Wise


I am using a * w/a PRI for the TDM interface to telco.
I am running Asterisk CVS-HEAD-05/29/05-03:59:44
All was working well until I needed a SIP ATA to be unlisted.

in sip.conf, on the account I used:
restrictcid=yes

I am getting the callerID through though.  I know that the ANI needs to 
be passed (for telco's to accept the call, for thier records) so I also 
tried assigning contexts to the account so I could modify the presentation.


[internationalul]
exten =s,1,SetCallerPres(prohib)
exten =s,2,Goto(international,${BYEXTENSION},1)

or with this context It would need to be done on a phone by phone basis 
(but none of these worked).


[3215559876outbound]
exten=s,1,set CallerID(UnKnown3215559876,A)
exten=s,2,Ser CallerID(UnKnown)
exten=s,3,Goto(international,${BYEXTENSION},1)

on the wiki, for Asterisk+sip.conf it says:
* restrictcid 
http://www.voip-info.org/wiki/edit.php?page=Asterisk+sip+restrictcid*: 
(yes/no) To have the callerid restricted - sent as ANI; use this to 
hide the caller ID. This does not seem to work.


Any ideas?  Is there a patch for this or does a newer version of head 
fix this?



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RE: [Asterisk-Users] Asterisk Redundency

2005-10-19 Thread Benjamin Lawetz

 
 Since I can't do that, what I've settled on is heartbeat + mon.  
 Heartbeat will monitor for a system level failure and switch to the backup
machine if neccesary; and mon will watch the asterisk (or any
 other) service and restart it and/or alert me if it fails.

What kind of monitor are you using to monitor asterisk?


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[Asterisk-Users] DNIS/DNID

2005-10-19 Thread James Steven




Hi
Is it possible with 
Asterisk to tellthe called party which number was dialled by the 
caller? Or in place of the number dialled have a description such as 
'Sales' or 'Accounts'? Ideally, I would like to show a description 
corresponding to the number dialled followed by CIDName. 
How might this be set up? 

Currently my 
extensions.conf is:

exten = 
xx,1,LookupCIDNameexten = xx,2,Dial(SIP/xx,50)exten = 
xx,3,Voicemail(xx)exten = xx,4,Hangup

Thanks for your 
help.

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[Asterisk-Users] Caller ID

2005-10-19 Thread Michael J. Lynch


If this question has an obvious answer forgive me, I'm a noob.  I'm
planning to make a system configured as below:


POTS --- FXO (400P) -- Asterisk --- FXS (400P)  Analog phone

The question I have is, if an incoming call from the POTS line has
caller ID information, does/is/can that information be passed onto
the analog phone so it's caller id display will show the info?  If
so, is there anything I need to do to make this happen or does it
*just work*?  Thanks.


--
Michael J. Lynch

What if the hokey pokey IS what it's all about -- author unknown

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Re: [Asterisk-Users] Caller ID

It just works has been my experience.

Thanks,
Steve

- Original Message - 
From: Michael J. Lynch [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, October 19, 2005 9:31 AM
Subject: [Asterisk-Users] Caller ID


 If this question has an obvious answer forgive me, I'm a noob.  I'm
 planning to make a system configured as below:


 POTS --- FXO (400P) -- Asterisk --- FXS (400P)  Analog phone

 The question I have is, if an incoming call from the POTS line has
 caller ID information, does/is/can that information be passed onto
 the analog phone so it's caller id display will show the info?  If
 so, is there anything I need to do to make this happen or does it
 *just work*?  Thanks.


 -- 
 Michael J. Lynch

 What if the hokey pokey IS what it's all about -- author unknown

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Re: [Asterisk-Users] DNIS/DNID




exten = xx,2,SetCIDName(*-SALES-* 
${CALLERIDNAME})

  - Original Message - 
  From: 
  James Steven 
  To: asterisk-users@lists.digium.com 
  
  Sent: Wednesday, October 19, 2005 9:33 
  AM
  Subject: [Asterisk-Users] DNIS/DNID
  
  Hi
  Is it possible 
  with Asterisk to tellthe called party which number was dialled by the 
  caller? Or in place of the number dialled have a description such as 
  'Sales' or 'Accounts'? Ideally, I would like to show a description 
  corresponding to the number dialled followed by CIDName. 
  How might this be set up? 
  
  Currently my 
  extensions.conf is:
  
  exten = 
  xx,1,LookupCIDNameexten = xx,2,Dial(SIP/xx,50)exten = 
  xx,3,Voicemail(xx)exten = xx,4,Hangup
  
  Thanks for your 
  help.
  
  
  

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Re: [Asterisk-Users] Problem with select correct network interface (oh323)

2005-10-19 Thread Moises Silva

it seems to me that your problem is not Asterisk configuration, but
iproute configuration. Look in google about iproute and kernel routing
tables.
In order to help you, it would be desireable to know how are you dialing.

Best Regards.On 10/19/05, Oleh Mukha [EMAIL PROTECTED] wrote:
i build asterisk on pc with 3 network inerfaceeth0 (yyy.yyy.yyy.yyy) main public ipeth1 (xxx.xxx.xxx.xxx) seconf public ip used only for voip connectioneth2 (zzz.zzz.zzz.zzz) local ipi config oh323 to bind eth1 interface
i try make callfrom my local network - Asterisk - provider h323when i try to call from ata 186 throught my astersik oh323 moduleasetrisk resive calls from ata but send it to my oh323 providet not from eth1
(with ip xxx.xxx.xxx.xxx) or from eth0 (ip yyy.yyy.yyy.yyy)how can i tel asterisk send data from me to my provider from eth1 (ipxxx.xxx.xxx.xxx)Oleh MukhaIClub380322722738
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Re: [Asterisk-Users] Caller ID

2005-10-19 Thread Nathan Pralle




The question I have is, if an incoming call from the POTS line has
caller ID information, does/is/can that information be passed onto
the analog phone so it's caller id display will show the info?  If
so, is there anything I need to do to make this happen or does it
*just work*?  Thanks.


It should just work.  You can modify it if you want, but if you don't, 
it should just pass it on through.


Nathan


--
-
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Re: [Asterisk-Users] more dids added to goiax.com

2005-10-19 Thread John Novack







Steve Totaro wrote:

  
  
  
I made two attempts this morning to send some comments off list, and both got returned due to some sort of spam filter, so would hope that any future controls will not suffer from that inability to communicate.

  
  
Maybe if you posted what the returned email said then he could remove or alter that filter.  "both got returned due to some sort of spam filter", doesn't help anyone.

  

I saw no reason to clog up this list with details that aren't Asterisk
related.
If he wants further information he can communicate directly with me.

JN


  
  



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Re: SV: [Asterisk-Users] Queues and call waiting indication

2005-10-19 Thread Tom Rymes


On Oct 18, 2005, at 9:08 AM, Adam Goryachev wrote:


On Tue, 2005-10-18 at 14:35 +0200, [EMAIL PROTECTED] wrote:


 Hi,

This issue has been discussed probably a million times on every  
asterisk forum in the world and I have the same problem too.  
Another problem you would have with the agents is that when they  
make an outgoing call they are not regarded as busy by asterisk  
and it sends more calls to the agent if it has call waiting enabled.


This behaviour is totally senseless since the whole purouse of  
queues is to _queue_ the callers until the agent is available.  
available usually means not on the phone -- whether or not  
it's an incoming or outgoing call.


I solved this problem by using single-line clients and phones  
where you can turn off call wating.




Actually this can simply be solved in your dialplan Just use the
setgroup/checkgroup values, and use the AgentCallbackLogin instead of
AgentLogin 

This is what I used, and it seems to work quite well so far... well, I
haven't actually added the bits for the outbound calls yet on my own
system, but I've done it on others, and they seem to be quite happy  
with

it...


Can you provide some more specifics? Maybe an example for the  
dialplan? Does this keep the queue from sending multiple calls to  
agents who have call waiting enabled?


Tom
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[Asterisk-Users] Connection question

2005-10-19 Thread Joao Carneiro - DLS

Asterisk seems to be a very good peace of software, but i am interested
to know if i can use plain ISDN cards with it, i mean use the isdn cards
as a passthrough device between my alcatel pbx and voip users.

thanks 


DLS - Projectos, Automação e Manutenção, Lda. 
João Carneiro, Tecnico 
Dep. Sistemas de Informação 
Rua da Boavista S/N - P.O.Box 313
4416-901 Grijó 
www.dls.pt

Email: [EMAIL PROTECTED]

Tel : +351 227 470 786

Fax : +351 227 470 787

Tlm : 




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Re: [Asterisk-Users] Voicemail as an email attachement

2005-10-19 Thread [EMAIL PROTECTED]

Yes. I am interested. I will make provisions for the upload. How big are 
the files?


Thanks

BEN

Goran Skular wrote:

I changed my app_voicemail.c to work not with sendmail but with sendEmail
that connects to any SMTP and sends email with attachment...

It's dirty, but it works.

If you are interested I can upload app_voicemail.c and sendEmail package
somewhere..




I have configured the voicemail.conf file as per the wiki to email
voicemails as an attachment. I cannot find any instructions/locations to
set the outgoing server login information. Furthermore, I can get no
emails from asterisk. Can anyone point me to the next step to setup the
attachment of voicemail messages to an email?

Thanks

BEN
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Re: [Asterisk-Users] Recomendations for utility togenerateAsteriskconfiguration

2005-10-19 Thread Tom Rymes

On Oct 19, 2005, at 11:04 AM, asterisk wrote: AMP's dialplan and setup is quite complex.   Requires, e.g, a number of  AGIs.This is normally not the type of thing you'd like   to hand-edit later  after the initial adaptation to the target   system.Who said anything about hand editing?That is why you would want to keep the old computer running [EMAIL PROTECTED].  Insteadof hand editing anything, make the changes on the [EMAIL PROTECTED] box's AMP GUI andcopy them over again.  Very simple and most tech folks have an old computerlaying around somewhere that could be put to use.  Why wouldn't you just install [EMAIL PROTECTED] on your main server then? Why install a   second server and go through the trouble of using scp to copy files back and   forth?Tom     Maybe because you snipped the beginning of the   thread without reading the entire thread's context, but he is running on   Solaris.  I am not sure what all is involved with installing [EMAIL PROTECTED] on solaris but I assume it is no trivial   task.  WinSCP is very trivial IMHO and there is no "copying files back   and forth", just one direction, takes about twenty seconds and maybe 30 if you   are slow.  Now you also have an almost hot swap server in case the   Solaris machine goes down, just swap IP addresses and hardware.OK, my bad, but the point is still valid. If you are dead set on running Solaris, install AMP on the Solaris server, don't go to the trouble of creating a second machine to generate your configuration when you can just eliminate the extra steps and create the config on the main machine. It would be simpler to set up, maintain, and make testing config changes much easier and faster.Tom___
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Re: [Asterisk-Users] Polycom IP501 and record on demand

2005-10-19 Thread Matthew T. O'Connor


Matt Gibson wrote:
You could also take a look at features.conf, and use ** for blind 
transfers, ## for attended transfers, *0 for recording, and *1 to hangup.


I haven't tried mapping them to polycom buttons, but there was 
recently a discussion about that, just this week you can search the 
archives. 


There was a discussion (of which I was a part of) however there was no 
resolution.  I have not found any good documentation on how to remap 
Polycom buttons.  At this point I'm willing to pay for some help.  
Anybody got some better info on this?


Thanks,

Matthew O'Connor


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RE: [Asterisk-Users] DNIS/DNID

2005-10-19 Thread James Steven




That worked great. Thanks


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Steve 
TotaroSent: 19 October 2005 14:45To: Asterisk Users 
Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] 
DNIS/DNID

exten = xx,2,SetCIDName(*-SALES-* 
${CALLERIDNAME})

  - Original Message - 
  From: 
  James Steven 
  To: asterisk-users@lists.digium.com 
  
  Sent: Wednesday, October 19, 2005 9:33 
  AM
  Subject: [Asterisk-Users] DNIS/DNID
  
  Hi
  Is it possible 
  with Asterisk to tellthe called party which number was dialled by the 
  caller? Or in place of the number dialled have a description such as 
  'Sales' or 'Accounts'? Ideally, I would like to show a description 
  corresponding to the number dialled followed by CIDName. 
  How might this be set up? 
  
  Currently my 
  extensions.conf is:
  
  exten = 
  xx,1,LookupCIDNameexten = xx,2,Dial(SIP/xx,50)exten = 
  xx,3,Voicemail(xx)exten = xx,4,Hangup
  
  Thanks for your 
  help.
  
  
  

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RE: [Asterisk-Users] Polycom IP501 and record on demand

2005-10-19 Thread Jonathan k. Creasy

I probably can't provide any better information for you, however, have
you looked through the Polycom configuration files. The button mappings
are there. I haven't spent much time with it so I can not attest to what
you can map them to do. 

Hope this helps you a little. 

-Jonathan

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew T.
O'Connor
Sent: Wednesday, October 19, 2005 10:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Polycom IP501 and record on demand

Matt Gibson wrote:
 You could also take a look at features.conf, and use ** for blind 
 transfers, ## for attended transfers, *0 for recording, and *1 to
hangup.

 I haven't tried mapping them to polycom buttons, but there was 
 recently a discussion about that, just this week you can search the 
 archives. 

There was a discussion (of which I was a part of) however there was no 
resolution.  I have not found any good documentation on how to remap 
Polycom buttons.  At this point I'm willing to pay for some help.  
Anybody got some better info on this?

Thanks,

Matthew O'Connor


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[Asterisk-Users] Trunk Dialing rules

2005-10-19 Thread bails

Hi i have posted before about this problem, and have had several 
suggestions, that i can use contexts to overcome this.


The situation. [EMAIL PROTECTED] 1.5

I have 3 sets of users say sales, admin and tech with the numbers

sales200 201
admin 202 203
tech   204 205

They all need to be able to ring each other hence they are all in 
[ext-local]


Each group has its own trunk, which is unavailable to other users, at 
this point I am failing.  If i make say sales members of [ext-local-1], 
admin members of [ext-local-2] and tech members of [ext-local-3], they 
cannot call each other until i add then to


[ext-local]
include = ext-local-1
include = ext-local-2
include = ext-local-3

then of course thay can call each other, the trouble is they can then 
call [all-routes-outbound] which is not what i want.


if i remove [all-routes-outbound]
noone can call out over the trunks.

so i create
[from-internal-local-1]
include = ext-local-1
include = outbound-outrt-1  ;iax route out 1

and
[from-internal]
include = from-internal-local-1

However this means that anyone can dial out over outbound-outrt-1

Which is what i was trying to avoid in the outset.

Is this possible with [EMAIL PROTECTED]

if so how? (my head is bruised from repeatedly banging it against the wall)

wishlist
I would love to have this funtionality available from the amportal, 
something like add extensions totrunks.

/wishlist

Thanks

Bails
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[Asterisk-Users] E1 PRI error: !! Got I-frame while link state 2 and !! Got a UA, but i'm in state 1 (long)

2005-10-19 Thread Dinesh Nair




 Original Message 
Subject: E1 PRI error: !! Got I-frame while link state 2 and !! Got a 
UA, but i'm in state 1

Date: Wed, 19 Oct 2005 23:46:01 +0800
From: Dinesh Nair [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com, Asterisk on BSD discussion 
asterisk-bsd@lists.digium.com


hey * folk,

i've got a TE410P (generation 1 firmware) stuck in a box with a single xeon
2.8Ghz and 1GB RAM. there's a loopback E1 cable connecting span 1 to span 4
(zaptel.conf and zapata.conf below). upon starting up asterisk, i see the
following errors consistently on the screen,

!! Got I-frame while link state 2
!! Got a UA, but i'm in state 1

they seem to be coming from libpri.so.1 and the spans seem to be restarting
each other infinitely. i also get a number of the following messages from
chan_zap.so:

B-channel 0/6 restarted on span 1
B-channel 0/6 restarted on span 4
B-channel 0/7 restarted on span 1
B-channel 0/7 restarted on span 4
B-channel 0/8 restarted on span 1
B-channel 0/8 restarted on span 4
B-channel 0/9 restarted on span 1
B-channel 0/9 restarted on span 4

No D-channels available! Using Primary Channel 16 as D-channel anyway!
No D-channels available! Using Primary Channel 109 as D-channel anyway!

both spans show Provisioned,Up, Active in pri show span, and zttest shows
100% all the way.

a snapshot of pri debug span 1, shows:

 Protocol Discriminator: Q.931 (8)  len=13
 Call Ref: len= 2 (reference 0/0x0) (Originator)
 Message type: RESTART (70)
 [18 03 a9 83 83]
 Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, Exclusive 
Dchan: 0

ChanSel: Reserved
   Ext: 1  Coding: 0   Number Specified   Channel Type: 3
   Ext: 1  Channel: 3 ]
 [79 01 80]
 Restart Indentifier (len= 3) [ Ext: 1  Spare: 0  Resetting Indicated 
Channel (0) ]

 Protocol Discriminator: Q.931 (8)  len=13
 Call Ref: len= 2 (reference 0/0x0) (Terminator)
 Message type: RESTART ACKNOWLEDGE (78)
 [18 03 a9 83 83]
 Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, Exclusive 
Dchan: 0

ChanSel: Reserved
   Ext: 1  Coding: 0   Number Specified   Channel Type: 3
   Ext: 1  Channel: 3 ]
 [79 01 80]
 Restart Indentifier (len= 3) [ Ext: 1  Spare: 0  Resetting Indicated 
Channel (0) ]

-- Processing IE 24 (cs0, Channel Identification)
-- Processing IE 121 (cs0, Restart Indicator)
 Protocol Discriminator: Q.931 (8)  len=13
 Call Ref: len= 2 (reference 0/0x0) (Originator)
 Message type: RESTART (70)
 [18 03 a9 83 84]
 Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, Exclusive 
Dchan: 0

ChanSel: Reserved
   Ext: 1  Coding: 0   Number Specified   Channel Type: 3
   Ext: 1  Channel: 4 ]
 [79 01 80]
 Restart Indentifier (len= 3) [ Ext: 1  Spare: 0  Resetting Indicated 
Channel (0) ]

!! Got I-frame while link state 2
 Protocol Discriminator: Q.931 (8)  len=13
 Call Ref: len= 2 (reference 0/0x0) (Originator)
 Message type: RESTART (70)
 [18 03 a9 83 83]
 Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, Exclusive 
Dchan: 0

ChanSel: Reserved
   Ext: 1  Coding: 0   Number Specified   Channel Type: 3
   Ext: 1  Channel: 3 ]
 [79 01 80]
 Restart Indentifier (len= 3) [ Ext: 1  Spare: 0  Resetting Indicated 
Channel (0) ]

 Protocol Discriminator: Q.931 (8)  len=13
 Call Ref: len= 2 (reference 0/0x0) (Originator)
 Message type: RESTART (70)
 [18 03 a9 83 84]
 Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, Exclusive 
Dchan: 0

ChanSel: Reserved
   Ext: 1  Coding: 0   Number Specified   Channel Type: 3
   Ext: 1  Channel: 4 ]
 [79 01 80]
 Restart Indentifier (len= 3) [ Ext: 1  Spare: 0  Resetting Indicated 
Channel (0) ]

-- Processing Q.931 Restart
-- Processing IE 24 (cs0, Channel Identification)
-- Processing IE 121 (cs0, Restart Indicator)
 Protocol Discriminator: Q.931 (8)  len=13
 Call Ref: len= 2 (reference 0/0x0) (Terminator)
 Message type: RESTART ACKNOWLEDGE (78)
 [18 03 a9 83 84]
 Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, Exclusive 
Dchan: 0

ChanSel: Reserved
   Ext: 1  Coding: 0   Number Specified   Channel Type: 3
   Ext: 1  Channel: 4 ]
 [79 01 80]
 Restart Indentifier (len= 3) [ Ext: 1  Spare: 0  Resetting Indicated 
Channel (0) ]

!! Got I-frame while link state 2
 Protocol Discriminator: Q.931 (8)  len=13
 Call Ref: len= 2 (reference 0/0x0) (Originator)
 Message type: RESTART (70)
 [18 03 a9 83 85]
 Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, Exclusive 
Dchan: 0

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

Fwd: Re: [Asterisk-Users] IAX only speech one way

2005-10-19 Thread Jerry Richmond

IAX may be as bad as what we are doing?Note: forwarded message attached.---BeginMessage---
Mir wrote:
 Thanks for your suggestion.
 
 Unfortunately, it didnt change anything, A can still not hear B, but B
 can hear A, strange..
 

I had the same problem with one of my IAX providers in AUS.

Both ends turned of trunking and all was fine with the world again.

Not sure what was the cause but that was my solution for EXACTLY the
same problem that you explain.

David



 Michael
 
 2005/10/19, Rich Adamson [EMAIL PROTECTED]:
 
I have two Asterisk's connected via IAX, they are sitting on the same
network, via a VPN, so there should be no problems with firewalls.

My problem is that when a person calls from A to B, A will not hear B
speak. B hears A fine.

I doesn't matter who initiates the call.

One of the Asterisk'ses is a new installation, just installed, but
with the Conf-files from an earlier setup, that worked fine.

Asterisk version on computer A is Asterisk CVS-v1-0-12/09/04-08:58:31
Asterisk version on computer B is Asterisk CVS-D2005.05.28.22.00.00-10/17/05

Two different versions, but I dont think it should matter?

Not sure this applies, but I was having the same problem with teliax.com
and turning off the jitterbuffer in iax.conf fixed the problem. Kind
of looks like we are running two different versions of asterisk as
well, but I'd suspect that teliax has modified their system for
other business purposes.

Try jitterbuffer=no and see if it helps.

Rich


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[Asterisk-Users] Caller-ID via database lookup

2005-10-19 Thread Doug Lytle


Hey everybody,

I'm having issues with one of our facilities, concerning caller-id.   
The system is a Definity that hits a second Definity.  The 2nd Definity 
trunks the call to my Asterisk server via a TE110P.  I can only get 
Caller-ID name.  Nothing in the From: field.  I thought I would be able 
to do a database lookup against name to match extension, but


When doing this and setting Caller-ID number, it still shows on the 
Polycom IP501 as Unknown/Unknown. Dial plan below:


exten = s,1,Set(dnd=${DB(DND/${ARG1})})
exten = s,2,Set(CIDNUMB=${DB(cidname/${CALLERIDNAME})})
exten = s,3,Set(CALLERID(Name)=${CALLERIDNAME})
exten = s,4,Set(CALLERID(Number)=${CIDNUMB})


CLI output below:

CLI -- Accepting AUTHENTICATED call from 192.168.101.10:
   requested format = gsm,
   requested prefs = (),
   actual format = gsm,
   host prefs = (gsm),
   priority = mine
   -- Executing Macro(IAX2/bc-asterisk-16384, sip.extensions|4483|) 
in new stack

   -- Executing Set(IAX2/bc-asterisk-16384, dnd=) in new stack
   -- Executing Set(IAX2/bc-asterisk-16384, CIDNUMB=5574) in new stack
   -- Executing Set(IAX2/bc-asterisk-16384, CALLERID(Name)=Lytle, 
Doug) in new stack
   -- Executing Set(IAX2/bc-asterisk-16384, CALLERID(Number)=5574) 
in new stack

   -- Executing GotoIf(IAX2/bc-asterisk-16384, 0?8:6) in new stack
   -- Goto (macro-sip.extensions,s,6)
   -- Executing SetMusicOnHold(IAX2/bc-asterisk-16384, epi-cd) in 
new stack
   -- Executing Dial(IAX2/bc-asterisk-16384, SIP/4483|28|t) in new 
stack

   -- Called 4483
   -- SIP/4483-04ba is ringing

Debug output from the 'receiving Asterisk' server via IAX below:


   -- SIP/4483-14d3 is ringing
Reliably Transmitting (no NAT) to 192.168.101.64:5060:
CANCEL sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.104.40:5060;branch=z9hG4bK73d230ce;rport
From: Unknown sip:[EMAIL PROTECTED];tag=as23c39fbe
To: sip:[EMAIL PROTECTED]
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0

What variable needs to be set to change it from Unknown to 5574?

Any help would be appreciated.

Doug

--

Ben Franklin quote:

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


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Re: [Asterisk-Users] Voicemail as an email attachement

2005-10-19 Thread Tzafrir Cohen

On Tue, Oct 18, 2005 at 11:22:52PM -0500, Ben Brown wrote:
 I have configured the voicemail.conf file as per the wiki to email 
 voicemails as an attachment. I cannot find any instructions/locations to 
 set the outgoing server login information. Furthermore, I can get no 
 emails from asterisk. Can anyone point me to the next step to setup the 
 attachment of voicemail messages to an email?

Set up a sendmail. Or basically: an MTA. Any linux distro comes with
at least one (postfix seems to be the preffered choice nowadays). Which
one do you use?

There are a bunch of programs that provide /usr/sbin/sendmail but don't
spool the result. Check msmtp, ssmtp, masqmail and nullmailer. There are
probably others.

The downside is that messages that have, for some reason, not been
delivered in the first shot (e.g: due to some transient network error)
will be dropped rather than queued.

-- 
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il |   | a Mutt's  
[EMAIL PROTECTED] |   |  best
ICQ# 16849755 |   | friend
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[Asterisk-Users] Re: Polycom IP501 and record on demand

2005-10-19 Thread Noah Miller


Hi Matthew -


You could also take a look at features.conf, and use ** for blind
transfers, ## for attended transfers, *0 for recording, and *1 to  
hangup.


I haven't tried mapping them to polycom buttons, but there was
recently a discussion about that, just this week you can search the
archives.


There was a discussion (of which I was a part of) however there was no
resolution.  I have not found any good documentation on how to remap
Polycom buttons.  At this point I'm willing to pay for some help.
Anybody got some better info on this?


The best documentation I found is the Polycom manual.  It is fairly  
clear, though they don't provide a lot of examples.  Also, for some  
reason they put the button remapping documentation in one section of  
the manual and the button map in a completely different section.  A  
bit annoying.


I've remapped the transfer key to #, so I can do an asterisk  
unattended transfer using the transfer key.  To do this, I just added  
the following line to my ipmid.cfg (or sip.cfg if you are using  
firmware version 1.5.x or later):


keys key.scrolling.timeout=1  
key.IP_500.37.function.prim=DialpadPound  
key.IP_600.37.function.prim=DialpadPound/


Thanks,
Noah
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RE: [Asterisk-Users] Caller-ID via database lookup

2005-10-19 Thread O'Connor, Jonathan

I have my Definity attached to my Asterisk box with a PRI Trunk.  The
guides and seemingly most people say to use a tie type connection,
however I did not get correct caller-id and setup until I:

1) Set the trunk-group on the Definity to isdn
2) Carrier/Medium to PRI
3) Trunk group numbering format to Public

At this point I had to delete all the 23 ports from the trunk, busy it
out, change to the above, add the ports and release the trunk (a royal
pain).

My Asterisk box just uses:

loadzone= us
defaultzone = us
span=1,0,0,esf,b8zs
bchan=1-23 
dchan=24 



Once I had all of this done it was back to the Definity and into:

change isdn public-unknown-numbering

In mine, trunk group 4 is the Sprint PRIs used for normal calling.
Using 3742 as an example extension:

Ext Len 4
Ext Code37
Trk Grp 4   
CPN Prefix  614791
Ext Len 10

Therefore on trunk 4 it sends a caller id number of 6147913701

To make that send just 4 digits to Asterisk I added entries for:

Ext Len 4
Ext Code37
Trk Grp 1   
CPN Prefix  
Ext Len 4

And when 3742 calls an Asterisk box the Avaya send sonly its 4 digits as
caller ID on trunk 1.


I think the main change is the PRI type instead of tie, my system works
great since I did that, transferring back and forth no problem.  Oddly
tie only worked well with CSUs in place, PRI doesn't seem to care with
just a twist cable

Hope that helped, and didn't confuse you :)


-Jonathan


 
Jonathan O'Connor
System Administrator
Inoveris LLC
Direct Line (614) 791-3742
Fax (614) 791-3748
Helpdesk 866-456-1566
 
 
 

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Doug Lytle
 Sent: Wednesday, October 19, 2005 12:04 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [Asterisk-Users] Caller-ID via database lookup
 
 Hey everybody,
 
 I'm having issues with one of our facilities, concerning caller-id.   
 The system is a Definity that hits a second Definity.  The 
 2nd Definity trunks the call to my Asterisk server via a 
 TE110P.  I can only get Caller-ID name.  Nothing in the From: 
 field.  I thought I would be able to do a database lookup 
 against name to match extension, but
 
 When doing this and setting Caller-ID number, it still shows 
 on the Polycom IP501 as Unknown/Unknown. Dial plan below:
 
 exten = s,1,Set(dnd=${DB(DND/${ARG1})}) exten = 
 s,2,Set(CIDNUMB=${DB(cidname/${CALLERIDNAME})})
 exten = s,3,Set(CALLERID(Name)=${CALLERIDNAME})
 exten = s,4,Set(CALLERID(Number)=${CIDNUMB})
 
 
 CLI output below:
 
 CLI -- Accepting AUTHENTICATED call from 192.168.101.10:
 requested format = gsm,
 requested prefs = (),
 actual format = gsm,
 host prefs = (gsm),
 priority = mine
 -- Executing Macro(IAX2/bc-asterisk-16384, 
 sip.extensions|4483|) in new stack
 -- Executing Set(IAX2/bc-asterisk-16384, dnd=) in new stack
 -- Executing Set(IAX2/bc-asterisk-16384, 
 CIDNUMB=5574) in new stack
 -- Executing Set(IAX2/bc-asterisk-16384, CALLERID(Name)=Lytle,
 Doug) in new stack
 -- Executing Set(IAX2/bc-asterisk-16384, 
 CALLERID(Number)=5574) in new stack
 -- Executing GotoIf(IAX2/bc-asterisk-16384, 0?8:6) in 
 new stack
 -- Goto (macro-sip.extensions,s,6)
 -- Executing SetMusicOnHold(IAX2/bc-asterisk-16384, 
 epi-cd) in new stack
 -- Executing Dial(IAX2/bc-asterisk-16384, 
 SIP/4483|28|t) in new stack
 -- Called 4483
 -- SIP/4483-04ba is ringing
 
 Debug output from the 'receiving Asterisk' server via IAX below:
 
 
 -- SIP/4483-14d3 is ringing
 Reliably Transmitting (no NAT) to 192.168.101.64:5060:
 CANCEL sip:[EMAIL PROTECTED] SIP/2.0
 Via: SIP/2.0/UDP 192.168.104.40:5060;branch=z9hG4bK73d230ce;rport
 From: Unknown sip:[EMAIL PROTECTED];tag=as23c39fbe
 To: sip:[EMAIL PROTECTED]
 Contact: sip:[EMAIL PROTECTED]
 Call-ID: [EMAIL PROTECTED]
 CSeq: 102 CANCEL
 User-Agent: Asterisk PBX
 Content-Length: 0
 
 What variable needs to be set to change it from Unknown to 5574?
 
 Any help would be appreciated.
 
 Doug
 
 -- 
  
 Ben Franklin quote:
 
 Those who would give up Essential Liberty to purchase a 
 little Temporary Safety, deserve neither Liberty nor Safety.
 
 
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Re: [Asterisk-Users] Re: Polycom IP501 and record on demand

2005-10-19 Thread Mojo with Horan Company, LLC

That is perfect for one-button remaps! I guess I migrated away from 
one-button features in * but I see the light now.


The trouble Matthew and I were having was to stimulate presses of more 
than one button in a sequence -- SpeedDial function was the only one I 
could find that was close, but this opens a new call appearance for the 
call rather than just playing the dtmf over the open one.


Moj

Noah Miller wrote:

Hi Matthew -



You could also take a look at features.conf, and use ** for blind
transfers, ## for attended transfers, *0 for recording, and *1 to  
hangup.


I haven't tried mapping them to polycom buttons, but there was
recently a discussion about that, just this week you can search the
archives.


There was a discussion (of which I was a part of) however there was no
resolution.  I have not found any good documentation on how to remap
Polycom buttons.  At this point I'm willing to pay for some help.
Anybody got some better info on this?



The best documentation I found is the Polycom manual.  It is fairly  
clear, though they don't provide a lot of examples.  Also, for some  
reason they put the button remapping documentation in one section of  
the manual and the button map in a completely different section.  A  
bit annoying.


I've remapped the transfer key to #, so I can do an asterisk  
unattended transfer using the transfer key.  To do this, I just added  
the following line to my ipmid.cfg (or sip.cfg if you are using  
firmware version 1.5.x or later):


keys key.scrolling.timeout=1  
key.IP_500.37.function.prim=DialpadPound  
key.IP_600.37.function.prim=DialpadPound/


Thanks,
Noah
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--
Mojo [EMAIL PROTECTED]
Office Manger, Horan  Company, LLC
(907) 747- x112
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Re: [Asterisk-Users] Caller-ID via database lookup

2005-10-19 Thread Doug Lytle



O'Connor, Jonathan wrote:


I have my Definity attached to my Asterisk box with a PRI Trunk.  The
guides and seemingly most people say to use a tie type connection,
however I did not get correct caller-id and setup until I:

1) Set the trunk-group on the Definity to isdn
2) Carrier/Medium to PRI
3) Trunk group numbering format to Public
 


Thanks for the Info Jonathan,

I'll give that info to our Definity manager.  I should have been clearer 
on our setup though.


The Definity that hooks up to our Asterisk box is a Definity G3R and I 
am getting CIDNumber and CIDName.  The other Definity hooks up to the 
first via a DCS link.  It's this linked Definity that I am having issues 
with on CIDNumber.


Again,

Thanks for your input!

Doug


--

Ben Franklin quote:

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


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Re: [Asterisk-Users] DID setup from goiax.com

2005-10-19 Thread pbx

Trixter:

Thanks for the guide to setting this up:... I have tried the below
configuration with my settings, and when I place /goiax-in after my
register command, my register statement fails.

If i remove it. I get a Rejected connect attempt from goiax's server IP,
trying to reach 's@'

I have put my GoIAX # in default, local, as the extension, and nothing.

I dont know where to look next on why i'm getting the rejected connect
attempt.

Thanks..

./Ben

 On Wed, 2005-10-19 at 14:24 +0800, Ronald Wiplinger wrote:
 Can anybody post a step by step setup guide, please?

 Its like anything else once you have signed up ...

 in iax.conf
 register = PHONENUMBER:[EMAIL PROTECTED]/goiax-in

 [goiax]
 type= peer
 host= server1.goiax.com
 context = default
 secret  = PASSWORD
 allow   = gsm
 ;allow  = ulaw
 ;disallow   = all
 notransfer  = yes
 qualify = yes
 auth= md5
 username= PHONENUMBER


 replace PHONENUMBER with the 8782 number you were issued.  Replace
 PASSWORD with your password from you account signup.

 Then in extensions.conf
 ; for outbound
 exten = _1NX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN},90,R)
 exten = _1NX,2,Busy
 exten = _1NX,102,Congestion
 exten = _1NX,202,playback(tt-weasels)

 ; for inbound
 exten = goiax-in,1,DO WHATEVER HERE

 asterisk -rx reload

 you should be set.


 --
 Trixter http://www.0xdecafbad.com Bret McDanel
 UK +44 870 340 4605   Germany +49 801 777 555 3402
 US +1 360 207 0479 or +1 516 687 5200
 FreeWorldDialup: 635378
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[Asterisk-Users] uable to establish link between asterisk to external phone

2005-10-19 Thread kotesh m


Hi,

I am new Asterisk. I configured asterisk1.5 and be able to communicate from iaxComm dial pad to external computer i.e outside my router/LAN. When I make call fromiamComm of external computer to my cell phone, I am getting the ring but not able to listenvoice on both sides. DoI need to make any special configuration to make voice link.


I found the same problemwhen used Sipura SIP device. 

Please let me know if I am missing anything.

Appreciate any help

--k
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Re: [Asterisk-Users] Agent recording and muxmon

2005-10-19 Thread Kevin P. Fleming


Julian Lyndon-Smith wrote:

Torrow your time I presume - it's today in the uk:). Will this be in 
1.2, or is it a post 1.2 ?


It will be in 1.2.

I don't understand why they would be incompatible changes - could you 
not add a MuxMon facility as another option. e.g. in agents.conf:


RecordAgentCalls=no
MuxMonAgentCalls=yes


The existing monitor application supports behavior that is not 
implemented by the new one, like applications changing the monitor 
filename while the call is being monitored, started/stopping under 
application control, etc. The new application can eventually support 
that, but it does not do so currently.

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[Asterisk-Users] NEWBIE HELP : chan_zap.c: Exception on 16, channel 1, call not being picked up on incoming X1-100P zap

2005-10-19 Thread Paul Hussein

I am running [EMAIL PROTECTED] ( asterisk 1.2beta1 with two X100P cards ) on 
centos 4.1 box with a 2.6.12 kernel.


I ran  genzaptelconf 

and added two trunks for each of the devices however the incoming calls 
when I ring just get ignored.


asterisk -r tells me that it just gets hangupcall, and in the the log 
files I see exceptions.


I am running asterisk 1.2 beta.   Can someone help as to how to debug this

I am new to the asterisk game so any hints would be greatfully received.


== Manager 'admin' logged on from 127.0.0.1
   -- Starting simple switch on 'Zap/1-1'
   -- Executing Macro(Zap/1-1, hangupcall) in new stack
   -- Executing ResetCDR(Zap/1-1, w) in new stack
   -- Executing NoCDR(Zap/1-1, ) in new stack
   -- Executing Wait(Zap/1-1, 5) in new stack
   -- Executing Hangup(Zap/1-1, ) in new stack
 == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 
'Zap/1-1' in macro 'hangupcall'

 == Spawn extension (from-internal, s, 1) exited non-zero on 'Zap/1-1'
   -- Executing Macro(Zap/1-1, hangupcall) in new stack
   -- Executing ResetCDR(Zap/1-1, w) in new stack
   -- Executing NoCDR(Zap/1-1, ) in new stack
   -- Executing Wait(Zap/1-1, 5) in new stack
   -- Executing Hangup(Zap/1-1, ) in new stack
 == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 
'Zap/1-1' in macro 'hangupcall'

 == Spawn extension (from-internal, h, 1) exited non-zero on 'Zap/1-1'





Oct 19 19:01:58 VERBOSE[10782] logger.c: -- Starting simple switch 
on 'Zap/1-1'
[EMAIL PROTECTED] ~]# Oct 19 19:02:03 NOTICE[10782] chan_zap.c: Got event 
18 (Event 18)...
Oct 19 19:02:03 VERBOSE[10782] logger.c: -- Executing 
Macro(Zap/1-1, hangupcall) in new stack
Oct 19 19:02:03 VERBOSE[10782] logger.c: -- Executing 
ResetCDR(Zap/1-1, w) in new stack
Oct 19 19:02:03 DEBUG[10782] cdr_addon_mysql.c: cdr_mysql: inserting a 
CDR record.
Oct 19 19:02:03 DEBUG[10782] cdr_addon_mysql.c: cdr_mysql: SQL command 
as follows:  INSERT INTO cdr 
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode) 
VALUES ('2005-10-19 19:02:03','\device\ 
400','400','s','from-internal', 'Zap/1-1','','ResetCDR','w',0,0,'NO 
ANSWER',3,'')
Oct 19 19:02:03 VERBOSE[10782] logger.c: -- Executing 
NoCDR(Zap/1-1, ) in new stack

Oct 19 19:02:03 WARNING[10782] cdr.c: CDR on channel 'Zap/1-1' not posted
Oct 19 19:02:03 WARNING[10782] cdr.c: CDR on channel 'Zap/1-1' lacks end
Oct 19 19:02:03 VERBOSE[10782] logger.c: -- Executing 
Wait(Zap/1-1, 5) in new stack

Oct 19 19:02:03 DEBUG[10782] chan_zap.c: Exception on 16, channel 1
Oct 19 19:02:03 DEBUG[10782] chan_zap.c: Got event Ring/Answered(2) on 
channel 1 (index 0)
Oct 19 19:02:03 DEBUG[10782] chan_zap.c: Setting IDLE polarity due to 
ring. Old polarity was 0

Oct 19 19:02:08 DEBUG[10782] chan_zap.c: Exception on 16, channel 1
Oct 19 19:02:08 DEBUG[10782] chan_zap.c: Got event Event 18(18) on 
channel 1 (index 0)
Oct 19 19:02:08 DEBUG[10782] chan_zap.c: Dunno what to do with event 18 
on channel 1
Oct 19 19:02:08 DEBUG[10782] acl.c: # Testing 192.168.0.108 with 
192.168.0.0
Oct 19 19:02:08 NOTICE[10782] chan_sip.c: Registration from 'new 
sip:[EMAIL PROTECTED]' failed for '192.168.0.108' - Wrong password
Oct 19 19:02:08 VERBOSE[10782] logger.c: -- Executing 
Hangup(Zap/1-1, ) in new stack
Oct 19 19:02:08 VERBOSE[10782] logger.c:   == Spawn extension 
(macro-hangupcall, s, 4) exited non-zero on 'Zap/1-1' in macro 'hangupcall'
Oct 19 19:02:08 VERBOSE[10782] logger.c:   == Spawn extension 
(from-internal, s, 1) exited non-zero on 'Zap/1-1'
Oct 19 19:02:08 VERBOSE[10782] logger.c: -- Executing 
Macro(Zap/1-1, hangupcall) in new stack
Oct 19 19:02:08 VERBOSE[10782] logger.c: -- Executing 
ResetCDR(Zap/1-1, w) in new stack
Oct 19 19:02:08 VERBOSE[10782] logger.c: -- Executing 
NoCDR(Zap/1-1, ) in new stack
Oct 19 19:02:08 VERBOSE[10782] logger.c: -- Executing 
Wait(Zap/1-1, 5) in new stack

Oct 19 19:02:08 DEBUG[10782] chan_zap.c: Exception on 16, channel 1
Oct 19 19:02:08 DEBUG[10782] chan_zap.c: Got event Ring/Answered(2) on 
channel 1 (index 0)
Oct 19 19:02:08 DEBUG[10782] chan_zap.c: Setting IDLE polarity due to 
ring. Old polarity was 0





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Re: [Asterisk-Users] Asterisk management portal

2005-10-19 Thread Jason Becker


Tomislav Parčina wrote:

Does anybody have detailed instruction how to Install AMP? I have tried to 
install it using Installation Guide on their pages but I'm unable to satisfy 
AMP's PERL module dependencies.


Please post to the amportal-users list:

http://lists.sourceforge.net/lists/listinfo/amportal-users

and/or Help forum:

http://sourceforge.net/forum/forum.php?forum_id=414452

Please include in your post what PERL dependencies you are unable to 
satisfy and why. Please provide standard output in your post.


Regards,

--
Jason Becker
Director  CEO
Coalescent Systems Inc.
Enabling Open Source Telephony
403.244.8089
www.coalescentsystems.ca
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Re: [Asterisk-Users] DID setup from goiax.com

2005-10-19 Thread Tom Vile

for the incoming context put your goiax.com phone number not the free DID number but the other one.On 10/19/05, 
[EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Trixter:Thanks for the guide to setting this up:... I have tried the belowconfiguration with my settings, and when I place /goiax-in after myregister command, my register statement fails.If i remove it. I get a Rejected connect attempt from goiax's server IP,
trying to reach 's@'I have put my GoIAX # in default, local, as the extension, and nothing.I dont know where to look next on why i'm getting the rejected connectattempt.Thanks.../Ben
 On Wed, 2005-10-19 at 14:24 +0800, Ronald Wiplinger wrote: Can anybody post a step by step setup guide, please? Its like anything else once you have signed up ... in 
iax.conf register = PHONENUMBER:[EMAIL PROTECTED]/goiax-in [goiax] type= peer host= 
server1.goiax.com context = default secret= PASSWORD allow = gsm ;allow= ulaw ;disallow = all notransfer= yes
 qualify = yes auth= md5 username= PHONENUMBER replace PHONENUMBER with the 8782 number you were issued.Replace PASSWORD with your password from you account signup.
 Then in extensions.conf ; for outbound exten = _1NX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN},90,R) exten = _1NX,2,Busy exten = _1NX,102,Congestion
 exten = _1NX,202,playback(tt-weasels) ; for inbound exten = goiax-in,1,DO WHATEVER HERE asterisk -rx reload you should be set.
 -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378
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-- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephonywww.baldwintechsolutions.com
Phone: 518-631-2855 x205Phone: 845-652-4578 x205Phone: 978-203-3848 x205Fax: 518-631-2856
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Re: [Asterisk-Users] DID setup from goiax.com

2005-10-19 Thread pbx

That is What I stated in the email.. my GOIAX #. not the DID #.

That is not the issue.

 for the incoming context put your goiax.com http://goiax.com phone
 number
 not the free DID number but the other one.

 On 10/19/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:

 Trixter:

 Thanks for the guide to setting this up:... I have tried the below
 configuration with my settings, and when I place /goiax-in after my
 register command, my register statement fails.

 If i remove it. I get a Rejected connect attempt from goiax's server IP,
 trying to reach 's@'

 I have put my GoIAX # in default, local, as the extension, and nothing.

 I dont know where to look next on why i'm getting the rejected connect
 attempt.

 Thanks..

 ./Ben

  On Wed, 2005-10-19 at 14:24 +0800, Ronald Wiplinger wrote:
  Can anybody post a step by step setup guide, please?
 
  Its like anything else once you have signed up ...
 
  in iax.conf
  register =
 PHONENUMBER:[EMAIL PROTECTED]/goiax-inhttp://PHONENUMBER:[EMAIL 
 PROTECTED]/goiax-in
 
  [goiax]
  type = peer
  host = server1.goiax.com http://server1.goiax.com
  context = default
  secret = PASSWORD
  allow = gsm
  ;allow = ulaw
  ;disallow = all
  notransfer = yes
  qualify = yes
  auth = md5
  username = PHONENUMBER
 
 
  replace PHONENUMBER with the 8782 number you were issued. Replace
  PASSWORD with your password from you account signup.
 
  Then in extensions.conf
  ; for outbound
  exten = _1NX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN},90,R)
  exten = _1NX,2,Busy
  exten = _1NX,102,Congestion
  exten = _1NX,202,playback(tt-weasels)
 
  ; for inbound
  exten = goiax-in,1,DO WHATEVER HERE
 
  asterisk -rx reload
 
  you should be set.
 
 
  --
  Trixter http://www.0xdecafbad.com Bret McDanel
  UK +44 870 340 4605 Germany +49 801 777 555 3402
  US +1 360 207 0479 or +1 516 687 5200
  FreeWorldDialup: 635378
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Re: [Asterisk-Users] DID setup from goiax.com

2005-10-19 Thread Sergey Okhapkin

Replace 
[goiax]
with
[PHONENUMBER]

username= don't work for users in IAX channel.

On Wed, 2005-10-19 at 10:27 -0700, [EMAIL PROTECTED] wrote:
 That is What I stated in the email.. my GOIAX #. not the DID #.
 
 That is not the issue.
 
  for the incoming context put your goiax.com http://goiax.com phone
  number
  not the free DID number but the other one.
 
  On 10/19/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
 
  Trixter:
 
  Thanks for the guide to setting this up:... I have tried the below
  configuration with my settings, and when I place /goiax-in after my
  register command, my register statement fails.
 
  If i remove it. I get a Rejected connect attempt from goiax's server IP,
  trying to reach 's@'
 
  I have put my GoIAX # in default, local, as the extension, and nothing.
 
  I dont know where to look next on why i'm getting the rejected connect
  attempt.
 
  Thanks..
 
  ./Ben
 
   On Wed, 2005-10-19 at 14:24 +0800, Ronald Wiplinger wrote:
   Can anybody post a step by step setup guide, please?
  
   Its like anything else once you have signed up ...
  
   in iax.conf
   register =
  PHONENUMBER:[EMAIL PROTECTED]/goiax-inhttp://PHONENUMBER:[EMAIL 
  PROTECTED]/goiax-in
  
   [goiax]
   type = peer
   host = server1.goiax.com http://server1.goiax.com
   context = default
   secret = PASSWORD
   allow = gsm
   ;allow = ulaw
   ;disallow = all
   notransfer = yes
   qualify = yes
   auth = md5
   username = PHONENUMBER
  
  
   replace PHONENUMBER with the 8782 number you were issued. Replace
   PASSWORD with your password from you account signup.
  
   Then in extensions.conf
   ; for outbound
   exten = _1NX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN},90,R)
   exten = _1NX,2,Busy
   exten = _1NX,102,Congestion
   exten = _1NX,202,playback(tt-weasels)
  
   ; for inbound
   exten = goiax-in,1,DO WHATEVER HERE
  
   asterisk -rx reload
  
   you should be set.
  
  
   --
   Trixter http://www.0xdecafbad.com Bret McDanel
   UK +44 870 340 4605 Germany +49 801 777 555 3402
   US +1 360 207 0479 or +1 516 687 5200
   FreeWorldDialup: 635378
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  --
  Tom Vile
  Baldwin Technology Solutions, Inc
  Consulting - Web Design - VoIP Telephony
  www.baldwintechsolutions.com http://www.baldwintechsolutions.com
  Phone: 518-631-2855 x205
  Phone: 845-652-4578 x205
  Phone: 978-203-3848 x205
  Fax: 518-631-2856
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[Asterisk-Users] Re: Polycom IP501 and record on demand

2005-10-19 Thread Noah Miller


Hi Mojo -


The trouble Matthew and I were having was to stimulate presses of more
than one button in a sequence -- SpeedDial function was the only  
one I
could find that was close, but this opens a new call appearance for  
the

call rather than just playing the dtmf over the open one.


Yeah, I'd like to be able to make that work, too.  Your method is  
pretty ingenious, and gets further that I got to mapping a key  
sequence to a single key.


I think we might just have to petition Polycom for this feature.   
Back in the pre-1.5.x firmware days, I did a feature request with  
them for the ability to disable their call waiting.  I don't know if  
my request really had an effect on them or not, but you can  
effectively do this in the 1.5.x firmware.  Maybe if everybody that  
want this submits a feature request to Polycom, they might just add  
it in a future firmware release.


There also might be some hope to hack together a solution on our  
own.  Anybody good with XML?



- Noah
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Re: [Asterisk-Users] DID setup from goiax.com

2005-10-19 Thread pbx

That did it... Thank you.

putting /goiax.com phone number after the register line caused it to not
register any more... and i would get error

server1.goiax.com/my goiax# could not be found.

anyways.. thanks for your help guys :)


 Replace
 [goiax]
 with
 [PHONENUMBER]

 username= don't work for users in IAX channel.

 On Wed, 2005-10-19 at 10:27 -0700, [EMAIL PROTECTED] wrote:
 That is What I stated in the email.. my GOIAX #. not the DID #.

 That is not the issue.

  for the incoming context put your goiax.com http://goiax.com phone
  number
  not the free DID number but the other one.
 
  On 10/19/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
 
  Trixter:
 
  Thanks for the guide to setting this up:... I have tried the below
  configuration with my settings, and when I place /goiax-in after my
  register command, my register statement fails.
 
  If i remove it. I get a Rejected connect attempt from goiax's server
 IP,
  trying to reach 's@'
 
  I have put my GoIAX # in default, local, as the extension, and
 nothing.
 
  I dont know where to look next on why i'm getting the rejected
 connect
  attempt.
 
  Thanks..
 
  ./Ben
 
   On Wed, 2005-10-19 at 14:24 +0800, Ronald Wiplinger wrote:
   Can anybody post a step by step setup guide, please?
  
   Its like anything else once you have signed up ...
  
   in iax.conf
   register =
  PHONENUMBER:[EMAIL PROTECTED]/goiax-inhttp://PHONENUMBER:[EMAIL 
  PROTECTED]/goiax-in
  
   [goiax]
   type = peer
   host = server1.goiax.com http://server1.goiax.com
   context = default
   secret = PASSWORD
   allow = gsm
   ;allow = ulaw
   ;disallow = all
   notransfer = yes
   qualify = yes
   auth = md5
   username = PHONENUMBER
  
  
   replace PHONENUMBER with the 8782 number you were issued.
 Replace
   PASSWORD with your password from you account signup.
  
   Then in extensions.conf
   ; for outbound
   exten = _1NX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN},90,R)
   exten = _1NX,2,Busy
   exten = _1NX,102,Congestion
   exten = _1NX,202,playback(tt-weasels)
  
   ; for inbound
   exten = goiax-in,1,DO WHATEVER HERE
  
   asterisk -rx reload
  
   you should be set.
  
  
   --
   Trixter http://www.0xdecafbad.com Bret McDanel
   UK +44 870 340 4605 Germany +49 801 777 555 3402
   US +1 360 207 0479 or +1 516 687 5200
   FreeWorldDialup: 635378
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  --
  Tom Vile
  Baldwin Technology Solutions, Inc
  Consulting - Web Design - VoIP Telephony
  www.baldwintechsolutions.com http://www.baldwintechsolutions.com
  Phone: 518-631-2855 x205
  Phone: 845-652-4578 x205
  Phone: 978-203-3848 x205
  Fax: 518-631-2856
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[Asterisk-Users] unable to make connectivity between asterisk to external phone

2005-10-19 Thread kotesh m


Hi All,

I am new toAsterisk. I configured asterisk and be able to communicate from iaxComm dial pad to external computer i.e outside my router/LAN. When I make call fromiamComm of external computer to my cell phone, I am getting the ring but not able to listenvoice . DoI need to make any special configuration to make voice link. 


I found the same problemwhen used SIPURA SIP device. Does anybody can provide the configuartionsettings for Sipura SIP..

Please let me know if I am missing anything.

Appreciate any help

--kotesh
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[Asterisk-Users] teliax audio issues - response