date:20051210


 I'm a beginner here and am interested in Teliax.  I own a small business and 
 was 
wondering if you guys could help
 me out here.  I'm basically looking for 6-8 telephone lines, but I notice 
 that Teliax 
supports 4 simultaneous calls on
 their Corporate plan.  So could I get two Corporate plans and set Asterisk to 
 use 
both of them and then have, in
 essence, 8 people talking at the same time?  If someone tries to call, would 
 the 
phone ring busy or would it still go
 through?
 
 I plan on having a T1.

I'd suggest you call their sales folks as teliax is rather flexible; they
will likely work something out for you that fits your needs.

As others have mentioned, the bundled plans (eg, residential or corporate)
have a soft cap that essentially translates into $0.018 / minute, assuming
you use every single minute within the plan. If you don't use every minute,
the average cost/min goes up (1,000 minutes of corp plan use = $0.045 / min).

So, you are probably better off with their Pay as you go plan which
ensures your cost is always $0.02 / min with an unlimited number of 
simultanous calls.

If you combine the above with some thought as to what you are going to do
when calls can't be completed via teliax (for whatever reason), then you
are likely to conclude that having two providers at some flat cost per
minute is a positive move.

If you add to that thought process some probability that you can't complete
_any_ Internet-based calls (due to T1 failure or whatever), then you're
likely to approach a combination of itsp's and pstn lines for your business.


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[Asterisk-Users] Email to voice?

2005-12-10 Thread Chris Mason (Lists)

I want to have our monitoring system call me if  a high level alert 
event happens. I could use SMS to get my attention but I don't have an 
SMS gateway available to me. Is there a way I can have the system run a 
script (via ssh) on the pbx that will call my number and read me the 
message? Has anyone done anything like this? I also thought of setting 
the CallerID Name to Server5 down for example.

Any ideas on this appreciated.

--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED] 


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[Asterisk-Users] Possible bug in record?

2005-12-10 Thread Steve Hanselman









Im trying to get record to append to a file, Im
using this:-



exten = 2,n,record(/tmp/${UNIQUEID}.gsm|5|0|a)



And its creating a new file?



If I check /tmp I can see the same filename being reused
each time, but the file jjust contains the latest recording.



Can anybody else confirm this?



Steve










The information contained in this email is intended for the personal and confidential useof the addressee only. It may also be privileged information. If you are not the intendedrecipient then you are hereby notified that you have received this document in error andthat any review, distribution or copying of this document is strictly prohibited. If you have received  this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) LtdNevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX  UKRegistered Office as above. Registered in England No. 2764339See our current vacancies at www.brendataco.uk___
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[Asterisk-Users] Asterisk not Replying on Port Specified in the VIA header

2005-12-10 Thread Saad Siddiqi

Hello,
I am trying to send OPTIONS message asterisk in order to find out that
whether it is alive or not. everything is going fine except for the
port it is sending the reply to. The problem is that it is not replying
to the port specied in the VIA header, and is replying on the
port from which it has recieved the request. 
How can I be able to send the reply on the Port specified in the VIA header???

thanks-- Saad
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Re: [Asterisk-Users] Possible bug in record?

2005-12-10 Thread Doug Lytle


Steve Hanselman wrote:


I’m trying to get record to append to a file, I’m using this:-

exten = 2,n,record(/tmp/${UNIQUEID}.gsm|5|0|a)

And it’s creating a new file?


I don't think record has an option to append.  Use a script and cat the two 
files together.

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] Channel 0/1, span 1 got hangup request

Just a couple guesses on things to try.

Zapata.conf
1.  Changing switchtype variables (doubtful but give it a try).  
2.  Add a variable to define pridialplan (I remember someone setting
this to unknown to solve a similar issue)  Try pridialplan=unknown
and/or prilocaldialplan=local or some other valid option.

Zaptel.conf
1.  span=1,1,5,ccs,hdb3

I think that your dial statement or the pridialplan is your issue.  If
you look at the debug info 

Here is something suspicious:  -- Called g1/100 unless 100 is the
number you are trying to dial outbound.

If the above fails, then try below.

Try tweaking your settings here like span=1,0,0,ccs,hdb3

What is the provider expecting?

Thanks,
Steve


 Dear Users,
 
 I have an Digium Wildcard TE110P T1/E1 Card inserted in Linux box
runnig
 Asterisk 1.2.0
 All incoming calls from E1 interface to SIP-phone goes exellent, but
 calls from SIP to E1 gives the errors:
 
  -- Executing Dial(SIP/anton-6cf4, Zap/g1/100) in new stack
 -- Making new call for cr 32775
  -- Requested transfer capability: 0x00 - SPEECH
  Protocol Discriminator: Q.931 (8)  len=43
  Call Ref: len= 2 (reference 7/0x7) (Originator)
  Message type: SETUP (5)
  [04 03 80 90 a3]
  Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer
 capability: Speech (0)
   Ext: 1  Trans mode/rate: 64kbps,
 circuit-mode (16)
   Ext: 1  User information layer 1: A-Law
 (35)
  [18 03 a9 83 81]
  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: 1 ]
  [28 05 41 6e 74 6f 6e]
  Display (len= 5) ╫)[EMAIL PROTECTED]@[EMAIL PROTECTED]@│@[ Anton ]
  [6c 0d 21 81 38 34 37 37 33 36 31 38 31 38 33]
  Calling Number (len=15) [ Ext: 0  TON: National Number (2)  NPI:
 ISDN/Telephony Numbering Plan (E.164/E.163) (1)
Presentation: Presentation permitted, user
 number passed network screening (1) '84773618183' ]
  [70 04 a1 31 30 30]
  Called Number (len= 6) [ Ext: 1  TON: National Number (2)  NPI:
 ISDN/Telephony Numbering Plan (E.164/E.163) (1) '100' ]
  -- Called g1/100
  Protocol Discriminator: Q.931 (8)  len=9
  Call Ref: len= 2 (reference 7/0x7) (Terminator)
  Message type: DISCONNECT (69)
  [08 02 80 90]
  Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0: 0
 Location: User (0)
   Ext: 1  Cause: Unknown (16), class = Normal Event
(1) ]
 -- Processing IE 8 (cs0, Cause)
  -- Channel 0/1, span 1 got hangup request
 Dec  5 15:30:12 WARNING[30946]: app_dial.c:706 wait_for_answer: Unable
 to forward voice
 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication,
 peerstate Disconnect Request
  Protocol Discriminator: Q.931 (8)  len=9
  Call Ref: len= 2 (reference 7/0x7) (Originator)
  Message type: RELEASE (77)
  [08 02 81 90]
  Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0: 0
 Location: Private network serving the local user (1)
   Ext: 1  Cause: Unknown (16), class = Normal Event
(1) ]
  -- Hungup 'Zap/1-1'
== No one is available to answer at this time (1:0/0/0)
  Protocol Discriminator: Q.931 (8)  len=9
  Call Ref: len= 2 (reference 7/0x7) (Terminator)
  Message type: RELEASE COMPLETE (90)
  [08 02 80 d1]
  Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0: 0
 Location: User (0)
   Ext: 1  Cause: Unknown (81), class = Invalid
message
 (5) ]
 -- Processing IE 8 (cs0, Cause)
 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null
 NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null
  -- Timeout on SIP/anton-6cf4
== CDR updated on SIP/anton-6cf4
  -- Executing Hangup(SIP/anton-6cf4, ) in new stack
 
 
 /etc/zaptel.conf
 span=1,1,5,ccs,hdb3
 bchan=1-15,17-31
 dchan=16
 loadzone = nl
 defaultzone=nl
 
 /etc/asterisck/zapata.conf
 [trunkgroups]
 [channels]
 language=en
 signalling=pri_cpe
 switchtype=euroisdn
 echocancel=32
 echocancelwhenbridged=yes
 usecallerid=yes
 callerid=asreceived
 transfer=yes
 overlapdial=yes
 cancallforward=yes
 group=1
 context=zapata
 channel = 1-15,17-31
 
 Has anybody resolve this problem?
 
 --
 SY,
 Anton V Bakulev.
 MIPT-telecom.
 [EMAIL PROTECTED]

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Re: [Asterisk-Users] Via Epia

2005-12-10 Thread Roger Hill

After reading the first post, I went back into the makefile, and 
PROC=i586. (only in the one place, top level makefile)


Mine now works! No more 'illegal instruction'.

Roger

Roger Hill wrote:


Aha!

I was getting the same error and could not figure out why.

My CPU is a VIA Samuel.

So it's a VIA thing??

Roger

Andrew Nowrot wrote:


Hi,

Does anyone has some experience in installing * on Via Epia. I am
struggling with it for about two days. And when I finally managed to
install asterisk 1.0.9 after starting it I get this error or whatever:

-  Illegal instruction

I changed the variable in makefile to i586 (I also tried i686 because
that is what my uname - m says) but still I get the same problem.

I use Debian with 2.4.30 kernel.

Does anyone has some experience with Via Epia and Asterisk. Will this
mix work in appropriate way ;) ?

Cheers

Andrew
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--

Roger Hill  07739 707 180
Perseverance is the hard work you do after you get
tired of doing the hard work you already did.


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Re: [Asterisk-Users] Daily Reboot Script for Asterisk Question

2005-12-10 Thread Doug Lytle


Min Hwan Chang wrote:

Yes I understand that daily reboot is unnecessary but until I find the
problem, this works for our needs.  I'm wondering what that line does
because last night when the Cron job started running, it kept running
the job over and over until I got an out of memory error... as seen
below:

/var/log/messages
Nov  9 04:15:32 localhost kernel: Registered tone zone 0 (United States / North$
Nov  9 04:18:00 localhost kernel: Freed a Wildcard
Nov  9 04:18:02 localhost kernel: Freshmaker version: 71
Nov  9 04:18:02 localhost kernel: Freshmaker passed register test
Nov  9 04:18:02 localhost kernel: Module 0: Installed -- AUTO FXO (FCC mode)
Nov  9 04:18:02 localhost kernel: Module 1: Installed -- AUTO FXO (FCC mode)
Nov  9 04:18:02 localhost kernel: Module 2: Installed -- AUTO FXO (FCC mode)
  

Your logs show that the TDM was registered at 4:18am


Nov  9 07:07:12 localhost kernel: Out of Memory: Killed process 11022 (sendmail$
Nov  9 07:08:57 localhost kernel: Out of Memory: Killed process 2722 (sendmail).
Nov  9 07:09:04 localhost kernel: Out of Memory: Killed process 21792 (sendmail$
Nov  9 07:09:10 localhost kernel: Out of Memory: Killed process 24036 (sendmail$
Nov  9 07:11:00 localhost kernel: Out of Memory: Killed proc
  

Your out of memory errors occured at 7am and it appears to be sendmail causing 
the issue.

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] Email to voice?

 I want to have our monitoring system call me if  a high level alert 
 event happens. I could use SMS to get my attention but I don't have an 
 SMS gateway available to me. Is there a way I can have the system run a 
 script (via ssh) on the pbx that will call my number and read me the 
 message? Has anyone done anything like this? I also thought of setting 
 the CallerID Name to Server5 down for example.
 Any ideas on this appreciated.

A lot of that depends on your definition of an event, the detection
of that event, and what you might have for available resources to deal
with the event.

If you're definition of event is simply analyzing syslog messages and
notifying you based on keywords, there are several products that can be
combined to send voice messages to your cell phone, etc.

In the intrusion detection world, snortsnarf is a perl script that can be
used to analyze syslog-type text messages and do something with those that
match your defined criteria. That might include sending a text based
message to another asterisk system (that is assured of being up), and
then have it call you with a festival voice message (or something like 
that). Google for SnortSnarf.


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RE: [Asterisk-Users] Synthesized Voice for Asterisk

Cepstral sounds great.  You can test it for free but I will append some
message about being free until you pay for the license.  

I will be purchasing a license shortly but the way I read it (could be
wrong), the licensing is similar to g729 in that a license is only good
for a simultaneous use.

Thanks,
Steve


 -Original Message-
 From: John Cianfarani [mailto:[EMAIL PROTECTED]
 Sent: Friday, December 09, 2005 10:18 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] Synthesized Voice for Asterisk
 
 Cepstral has some pretty decent quality voices at like $29 they don't
 break the bank.
 
 https://www.cepstral.com
 
 It also can integrate directly into asterisk I believe.
 
 Hope that helps
 John
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Dakota
 Sent: Friday, December 09, 2005 7:49 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [Asterisk-Users] Synthesized Voice for Asterisk
 
 Are there any cool free software I can use to create automated voice
 message
 greetings for my PBX?
 
 I want to customized some of my messages, however prefer to use a
 standard
 voice.
 
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Re: [Asterisk-Users] Via Epia

2005-12-10 Thread Sergey Okhapkin





I run * on VIA EPIA M1 on gentoo. Here is my ~/.asterisk.makeopts:

K6OPT = -DK6OPT
DEBUG=
ASTVARRUNDIR=$(INSTALL_PREFIX)/var/run/asterisk
PROC=i686


On Sat, 2005-12-10 at 12:36 +0100, Maciej Kietlinski wrote:


Does anyone has some experience in installing * on Via Epia. I am
struggling with it for about two days. And when I finally managed to
install asterisk 1.0.9 after starting it I get this error or whatever:

-  Illegal instruction

I changed the variable in makefile to i586 (I also tried i686 because
that is what my uname - m says) but still I get the same problem.

I use Debian with 2.4.30 kernel.

Does anyone has some experience with Via Epia and Asterisk. Will this
mix work in appropriate way ;) ?


There is not only 1 makefile where you have to define i586.



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Re: [Asterisk-Users] Via Epia

2005-12-10 Thread Sergey Okhapkin





See http://www.epiawiki.org/wiki/tiki-index.php?page=EpiaInstallingGentoo regarding CFLAGS settings for different VIA CPUs.

On Sat, 2005-12-10 at 11:57 +, Roger Hill wrote:


Aha!

I was getting the same error and could not figure out why.

My CPU is a VIA Samuel.

So it's a VIA thing??

Roger

Andrew Nowrot wrote:

Hi,

Does anyone has some experience in installing * on Via Epia. I am
struggling with it for about two days. And when I finally managed to
install asterisk 1.0.9 after starting it I get this error or whatever:

-  Illegal instruction

I changed the variable in makefile to i586 (I also tried i686 because
that is what my uname - m says) but still I get the same problem.

I use Debian with 2.4.30 kernel.

Does anyone has some experience with Via Epia and Asterisk. Will this
mix work in appropriate way ;) ?

Cheers

Andrew
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Re: [Asterisk-Users] Email to voice?

2005-12-10 Thread Jean-Michel Hiver


Chris Mason (Lists) a écrit :

I want to have our monitoring system call me if  a high level alert 
event happens. I could use SMS to get my attention but I don't have an 
SMS gateway available to me. Is there a way I can have the system run 
a script (via ssh) on the pbx that will call my number and read me the 
message? Has anyone done anything like this? I also thought of setting 
the CallerID Name to Server5 down for example.

Any ideas on this appreciated.


Well, your monitoring program could write a .call file (which will make 
asterisk initiate a call) which would call your number whenever there is 
a problem. Then use a bit of scripting + festival to read out the 
message to you.


Cheers,
Jean-Michel.

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Re: [Asterisk-Users] Email to voice?

2005-12-10 Thread Peter Bowyer

On 10/12/05, Chris Mason (Lists) [EMAIL PROTECTED] wrote:
 I want to have our monitoring system call me if  a high level alert
 event happens. I could use SMS to get my attention but I don't have an
 SMS gateway available to me. Is there a way I can have the system run a
 script (via ssh) on the pbx that will call my number and read me the
 message? Has anyone done anything like this? I also thought of setting
 the CallerID Name to Server5 down for example.
 Any ideas on this appreciated.

Have you considered using an SMS gateway provider? We use both Bayham
Systems (www.bayhamsystems.com) and Connection Software
(www.csoft.co.uk - quote offer code PB45 for an introductory discount,
yes it's my affiliate code, feel free to ignore). Both providers can
easily be integrated over HTTP or SMTP using a shell script, perl, etc
and/or via AGI (we use Bayham for MWI notifications to cellphones).

Peter


--
Peter Bowyer
Email: [EMAIL PROTECTED]
Tel: +44 1296 768003
VoIP: sip:[EMAIL PROTECTED]
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Re: [Asterisk-Users] Asterisk, Small Business, and Teliax

2005-12-10 Thread Andrew Berman

Thank you very much for your responses. I like the idea of having Teliax as well as some PSTN lines in the event of the T1 going down. I've just started to read the Asterisk book by O'Reilly, so my understanding of Asterisk is limited right now. Consequently, if I get a TDM400P for the PSTN lines and get Teliax, can Asterisk be set up in such a way that if Teliax cannot be reached it uses the PSTN lines? If yes, I'm assuming it has to do with the proper diaplan, which I'll be reading up on soon.
Thanks again for your help,AndrewOn 12/10/05, Rich Adamson [EMAIL PROTECTED] wrote:
 I'm a beginner here and am interested in Teliax.I own a small business and was
wondering if you guys could help me out here.I'm basically looking for 6-8 telephone lines, but I notice that Teliaxsupports 4 simultaneous calls on their Corporate plan.So could I get two Corporate plans and set Asterisk to use
both of them and then have, in essence, 8 people talking at the same time?If someone tries to call, would thephone ring busy or would it still go through? I plan on having a T1.
I'd suggest you call their sales folks as teliax is rather flexible; theywill likely work something out for you that fits your needs.As others have mentioned, the bundled plans (eg, residential or corporate)
have a soft cap that essentially translates into $0.018 / minute, assumingyou use every single minute within the plan. If you don't use every minute,the average cost/min goes up (1,000 minutes of corp plan use = $0.045 / min).
So, you are probably better off with their Pay as you go plan whichensures your cost is always $0.02 / min with an unlimited number ofsimultanous calls.If you combine the above with some thought as to what you are going to do
when calls can't be completed via teliax (for whatever reason), then youare likely to conclude that having two providers at some flat cost perminute is a positive move.If you add to that thought process some probability that you can't complete
_any_ Internet-based calls (due to T1 failure or whatever), then you'relikely to approach a combination of itsp's and pstn lines for your business.___--Bandwidth and Colocation provided by 
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Re: [Asterisk-Users] Call Routing from GnuGK to Asterisk

2005-12-10 Thread Rafael Marconi


Hi,
im using debian with asterisk instaled via apt-get (1.0.7 + oh323)

my asterisk is not a Gk, but a GW.
Look this conf


/etc/asterisk/oh323.conf


[register]
alias=asterisk
gwprefix=9
gwprefix=8
gwprefix=7
gwprefix=6
gwprefix=5
gwprefix=4
gwprefix=3
gwprefix=2
gwprefix=1
gwprefix=0



/etc/gatekeeper.ini

[RasSrv::GWPrefixes]
127.0.0.1
asterisk=0,1,2,3,4,5,6,7,8,9




in this way i force all calls to asterisk
i use some extensions with tech-prefix to route calls

like this

exten = _130X.,1,SetCallerID(130)
exten = _130X.,2,Dial,SIP/${EXTEN:[EMAIL PROTECTED],tr


Rafael Marconi


Em 09/12/2005, às 18:03, Code Lover escreveu:


Hi all,

I installed asterisk-oh323 and GnuGK on the linux box, and i wanted to
route all gnugk registered endpoint's call to oh323 GW (Asterisk)
which is already registered with GnuGK as Gatekeepr.

here is my gnugk.ini configuation.

 [Gatekeeper::Main]
 Fourtytwo=42
 TimeToLive=750
 Name=gnugk

 [RoutedMode]
 GKRouted=1
 H245Routed=0
 CallSignalPort=1720
 CallSignalHandlerNumber=1
 RemoveH245AddressOnTunneling=0
 AcceptNeighborsCalls=1
 AcceptUnregisteredCalls=1
 SupportNATedEndpoints=1
 DropCallsByReleaseComplete=1

 [RasSrv::ARQFeatures]
 ArjReasonRouteCallToSCN=0
 ArjReasonRouteCallToGatekeeper=1
 RoundRobinGateways=1

 [RoutingPolicy]
 default=neighbor

 [RasSrv::Neighbors]
 GK1=asterisk

 [Neighbor::GK1]
 GatekeeperIdentifier=GK1
 Host=212.xxx.xxx.xxx
 SendPrefixes=0
 AcceptPrefixes=*
 ForwardLRQ=always

Here is my error what i am getting in gnugk log.

admissionReject {
requestSeqNum = 8191
rejectReason = calledPartyNotRegistered null
  }

Please help me to solve this problem
--
--
Thank You,
Code Lover
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Re: [Asterisk-Users] Via Epia

2005-12-10 Thread Andrew Nowrot

Hi,

Could you specify the amount of makefiles because I use * from
Bristuff and  only changed the makefile in asterisk directory. What
others makefiles should I change?

Andrew
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Re: [Asterisk-Users] Asterisk Bounty Pool

2005-12-10 Thread Chris Tooley


 For information on the program go to
 http://www.unwiredbuyer.com/asterisk
 
I've updated this page to show the amount of money earned, as soon as
people have signed up and placed a bid, the $10 will be added to the
total here.  At the last time I looked several people had signed up but
no one had placed a bid.

-- 
Chris Tooley
512-646-1507
[EMAIL PROTECTED]

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RE: [Asterisk-Users] Door Phones

2005-12-10 Thread Anton Krall

I was asking for if somebody in Mexico were already using some because I
wanted to know if anybody is already importing these into MX. Of course all
the US doorphones work here but, with importing costs, some become way to
expensive for the end user.

Cheers! 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of C F
|Sent: Wednesday, December 07, 2005 1:51 PM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] Door Phones
|
|I'm not in Mexico, but I'm sure what I use here works in 
|Mexico as well (BTW, it's on the wiki). I have successfuly used:
|Valcom
|VikingElectronics
|doorfonebell
|I might have spelling wrong on the last one. Too lazy to look 
|it up on the Wiki :)
|
|
|On 12/7/05, Anton Krall [EMAIL PROTECTED] wrote:
| Guys, Im wondering, is anybody in Mexico using any kind of 
|door phone 
| with asterisk?
|
| Please drop me a note.
|
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[Asterisk-Users] Re: Asterisk-Users Digest, Vol 17, Issue 61

2005-12-10 Thread Abhishek Gangal

Sir
 I am a novice user and want to set up the asterix for only Voip as a project in my final yr. computer engineeering.
 Pl. help me to do so . I will be highly thankful


Abhishek Gangal
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Re: [Asterisk-Users] Asterisk, Small Business, and Teliax

Your assumptions are right on the mark. However, keep in mind that
regardless of how much effort you put into trying to figure out whether
teliax is up/down, there are always things that can happen that you can't
discover. For example, teliax (or any other itsp) might accept your outgoing
call and its not processed for whatever reason. Such occurances can only be
addressed if you provide your users with an alternative way to dial. Common
approaches would be to include something like: a) dial 9+digits for an pstn
call, b) dial 8+digits for teliax calls, and, c) all 1+digits calls are
automatically routed based on whatever you set up in the dialplan.

Using such an approach essentially has your users dialing whatever number 
they need to (c) under normal conditions, but should there be a problem, the 
user can still call outbound by directing their calls to (a) or (b).

Your thought process also addresses 911 calls, etc, by you programming your
dialplan to route those calls via the pstn lines. No need to even think 
about routing 911 calls via teliax.

Keep in mind that whatever you do with fax'ing probably will not work through
voip and the TDM card. Lots of postings in the list archives if you need to
research that. (Since you are likely to have pstn lines, consider attaching
a fax machine to one of those lines and not let asterisk answer incoming calls
on that line. Or, subscribe to an external fax service and have them email 
pdf files instead of messing around with paper, toner, questionable fax 
machines, modems, etc.)


 Thank you very much for your responses.  I like the idea of having Teliax as 
 well as 
some PSTN lines in the event
 of the T1 going down.  I've just started to read the Asterisk book by 
 O'Reilly, so my 
understanding of Asterisk is
 limited right now.  Consequently, if I get a TDM400P for the PSTN lines and 
 get 
Teliax, can Asterisk be set up in
 such a way that if Teliax cannot be reached it uses the PSTN lines?  If yes, 
 I'm 
assuming it has to do with the
 proper diaplan, which I'll be reading up on soon.
 
 Thanks again for your help,
 
 Andrew
 
 On 12/10/05, Rich Adamson [EMAIL PROTECTED] wrote:
 
  I'm a beginner here and am interested in Teliax.  I own a small 
 business and 
was
 wondering if you guys could help
  me out here.  I'm basically looking for 6-8 telephone lines, but I 
 notice that 
Teliax
 supports 4 simultaneous calls on
  their Corporate plan.  So could I get two Corporate plans and set 
 Asterisk to 
use
 both of them and then have, in
  essence, 8 people talking at the same time?  If someone tries to call, 
 would 
the
 phone ring busy or would it still go
  through?
 
  I plan on having a T1.
 
 I'd suggest you call their sales folks as teliax is rather flexible; they
 will likely work something out for you that fits your needs.
 
 As others have mentioned, the bundled plans (eg, residential or corporate)
 have a soft cap that essentially translates into $0.018 / minute, assuming
 you use every single minute within the plan. If you don't use every 
 minute,
 the average cost/min goes up (1,000 minutes of corp plan use = $0.045 / 
 min).
 
 So, you are probably better off with their Pay as you go plan which
 ensures your cost is always $0.02 / min with an unlimited number of
 simultanous calls.
 
 If you combine the above with some thought as to what you are going to do
 when calls can't be completed via teliax (for whatever reason), then you
 are likely to conclude that having two providers at some flat cost per
 minute is a positive move.
 
 If you add to that thought process some probability that you can't 
 complete
 _any_ Internet-based calls (due to T1 failure or whatever), then you're
 likely to approach a combination of itsp's and pstn lines for your 
 business.
 
 ___
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 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
---End of Original Message-


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RE: [Asterisk-Users] Re: Asterisk-Users Digest, Vol 17, Issue 61

 
 Sir
I am a novice user and want to set up the asterix for only Voip as
a
 project in my final yr. computer engineeering.
  Pl.  help me to do so .   I will be highly thankful
 
 
 Abhishek Gangal

RTFM especially for a high level project in computer engineering
schooling.

I once met a guy with a degree in Computer Science from a pretty
respectable college and he could not even log into a Windows machine.

Thanks,
Steve
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Re: [Asterisk-Users] Re: Asterisk-Users Digest, Vol 17, Issue 61

2005-12-10 Thread Steve Underwood


Steve Totaro wrote:


Sir
  I am a novice user and want to set up the asterix for only Voip as
   


a
 


project in my final yr. computer engineeering.
Pl.  help me to do so .   I will be highly thankful


Abhishek Gangal
   



RTFM especially for a high level project in computer engineering
schooling.

I once met a guy with a degree in Computer Science from a pretty
respectable college and he could not even log into a Windows machine.
 

I think I know how he felt. I really have to struggle to make myself log 
on to Windows machines. :-)


Steve


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[Asterisk-Users] agi variables list

2005-12-10 Thread Olivier Taylor

Title: Message



hello 
all,

where 
can I find a list of agi variables that can be read by a external 
script?

Thanks,

Olivier
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[Asterisk-Users] Problems with zaptel channels not properly being answered...

2005-12-10 Thread Jim Hodapp

I have a TDM400P card with 2 incoming analog lines on it. I have a  
working dialplan for 2 IAX lines that work perfectly. I want these 2  
analog lines to also utilize my dialplan for both incoming and  
outgoing. However, as my config stands right now, any incoming calls  
from the analog lines aren't working perfectly yet.


First, the analog calls are seen by Asterisk and Asterisk looks like  
it's picking the call up and the dialplan executes. I can see this  
all happening if I run sudo asterisk - from a command line and  
watch the console print out messages. However, from the analog  
caller's side, it sounds like Asterisk never picked up as you always  
hear ringing and that's all. I will place my /etc/zaptel.conf and / 
etc/asterisk/zapata.conf files inline with this message. Any  
suggestions are greatly appreciated.


Thanks,

Jim Hodapp

 /etc/zaptel.conf -
loadzone=us
defaultzone=us
fxoks=1-2
fxsks=3-4
---


 /etc/asterisk/zapata.conf -
[trunkgroups]

[channels]

language=en
context=default
busydeteect=yes
busycount=6
echotraining=800
echocancel=yes
immediate=no
usecallerid=yes
callwaiting=yes
hidecallerid=no
threewaycalling=yes
transfer=yes

;;;[400]
signalling=fxo_ks
[EMAIL PROTECTED]
echotraining=800
echocancelwhenbridge=no
echocancel=yes
context=longdistance
callprogress=no
callerid=Bowen  Fenwick400
busydetect=no
busycount=7
channel = 1

[401]
signalling=fxo_ks
[EMAIL PROTECTED]
echotraining=800
echocancelwhenbridge=no
echocancel=yes
context=longdistance
callprogress=no
callerid=Conference Room #1401
busydetect=no
busycount=7
channel = 2

group=1
signalling=fxo_ks   ; Drive a handset or other station device.
context=default
channel = 1-2

group=2
signalling=fxs_ks   ; Signals the phone company.
context=longdistance
channel = 3-4
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RE: [Asterisk-Users] agi variables list

Is this what you mean?

http://www.voip-info.org/tiki-index.php?page=Asterisk+AGI

 -Original Message-
 From: Olivier Taylor [mailto:[EMAIL PROTECTED]
 Sent: Saturday, December 10, 2005 12:44 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: [Asterisk-Users] agi variables list
 
 hello all,
 
 where can  I find a list of agi variables that can be read by a
external
 script?
 
 Thanks,
 
 Olivier
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RE: [Asterisk-Users] Problems with zaptel channels not properly beinganswered...

 
 I have a TDM400P card with 2 incoming analog lines on it. I have a
 working dialplan for 2 IAX lines that work perfectly. I want these 2
 analog lines to also utilize my dialplan for both incoming and
 outgoing. However, as my config stands right now, any incoming calls
 from the analog lines aren't working perfectly yet.
 
 First, the analog calls are seen by Asterisk and Asterisk looks like
 it's picking the call up and the dialplan executes. I can see this
 all happening if I run sudo asterisk - from a command line and
 watch the console print out messages. However, from the analog
 caller's side, it sounds like Asterisk never picked up as you always
 hear ringing and that's all. I will place my /etc/zaptel.conf and /
 etc/asterisk/zapata.conf files inline with this message. Any
 suggestions are greatly appreciated.
 
 Thanks,
 
 Jim Hodapp
 
  /etc/zaptel.conf -
 loadzone=us
 defaultzone=us
 fxoks=1-2
 fxsks=3-4
 ---
 
 
  /etc/asterisk/zapata.conf -
 [trunkgroups]
 
 [channels]
 
 language=en
 context=default
 busydeteect=yes
 busycount=6
 echotraining=800
 echocancel=yes
 immediate=no
 usecallerid=yes
 callwaiting=yes
 hidecallerid=no
 threewaycalling=yes
 transfer=yes
 
 ;;;[400]
 signalling=fxo_ks
 [EMAIL PROTECTED]
 echotraining=800
 echocancelwhenbridge=no
 echocancel=yes
 context=longdistance
 callprogress=no
 callerid=Bowen  Fenwick400
 busydetect=no
 busycount=7
 channel = 1
 
 [401]
 signalling=fxo_ks
 [EMAIL PROTECTED]
 echotraining=800
 echocancelwhenbridge=no
 echocancel=yes
 context=longdistance
 callprogress=no
 callerid=Conference Room #1401
 busydetect=no
 busycount=7
 channel = 2
 
 group=1
 signalling=fxo_ks   ; Drive a handset or other station device.
 context=default
 channel = 1-2
 
 group=2
 signalling=fxs_ks   ; Signals the phone company.
 context=longdistance
 channel = 3-4


Please post the output from your console.  Set debug on and verbose 60
or something.

Thanks,
Steve
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RE: [Asterisk-Users] Re: Asterisk-Users Digest, Vol 17, Issue 61

2005-12-10 Thread Ossi Sariola

Windows? login?. MACHINE???!???!..

-Original Message-
From: Steve Underwood [mailto:[EMAIL PROTECTED] 
Sent: Saturday, December 10, 2005 3:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: Asterisk-Users Digest, Vol 17, Issue 61

Steve Totaro wrote:

Sir
   I am a novice user and want to set up the asterix for only Voip as

a

project in my final yr. computer engineeering.
 Pl.  help me to do so .   I will be highly thankful

Abhishek Gangal

RTFM especially for a high level project in computer engineering
schooling.

I once met a guy with a degree in Computer Science from a pretty
respectable college and he could not even log into a Windows machine.

I think I know how he felt. I really have to struggle to make myself log 
on to Windows machines. :-)

Steve

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RE : [Asterisk-Users] agi variables list

2005-12-10 Thread Olivier Taylor

Not really, I am looking for a list of available headers in agi.

I know the way to read them, but for example, I need to read the contact
header, but I don't know the variable name in agi.

Olivier



-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Steve Totaro
Envoyé : samedi 10 décembre 2005 18:52
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : RE: [Asterisk-Users] agi variables list


Is this what you mean?

http://www.voip-info.org/tiki-index.php?page=Asterisk+AGI

 -Original Message-
 From: Olivier Taylor [mailto:[EMAIL PROTECTED]
 Sent: Saturday, December 10, 2005 12:44 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: [Asterisk-Users] agi variables list
 
 hello all,
 
 where can  I find a list of agi variables that can be read by a
external
 script?
 
 Thanks,
 
 Olivier
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Re: [Asterisk-Users] Asterisk, Small Business, and Teliax

2005-12-10 Thread Andrew Berman

Thank you very much for your help Rich, I really appreciate it.

--AndrewOn 12/10/05, Rich Adamson [EMAIL PROTECTED] wrote:
Your assumptions are right on the mark. However, keep in mind thatregardless of how much effort you put into trying to figure out whetherteliax is up/down, there are always things that can happen that you can't
discover. For example, teliax (or any other itsp) might accept your outgoingcall and its not processed for whatever reason. Such occurances can only beaddressed if you provide your users with an alternative way to dial. Common
approaches would be to include something like: a) dial 9+digits for an pstncall, b) dial 8+digits for teliax calls, and, c) all 1+digits calls areautomatically routed based on whatever you set up in the dialplan.
Using such an approach essentially has your users dialing whatever numberthey need to (c) under normal conditions, but should there be a problem, theuser can still call outbound by directing their calls to (a) or (b).
Your thought process also addresses 911 calls, etc, by you programming yourdialplan to route those calls via the pstn lines. No need to even thinkabout routing 911 calls via teliax.Keep in mind that whatever you do with fax'ing probably will not work through
voip and the TDM card. Lots of postings in the list archives if you need toresearch that. (Since you are likely to have pstn lines, consider attachinga fax machine to one of those lines and not let asterisk answer incoming calls
on that line. Or, subscribe to an external fax service and have them emailpdf files instead of messing around with paper, toner, questionable faxmachines, modems, etc.) Thank you very much for your responses.I like the idea of having Teliax as well as
some PSTN lines in the event of the T1 going down.I've just started to read the Asterisk book by O'Reilly, so myunderstanding of Asterisk is limited right now.Consequently, if I get a TDM400P for the PSTN lines and get
Teliax, can Asterisk be set up in such a way that if Teliax cannot be reached it uses the PSTN lines?If yes, I'massuming it has to do with the proper diaplan, which I'll be reading up on soon.
 Thanks again for your help, Andrew On 12/10/05, Rich Adamson [EMAIL PROTECTED] wrote:  I'm a beginner here and am interested in Teliax.I own a small business and
was wondering if you guys could help  me out here.I'm basically looking for 6-8 telephone lines, but I notice thatTeliax supports 4 simultaneous calls on  their Corporate plan.So could I get two Corporate plans and set Asterisk to
use both of them and then have, in  essence, 8 people talking at the same time?If someone tries to call, wouldthe phone ring busy or would it still go  through?
   I plan on having a T1. I'd suggest you call their sales folks as teliax is rather flexible; they will likely work something out for you that fits your needs.
 As others have mentioned, the bundled plans (eg, residential or corporate) have a soft cap that essentially translates into $0.018 / minute, assuming you use every single minute within the plan. If you don't use every minute,
 the average cost/min goes up (1,000 minutes of corp plan use = $0.045 / min). So, you are probably better off with their Pay as you go plan which ensures your cost is always $0.02 / min with an unlimited number of
 simultanous calls. If you combine the above with some thought as to what you are going to do when calls can't be completed via teliax (for whatever reason), then you are likely to conclude that having two providers at some flat cost per
 minute is a positive move. If you add to that thought process some probability that you can't complete _any_ Internet-based calls (due to T1 failure or whatever), then you're
 likely to approach a combination of itsp's and pstn lines for your business. ___ --Bandwidth and Colocation provided by 
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[Asterisk-Users] Good Dialing Macros

2005-12-10 Thread gw

Hello All,
I noticed that AAH seems to have a macro setup that if an extension is
unavailable, asterisk will go auto to menu and such.  It seems to do it
before the dial attempt, so it must be using a macro to obtain the
information.  For example, if a call is forwarded on a phone, it can
skip that phone on an incoming hunt group.

What I am looking for is a script to do the following on HEAD.

Rings in
Check an extension, if busy goto queue, if forwarded take a different
route, and if unavailable bounce to a cell phone.  They key is, I need
to be able to either check the status before the dial, or have it ring
multiple phones  methods, such as an IAX  SIP connection
simultaneously.

Does anyone have a good macro in their dialplan with something like
this?  I don't want to have to load up another aah to reverse engineer
it...

Searched the wiki but turned up nothing...

Regards,
Greg
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Re: [Asterisk-Users] agi variables list

2005-12-10 Thread Asterisk [Submusic]

Title: Message



Hi,

try: AGI debug , in you Asterisk 
Console,
You'll see some variables.

an exemple:

AGI Tx  agi_request: get_dnd.agiAGI Tx 
 agi_channel: SIP/3220-bc90AGI Tx  agi_language: frAGI 
Tx  agi_type: SIPAGI Tx  agi_uniqueid: 1134238803.113AGI 
Tx  agi_callerid: 3220AGI Tx  agi_calleridname: Fred 
LaptopAGI Tx  agi_callingpres: 0AGI Tx  agi_callingani2: 
0AGI Tx  agi_callington: 0AGI Tx  agi_callingtns: 
0AGI Tx  agi_dnid: 3202AGI Tx  agi_rdnis: unknownAGI 
Tx  agi_context: macro-stdextenAGI Tx  agi_extension: 
sAGI Tx  agi_priority: 3AGI Tx  agi_enhanced: 0.0AGI 
Tx  agi_accountcode:

Fred

  - Original Message - 
  From: 
  Olivier Taylor 
  To: 'Asterisk Users Mailing List - 
  Non-Commercial Discussion' 
  Sent: Saturday, December 10, 2005 6:44 
  PM
  Subject: [Asterisk-Users] agi variables 
  list
  
  hello all,
  
  where can I find a list of agi variables that 
  can be read by a external script?
  
  Thanks,
  
  Olivier
  
  

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Re: [Asterisk-Users] Problems with zaptel channels not properly being answered...

Comments inline...

 I have a TDM400P card with 2 incoming analog lines on it. I have a  
 working dialplan for 2 IAX lines that work perfectly. I want these 2  
 analog lines to also utilize my dialplan for both incoming and  
 outgoing. However, as my config stands right now, any incoming calls  
 from the analog lines aren't working perfectly yet.
 
 First, the analog calls are seen by Asterisk and Asterisk looks like  
 it's picking the call up and the dialplan executes. I can see this  
 all happening if I run sudo asterisk - from a command line and  
 watch the console print out messages. However, from the analog  
 caller's side, it sounds like Asterisk never picked up as you always  
 hear ringing and that's all. I will place my /etc/zaptel.conf and / 
 etc/asterisk/zapata.conf files inline with this message. Any  
 suggestions are greatly appreciated.
 
 Thanks,
 
 Jim Hodapp
 
  /etc/zaptel.conf -
 loadzone=us
 defaultzone=us
 fxoks=1-2
 fxsks=3-4
 ---
 
 
  /etc/asterisk/zapata.conf -
 [trunkgroups]
 
 [channels]
 
 language=en
 context=default
 busydeteect=yes
 busycount=6
 echotraining=800
 echocancel=yes
 immediate=no
 usecallerid=yes
 callwaiting=yes
 hidecallerid=no
 threewaycalling=yes
 transfer=yes

In case you didn't realize it, the above parameters are inherieted by
the sections below assuming a parameter hasn't been redefined below. 
So, there is no value for having echotraining=800 above, when its
also below (as one example only).

 ;;;[400]
 signalling=fxo_ks
 [EMAIL PROTECTED]
 echotraining=800
 echocancelwhenbridge=no
 echocancel=yes
 context=longdistance
 callprogress=no
 callerid=Bowen  Fenwick400
 busydetect=no
 busycount=7
 channel = 1
 
 [401]
 signalling=fxo_ks
 [EMAIL PROTECTED]
 echotraining=800
 echocancelwhenbridge=no
 echocancel=yes
 context=longdistance
 callprogress=no
 callerid=Conference Room #1401
 busydetect=no
 busycount=7
 channel = 2
 
 group=1
 signalling=fxo_ks   ; Drive a handset or other station device.
 context=default
 channel = 1-2

What are you trying to do with the above? You already defined channels
1 and 2, and now you're doing something more with channel = 1-2?

 group=2
 signalling=fxs_ks   ; Signals the phone company.
 context=longdistance
 channel = 3-4

From what it would appear, channel 3 and 4 are your incoming pstn
lines. The majority of the parameters that you've used on channels
1  2 (telephones) really belong on channels 3  4. Examples include:
buydetect and busycount having nothing to do with telephones, but do
apply to fxo/pstn lines. Same with echotraining=800, echocancelwhenbridge=no
and echocancel=yes.

From what I see above, channels 1  2 are your telephones and you first
defined them to be processed in your extensions.conf longdistance context.
Then you redefined those two channels to be processed in the default
context.

Your incoming pstn calls (channels 3  4) are also processed in the
longdistance context. Without you showing us what the longdistance
context looks like in extensions.conf, its impossible to know what your
trying to accomplish. My guess is that longdistance context (in 
extensions.conf) does not have any dialplan statements in it to handle
the incoming calls.

So, I'd suggest changing your channels 3  4 to something like
 context=pstnline
and in your extensions.conf, put something like this:
[pstnline]
exten = s,1,NoOp,${CALLERID}
exten = s,2,Dial(Zap/1)

Rich


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Re: [Asterisk-Users] Call Routing from GnuGK to Asterisk

2005-12-10 Thread Code Lover

Hi Rafael Marconi,

I follow your configuration but it does not seems to work. and i am
getting some error.

oh323.conf
-
GKID
GnuGk

gatekeeper=asteriskserveronsamemachine.com
gatekeeperTTL=600
userInputMode=TONE
amaFlags=default
accountCode=H323
language=en
musiconhold=default
[register]
alias=asterisk
gwprefix=9
gwprefix=8
gwprefix=7
gwprefix=6
gwprefix=5
gwprefix=4
gwprefix=3
gwprefix=2
gwprefix=1
gwprefix=0
context=from-oh323

my.ini
-
[RasSrv::GWPrefixes]
127.0.0.1
asterisk=0,1,2,3,4,5,6,7,8,9

It is not sending any request on asterisk server.

disengageReject {
requestSeqNum = 1
rejectReason = requestToDropOther null
  }

--
Thank You,
Code Lover
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Re: [Asterisk-Users] Via Epia

2005-12-10 Thread Tzafrir Cohen

On Sat, Dec 10, 2005 at 09:25:47AM -0500, Sergey Okhapkin wrote:
 See
 http://www.epiawiki.org/wiki/tiki-index.php?page=EpiaInstallingGentoo
 regarding CFLAGS settings for different VIA CPUs.

See also http://www.courville.org/mediawiki/index.php/EpiaM for Debian
on Via Epia M.

However you don't have anything much to gain over setting PROC=i586 .
But I'd love to hear from someone of some numbers that prove me wrong.

-- 
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: Teliax experiences

2005-12-10 Thread Asterisk Users - Dovid




I have been using Teliax for several months now 
with no problems what so ever. However I did have problems with Broadvoice. The 
voice quality isnt allways that great. Sometimes the DTMF wouldt work. It was 
very frustrating when I dialed a company over my Broadvoice line and I tried to 
enter a number and nothing happend. Just my 2 cents.
Regards,Dovid
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Re: [Asterisk-Users] What's the best opensource web interface for customer portal

2005-12-10 Thread Dan Littlejohn

On 12/9/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
 I'm looking for a good web interface for a customer portal for a residential
 Voip business.  It should give the customer the ability to set check
 voicemail, set call handling options (forwarding, blocking, do not disturb,
 etc), check usage, pay bills etc.  I would like it if it were comparable to
 the user portal for companies like Broadvoice, Vonage, Voicepulse, or
 Sunrocket.

 Does anyone know of a good opensource solution for this? Please don't
 suggest commercial packages unless they are really cheap (under $500), and
 head and shoulders above the opensource solutions.


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ARI does most of what you need, although it does not yet have all of
those features.
  http://www.littlejohnconsulting.com/?q=ari

You could develop a spec and sponsor development if you wanted to stay
with an open source solution.

Dan Littlejohn
512.791.0137
littlejohnconsulting.com
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[Asterisk-Users] extensions and regular expressions ( probably an easy question )


Hi all,

I'm having a hard time finding information related to the regular 
expressions that can be used in a dialplan, specifically as an 
extension.  For example, I have an 800 number which I'd like to jump 
directly to if my users dial it, instead of going over my pstn 
termination.  Currently, it looks like this:


exten = 8661234567,1,Goto(800-in)

However, I'd like 1866123456 to match as well.  I can't find in the wiki 
or sample configs how to say match this 0 or 1 times. 

Can anybody provide a link that would go over this?  Again, I've been 
digging through the wiki, but I seem to be missing it.


Thanks

Sean

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[Asterisk-Users] Echo on incoming sip provider- asterisk - sip phone / ata

2005-12-10 Thread Bernard


Hi,

I am having intermittend problems with echo on the line (sometimes a 
clear connections, other times lots of echo on the line), especially on 
incoming connections from my sip provider.
Dialing out through this provider (budgetphone.nl), is most of the time 
no problem, dialing out through other providers (like voipbuster) is 
more stable.


On my client side I am using  Sipura 2002 and 3000 devices and  a 
Grandstream  2000.

The Asterisk is running on:
Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k built by [EMAIL PROTECTED] on a x86_64 running 
Linux


1.
Is it possible to improve sound quality by tweaking the Asterisk, or is 
this echo a problem between the Sip provider and the client devices on 
my side?

2.
How can I check (preferably in the logs) which codecs  were used for a 
specific session.

3.
Is it possible to force incoing connections form an specific  incoming 
connection  to us a specific codec? (I am currently using a list of 
preferred codecs ulaw, alaw and gsm.)

4.
What ata's are better in removing echo???

Any help apreciated.

Bernard.


--
No virus found in this outgoing message.
Checked by AVG Free Edition.
Version: 7.1.371 / Virus Database: 267.13.13/197 - Release Date: 09/12/2005

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Re: [Asterisk-Users] What's the best opensource web interface for customer portal

2005-12-10 Thread Tom Vile

I too like ARI and cant wait for some of the additional options that are coming.

On 12/10/05, Dan Littlejohn [EMAIL PROTECTED] wrote:
 On 12/9/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
  I'm looking for a good web interface for a customer portal for a residential
  Voip business.  It should give the customer the ability to set check
  voicemail, set call handling options (forwarding, blocking, do not disturb,
  etc), check usage, pay bills etc.  I would like it if it were comparable to
  the user portal for companies like Broadvoice, Vonage, Voicepulse, or
  Sunrocket.
 
  Does anyone know of a good opensource solution for this? Please don't
  suggest commercial packages unless they are really cheap (under $500), and
  head and shoulders above the opensource solutions.
 
 
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 http://lists.digium.com/mailman/listinfo/asterisk-users
 
 ARI does most of what you need, although it does not yet have all of
 those features.
  http://www.littlejohnconsulting.com/?q=ari

 You could develop a spec and sponsor development if you wanted to stay
 with an open source solution.

 Dan Littlejohn
 512.791.0137
 littlejohnconsulting.com
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--
Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone: 518-631-2855 x205
Phone: 978-203-3848 x205
Fax: 518-631-2856
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[Asterisk-Users] RE: chan_grab.c for version 1.0X

2005-12-10 Thread Ronald Hartmann

Anyone who would be so kind to send me a copy of app_changrab.c would
absolutemy make my day.

Thanks in advanced, as I have googled and googled and no go.

~ron 

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Re: [Asterisk-Users] Re: Asterisk-Users Digest, Vol 17, Issue 61

2005-12-10 Thread Brian Capouch


Steve Totaro wrote:



I once met a guy with a degree in Computer Science from a pretty
respectable college and he could not even log into a Windows machine.



I'm a *professor* of CS, and I cannot even log onto a Window$ machine.

I've never owned one, and never will, if I can help it.

I've never shot heroin, either, nor ridden in a Ferrari.

What does your statement have to do with *anyone's* knowledge of CS?

B.
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Re: [Asterisk-Users] Via Epia

2005-12-10 Thread Andrew Nowrot

Hi,

I use VIA-C3 Processor family for ezra CPU. Does it make my situation
any better? I managed to compile a new kernel 2.4.30 on this Via Epia.
I have also installed Asterisk with no problems but the after the
start I get -- illegal instruction8(.

If the PROC=i5(6)86 will not change anything what should I do make * run?

Cheers

Andrew
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Re: [Asterisk-Users] Re: Asterisk-Users Digest, Vol 17, Issue 61

2005-12-10 Thread pdhales




I am heppy to meet up for a coffee.

I am in Melbourne, Australia. Where are 
you?

PaulH

  - Original Message - 
  From: 
  Abhishek Gangal 
  To: asterisk-users@lists.digium.com 
  
  Sent: Sunday, December 11, 2005 3:08 
  AM
  Subject: [Asterisk-Users] Re: 
  Asterisk-Users Digest, Vol 17, Issue 61
  
  Sir
   I am a novice user and want to set up the asterix for only 
  Voip as a project in my final yr. computer engineeering.
   Pl. help me to do so . I will 
  be highly thankful
  
  
  Abhishek Gangal
  
  

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[Asterisk-Users] Using SIP_HEADER() Function correctly

2005-12-10 Thread Rushowr

(oops, wrong account, let's try again, without the work email getting it
blocked)

Am I correct in my thought that if I was to issue:

Exten = 1234,1,VERBOSE(1|${SIP_HEADER(HEADERNAME)})

In the dialplan that asterisk would return the value of the HEADERNAME
field, if there was one attached to the invite?

I've been hunting around, but have found no working examples of this. 

Thanks

SKM

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[Asterisk-Users] Updated Guide to SPA-3000

2005-12-10 Thread Kerry Garrison

With some of the newer versions of Asterisk and AMP, many people have been
having problems getting the Linksys/Sipura SPA-3000 working properly. We
have just posted an all-new guide to getting the SPA-3000 up and running.

http://voipspeak.net/index.php?option=com_contenttask=viewid=51Itemid=28

-Kerry


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Re: [Asterisk-Users] extensions and regular expressions ( probably an easy question )

2005-12-10 Thread Daniel Wright


Sean Kennedy wrote:

Hi all,

I'm having a hard time finding information related to the regular 
expressions that can be used in a dialplan, specifically as an 
extension.  For example, I have an 800 number which I'd like to jump 
directly to if my users dial it, instead of going over my pstn 
termination.  Currently, it looks like this:


exten = 8661234567,1,Goto(800-in)

However, I'd like 1866123456 to match as well.  I can't find in the 
wiki or sample configs how to say match this 0 or 1 times.
Can anybody provide a link that would go over this?  Again, I've been 
digging through the wiki, but I seem to be missing it.


Thanks

Sean


You could do it like this:

exten = _866.,1,GoTo(800-in)

The period means match one or more characters.

You can find reference to expressions and how they work  in this pdf 
book http://www.nufone.net/downloads/asteriskdocs/AsteriskTFOT.zip


Dan
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[Asterisk-Users] mpg123 on x86_64 (Opteron MP)

2005-12-10 Thread Brad


   Subject: Re: [Asterisk-Users] mpg123 on x86_64 (Opteron MP)
   Joseph wrote:
   Why do you need to compile it?
   Isn't it available as an rpm package?
  
   I will assume he knows why he needs to compile it.
  
   See if the source for the rpm, deb, or whatever from the distro you
   are
   running will build for you. That will often get your system to the
   point
   where any header, tools and libraries needed are now installed. That's
   how I do it when I want to use something from cvs.

 I'm compiling it because Redhat (FC3) uses mpg321 in the
 distribution, which doesn't work with *.  I can build *, zaptel, and
 libprc fine, as well as other apps (FC3 was not compiled, but
 installed from RPM).

 Looking at the compile error, I'm thinking there is a problem with an
 assembler subroutine (push and pop complaints) on the Opteron
 processor, or a compile flag that should be changed.

 Still stuck.

 Original Post:
 Has anyone successfully compiled mpg123 in the 1.0.x or 1.2beta1
 distributions (I'm running FC3 linux on an Opteron 2 processor
 system)?  Are there any patches out there to make it work?

 gcc -DI386_ASSEM -DPENTIUM_OPT -DREAL_IS_FLOAT -DLINUX  -
 DREAD_MMAP -DOSS -DTERM_CONTROL-Wall -O2 -
 m486 -fomit-frame-pointer -funroll-all-
 loops  -finline-functions -ffast-math   -c -o
 dct64_i386.o dct64_i386.c
 as   -o decode_i586.o decode_i586.s
 decode_i586.s: Assembler messages:
 decode_i586.s:44: Error: suffix or operands invalid for `push'
 snip
 decode_i586.s:161: Error: suffix or operands invalid for `pop'
 snip


I am having the same issues as outlined above. The processor is a semperon.

# uname -a
Linux amd64 2.6.12-9-amd64-generic #1 Sat Oct 1 01:11:30 BST 2005 x86_64 
GNU/Linux


When trying to compile mpg123 I get the following errors:

[EMAIL PROTECTED]:/usr/src/asterisk-1.0.10/mpg123-0.59r# make linux

...
make[2]: Entering directory `/usr/src/asterisk-1.0.10/mpg123-0.59r'
as   -o decode_i586.o decode_i586.s
decode_i586.s: Assembler messages:
decode_i586.s:44: Error: suffix or operands invalid for `push'
decode_i586.s:45: Error: suffix or operands invalid for `push'
decode_i586.s:46: Error: suffix or operands invalid for `push'
decode_i586.s:47: Error: suffix or operands invalid for `push'
decode_i586.s:67: Error: suffix or operands invalid for `push'
decode_i586.s:70: Error: suffix or operands invalid for `push'
decode_i586.s:81: Error: suffix or operands invalid for `push'
decode_i586.s:83: Error: suffix or operands invalid for `push'
decode_i586.s:86: Error: suffix or operands invalid for `push'
decode_i586.s:161: Error: suffix or operands invalid for `pop'
decode_i586.s:211: Error: suffix or operands invalid for `pop'
decode_i586.s:296: Error: suffix or operands invalid for `pop'
decode_i586.s:315: Error: suffix or operands invalid for `pop'
decode_i586.s:316: Error: suffix or operands invalid for `pop'
decode_i586.s:317: Error: suffix or operands invalid for `pop'
decode_i586.s:318: Error: suffix or operands invalid for `pop'
make[2]: *** [decode_i586.o] Error 1
make[2]: Leaving directory `/usr/src/asterisk-1.0.10/mpg123-0.59r'
make[1]: *** [mpg123-make] Error 2
make[1]: Leaving directory `/usr/src/asterisk-1.0.10/mpg123-0.59r'
make: *** [linux] Error 2

Anyone able to point me in the right direction to compile this app? It 
is running ubuntu..


Thanks

Brad
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Re: [Asterisk-Users] extensions and regular expressions ( probably an easy question )


Hi Dan,

Thanks for the info, but what I'm after is the ability to match a 
digit/character 0 or 1 times at the beginning of the string.  If I'm 
reading your example right, it'll match anything starting with 866, 
which doesn't work for me.  I am trying to match:


18661234567 and 8661234567

Sean

ps:  The pdf doesn't have a good explaination of this either, although 
it occurs to me that this might not be possible with * if I'm having 
such a hard time finding it.

Daniel Wright wrote:


Sean Kennedy wrote:


Hi all,

I'm having a hard time finding information related to the regular 
expressions that can be used in a dialplan, specifically as an 
extension.  For example, I have an 800 number which I'd like to jump 
directly to if my users dial it, instead of going over my pstn 
termination.  Currently, it looks like this:


exten = 8661234567,1,Goto(800-in)

However, I'd like 1866123456 to match as well.  I can't find in the 
wiki or sample configs how to say match this 0 or 1 times.
Can anybody provide a link that would go over this?  Again, I've been 
digging through the wiki, but I seem to be missing it.


Thanks

Sean


You could do it like this:

exten = _866.,1,GoTo(800-in)

The period means match one or more characters.

You can find reference to expressions and how they work  in this pdf 
book http://www.nufone.net/downloads/asteriskdocs/AsteriskTFOT.zip


Dan



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RE: [Asterisk-Users] extensions and regular expressions ( probablyan easy question )

You just need separate extensions.


 
 Hi Dan,
 
 Thanks for the info, but what I'm after is the ability to match a
 digit/character 0 or 1 times at the beginning of the string.  If I'm
 reading your example right, it'll match anything starting with 866,
 which doesn't work for me.  I am trying to match:
 
 18661234567 and 8661234567
 
 Sean
 
 ps:  The pdf doesn't have a good explaination of this either, although
 it occurs to me that this might not be possible with * if I'm having
 such a hard time finding it.
 Daniel Wright wrote:
 
  Sean Kennedy wrote:
 
  Hi all,
 
  I'm having a hard time finding information related to the regular
  expressions that can be used in a dialplan, specifically as an
  extension.  For example, I have an 800 number which I'd like to
jump
  directly to if my users dial it, instead of going over my pstn
  termination.  Currently, it looks like this:
 
  exten = 8661234567,1,Goto(800-in)
 
  However, I'd like 1866123456 to match as well.  I can't find in the
  wiki or sample configs how to say match this 0 or 1 times.
  Can anybody provide a link that would go over this?  Again, I've
been
  digging through the wiki, but I seem to be missing it.
 
  Thanks
 
  Sean
 
  You could do it like this:
 
  exten = _866.,1,GoTo(800-in)
 
  The period means match one or more characters.
 
  You can find reference to expressions and how they work  in this pdf
  book http://www.nufone.net/downloads/asteriskdocs/AsteriskTFOT.zip
 
  Dan
 
 
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Re: [Asterisk-Users] extensions and regular expressions ( probably an easy question )

2005-12-10 Thread Daniel Wright

Sorry I read that wrong.

You could use something like this or just put two lines for each extension

exten = s,1,GotoIf($[$[${EXTEN:0:4} = 1866]?2:3)
exten = s,n,GoTo(wherever1)
exten = s,n,GoToif($[${EXTEN:0:3} = 866]?4:5)
exten = s,n GoTo(Wherever2)
exten = s,n,What-ever-could-come-next

You can also change the jump 2:3/4:5 to the contexts you want to go to. I am fairly new to the dialplan so not sure if this will work exactly like you
need, but this should give you an idea of where to go from here.

Hope this helps

Dan

Sean Kennedy wrote:

Hi Dan,

Thanks for the info, but what I'm after is the ability to match a
digit/character 0 or 1 times at the beginning of the string. If I'm
reading your example right, it'll match anything starting with 866,
which doesn't work for me. I am trying to match:

18661234567 and 8661234567

Sean

ps: The pdf doesn't have a good explaination of this either, although
it occurs to me that this might not be possible with * if I'm having
such a hard time finding it.

Daniel Wright wrote:

Sean Kennedy wrote:

Hi all,

I'm having a hard time finding information related to the regular
expressions that can be used in a dialplan, specifically as an
extension. For example, I have an 800 number which I'd like to jump
directly to if my users dial it, instead of going over my pstn
termination. Currently, it looks like this:

exten = 8661234567,1,Goto(800-in)

However, I'd like 1866123456 to match as well. I can't find in the
wiki or sample configs how to say match this 0 or 1 times.
Can anybody provide a link that would go over this? Again, I've
been digging through the wiki, but I seem to be missing it.

Thanks

Sean

You could do it like this:

exten = _866.,1,GoTo(800-in)

The period means match one or more characters.

You can find reference to expressions and how they work in this pdf
book http://www.nufone.net/downloads/asteriskdocs/AsteriskTFOT.zip

Dan

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RE: [Asterisk-Users] extensions and regular expressions ( probablyan easy question )

2005-12-10 Thread Rushowr

Definitely should just copy the 18661234567 extension set and then remove
the 1. I do not believe you can use regular expressions (I take it you _do_
mean regex, and not the standard pattern matching)

That's all I got...

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sean Kennedy
Sent: Saturday, December 10, 2005 5:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] extensions and regular expressions (
probablyan easy question )

Hi Dan,

Thanks for the info, but what I'm after is the ability to match a
digit/character 0 or 1 times at the beginning of the string.  If I'm reading
your example right, it'll match anything starting with 866, which doesn't
work for me.  I am trying to match:

18661234567 and 8661234567

Sean

ps:  The pdf doesn't have a good explaination of this either, although it
occurs to me that this might not be possible with * if I'm having such a
hard time finding it.
Daniel Wright wrote:

 Sean Kennedy wrote:

 Hi all,

 I'm having a hard time finding information related to the regular 
 expressions that can be used in a dialplan, specifically as an 
 extension.  For example, I have an 800 number which I'd like to jump 
 directly to if my users dial it, instead of going over my pstn 
 termination.  Currently, it looks like this:

 exten = 8661234567,1,Goto(800-in)

 However, I'd like 1866123456 to match as well.  I can't find in the 
 wiki or sample configs how to say match this 0 or 1 times.
 Can anybody provide a link that would go over this?  Again, I've been 
 digging through the wiki, but I seem to be missing it.

 Thanks

 Sean

 You could do it like this:

 exten = _866.,1,GoTo(800-in)

 The period means match one or more characters.

 You can find reference to expressions and how they work  in this pdf 
 book http://www.nufone.net/downloads/asteriskdocs/AsteriskTFOT.zip

 Dan


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Re: [Asterisk-Users] extensions and regular expressions ( probably an easy question )


Or, just do...
exten = 18661234567,1,Goto(800-in)
exten = 8661234567,1,Goto(800-in)

It's kind of tough to truly understand what you are trying to accomplish
(or ask for). Apparently you've got something more in mind that words 
are making it through the list. Reading between the lines, it would 
appear from the 800-in that calls are coming in from some external 
source, and you trying to do something with them. Can you be a little 
more explicit.





Hi Dan,

Thanks for the info, but what I'm after is the ability to match a 
digit/character 0 or 1 times at the beginning of the string.  If I'm 
reading your example right, it'll match anything starting with 866, 
which doesn't work for me.  I am trying to match:


18661234567 and 8661234567

Sean

ps:  The pdf doesn't have a good explaination of this either, although 
it occurs to me that this might not be possible with * if I'm having 
such a hard time finding it.

Daniel Wright wrote:


Sean Kennedy wrote:


Hi all,

I'm having a hard time finding information related to the regular 
expressions that can be used in a dialplan, specifically as an 
extension.  For example, I have an 800 number which I'd like to jump 
directly to if my users dial it, instead of going over my pstn 
termination.  Currently, it looks like this:


exten = 8661234567,1,Goto(800-in)

However, I'd like 1866123456 to match as well.  I can't find in the 
wiki or sample configs how to say match this 0 or 1 times.
Can anybody provide a link that would go over this?  Again, I've been 
digging through the wiki, but I seem to be missing it.


Thanks

Sean


You could do it like this:

exten = _866.,1,GoTo(800-in)

The period means match one or more characters.

You can find reference to expressions and how they work  in this pdf 
book http://www.nufone.net/downloads/asteriskdocs/AsteriskTFOT.zip


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Re: [Asterisk-Users] Re: Asterisk-Users Digest, Vol 17, Issue 56

2005-12-10 Thread C F

PaulH,
Emacs works over http?

On 12/10/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
 bah

 Emacs

 PaulH

 - Original Message -
 From: C F [EMAIL PROTECTED]
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Sent: Saturday, December 10, 2005 5:06 AM
 Subject: Re: [Asterisk-Users] Re: Asterisk-Users Digest, Vol 17, Issue 56


  I'm trying to figure out why you changed the subject?
  Anyhow, thirdlane makes something called asterisk PBX Manager. There
  is also another tool but it doesn't work (AFAIK) over http, amongst
  others that tool can:
  * Show you all the contexts
  * Show you all the extensions, and the DP that drive them
  * Show every sinlge configuration file that asterisk, or your linux
  system might use
  * Allow you edit contexts
  * Allow you to edit extensions
  * Allow you to edit every single configuration file on your entire system
  * Allow you to add contexts
  * Allow you to add extensions
  * Allow you to add configuration files
  * Allow you to delete contexts
  * Allow you to delete extensions
  * Allow you to delete items within any configuration file on your entire
 system.
 
  That tool is called vi
 
 
  P.S. If you find anything that can do all of the above over http
  please post them (but for Webmin file manager). Thank You
 
  On 12/9/05, James Horn [EMAIL PROTECTED] wrote:
   CM is the Cisco Call Manager and Astericks is the Asterisk Software.
  
  
  
-- Forwarded message --
From: C F [EMAIL PROTECTED] 
To: Asterisk Users Mailing List - Non-Commercial Discussion
   asterisk-users@lists.digium.com
Date: Fri, 9 Dec 2005 11:08:48 -0500
Subject: Re: [Asterisk-Users] Phone Information
Can you please explain?
Whats CM?
Whats Astericks?
  
  
  
  
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Re: [Asterisk-Users] extensions and regular expressions ( probablyan easy question )





Steve,

Yeah, that's what I've been doing, but I was hoping to make it a little
clearer in the dial plan.

Ah well, you win some and lose some. Thanks!

Sean

Steve Totaro wrote:

  You just need separate extensions.


  
  
Hi Dan,

Thanks for the info, but what I'm after is the ability to match a
digit/character 0 or 1 times at the beginning of the string.  If I'm
reading your example right, it'll match anything starting with 866,
which doesn't work for me.  I am trying to match:

18661234567 and 8661234567

Sean

ps:  The pdf doesn't have a good explaination of this either, although
it occurs to me that this might not be possible with * if I'm having
such a hard time finding it.
Daniel Wright wrote:



  Sean Kennedy wrote:

  
  
Hi all,

I'm having a hard time finding information related to the regular
expressions that can be used in a dialplan, specifically as an
extension.  For example, I have an 800 number which I'd like to

  

  
  jump
  
  

  
directly to if my users dial it, instead of going over my pstn
termination.  Currently, it looks like this:

exten = 8661234567,1,Goto(800-in)

However, I'd like 1866123456 to match as well.  I can't find in the
wiki or sample configs how to say "match this 0 or 1 times".
Can anybody provide a link that would go over this?  Again, I've

  

  
  been
  
  

  
digging through the wiki, but I seem to be missing it.

Thanks

Sean


  
  You could do it like this:

exten = _866.,1,GoTo(800-in)

The period means match one or more characters.

You can find reference to expressions and how they work  in this pdf
book http://www.nufone.net/downloads/asteriskdocs/AsteriskTFOT.zip

Dan
  


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Re: [Asterisk-Users] Re: Asterisk-Users Digest, Vol 17, Issue 56

2005-12-10 Thread pdhales

A good questionand not one I know the answer to...

PaulH

- Original Message - 
From: C F [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, December 11, 2005 12:22 PM
Subject: Re: [Asterisk-Users] Re: Asterisk-Users Digest, Vol 17, Issue 56

 PaulH,
 Emacs works over http?

 On 12/10/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
  bah

  Emacs

  PaulH

  - Original Message -
  From: C F [EMAIL PROTECTED]
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  asterisk-users@lists.digium.com
  Sent: Saturday, December 10, 2005 5:06 AM
  Subject: Re: [Asterisk-Users] Re: Asterisk-Users Digest, Vol 17, Issue
56

   I'm trying to figure out why you changed the subject?
   Anyhow, thirdlane makes something called asterisk PBX Manager. There
   is also another tool but it doesn't work (AFAIK) over http, amongst
   others that tool can:
   * Show you all the contexts
   * Show you all the extensions, and the DP that drive them
   * Show every sinlge configuration file that asterisk, or your linux
   system might use
   * Allow you edit contexts
   * Allow you to edit extensions
   * Allow you to edit every single configuration file on your entire
system
   * Allow you to add contexts
   * Allow you to add extensions
   * Allow you to add configuration files
   * Allow you to delete contexts
   * Allow you to delete extensions
   * Allow you to delete items within any configuration file on your
entire
  system.

   That tool is called vi

   P.S. If you find anything that can do all of the above over http
   please post them (but for Webmin file manager). Thank You

   On 12/9/05, James Horn [EMAIL PROTECTED] wrote:
CM is the Cisco Call Manager and Astericks is the Asterisk Software.

 -- Forwarded message --
 From: C F [EMAIL PROTECTED] 
 To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
 Date: Fri, 9 Dec 2005 11:08:48 -0500
 Subject: Re: [Asterisk-Users] Phone Information
 Can you please explain?
 Whats CM?
 Whats Astericks?

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Re: [Asterisk-Users] a few questions

2005-12-10 Thread C F

On 12/9/05, Kerry Garrison [EMAIL PROTECTED] wrote:
 On 12/9/05, Kerry Garrison [EMAIL PROTECTED] wrote:
  Overhead paging is totally possible, there are several articles
  available on how to do it. But you cannot have multiple zones today
  unless you use a sip device that has autoanswer.
 

 Why can mutilple zones not be done?, why do I need a
 sip device at all for the paging? any of the follwing (and I'm sure
 more) will do, even for multiple zones:
 * PC Sound Card
 * Digum hardware
 * any type of ata type gateway (SIP/h323 or whatever else that will
 interface with an analog 
 port), even one without auto answer


 I have yet to see an example of overhead paging with multiple zones using a
 soundcard, digium hardware, or an ata.
 -Kerry



Because you have never seen it, and you don't have the skill to figure
it out, therefore it never happened. Nice job. Are you a politician?

If you wish to pay my fee, I can give you a tour to a few buildings
where I have successfully done it.
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[Asterisk-Users] Setting Request URI

2005-12-10 Thread Douglas Garstang

Does anyone know how to set the request URI of SIP messages being sent from 
Asterisk to a peer? Asterisk always puts the IP address or hostname of the peer 
in the request URI. Eventhough Asterisk's SRV lookups are broken, I'd really 
like to put a domain name in the request URI (makes OpenSER routing easier and 
more logical).
 
Anyone know how to do this?
 
Doug.
 
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Re: [Asterisk-Users] Setting Request URI

2005-12-10 Thread Marc Storck


Hello Douglas,

I don't know if this is exactly what you need, but the fromdomain and 
fromuser in sip.conf (explained here: 
http://www.voip-info.org/wiki-Asterisk+config+sip.conf) change the From: 
header to [EMAIL PROTECTED]


Regards,

Marc

Douglas Garstang wrote:

Does anyone know how to set the request URI of SIP messages being sent from 
Asterisk to a peer? Asterisk always puts the IP address or hostname of the peer 
in the request URI. Eventhough Asterisk's SRV lookups are broken, I'd really 
like to put a domain name in the request URI (makes OpenSER routing easier and 
more logical).
 
Anyone know how to do this?
 
Doug.
 





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Re: [Asterisk-Users] Echo PSTN [EMAIL PROTECTED] 2.0 Digium TDM11B DSL

2005-12-10 Thread David K Parker

I hadn't heard of fxotune. I'll check it out. I had a little bit better luck today after replacing a 25 ft cable on the pstn side with a 7 ft cable. I'm beginning to wonder about ztmonitor though. Before replacing the cable I had to adjust Rx to 22 and Tx to -
7.5. Ztmonitor still showed the Tx gain to be hot. If I went below -7.5 I couldn't complete a call. Now Rx is at 2.5 and Tx is at 8. This is adjusted when caling the CO. When I call anyone, the sound is low. I can adjust Rx to
4.5 and its better. The other party hears me fine. I changed the echo canceller to ECHO_CAN_MG2 in Zaptel and the beginning of the call isn't as bad. If I unplug Asterisk from the PSTN and use the analog direct to the telco the quality is fine. Asterisk is poor in comparison on PSTN. Any calls with Asterisk to my Teliax ld trunk are fine.
On 12/9/05, Matthew Fredrickson [EMAIL PROTECTED] wrote:
On Dec 9, 2005, at 9:48 AM, David K Parker wrote: I have a Digium TDM11B, I'm fighting an issue with with echo on the PSTN side. I run [EMAIL PROTECTED] 2.0. I have an analog phone on the FXS channel 1 and Telco on the FXS channel 4. I also have a coupe of
softphones, 1 iax2, the other sip, and a LinkSys Sipura 941. I use a VOIP provider for long distance. I'm experiencing echo on all calls on any phone for calls going out over the PSTN, but no echo at all on
Long Distance calls with my VOIP provider or Internal calls. I think its safe to say that echo is occuring on the PSTN side on channel 4. I've followed the trouble shooting provedures on
voip-info.org for echo cancellation, even calling the local CO using ztmonitor to adjust rx tx gain. The only thing I haven't tried yet is installing shielded cable. I use Verizon DSL for Internet and have the
appropriate filter for my PSTN on channel 4. I'm beginning to wonder if the problem is due to DSL. Has anyone else had this experience.Have you tried running fxotune on the card?It's possible that your
echo problem is related to line impedance mismatch.For more details,see README.fxotune in the zaptel package.Matthew Fredrickson___--Bandwidth and Colocation provided by
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Re: [Asterisk-Users] extensions and regular expressions ( probablyan easy question )

2005-12-10 Thread Ryan Burke

I was reading the pdf and found a command that might be of some use: 
Prefix()


ex.

exten = 8661234567,1,Prefix(1)
exten = 18661234567,1,NoOp()
exten = 18661234567,2,Goto(800-in)

After the Prefix() the the next exten is n+1 (which is 2 in this example) 
with the new extension (which is 18661234567 instead of 8661234567 which was 
originally dialed).


Personally I think this is a bit more elegant than having a bunch of Goto's 
for each extension, but they do the same thing. Personal preference I guess.


Ryan

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

Sent: Saturday, December 10, 2005 7:08 PM
Subject: Re: [Asterisk-Users] extensions and regular expressions ( 
probablyan easy question )




Or, just do...
exten = 18661234567,1,Goto(800-in)
exten = 8661234567,1,Goto(800-in)

It's kind of tough to truly understand what you are trying to accomplish
(or ask for). Apparently you've got something more in mind that words are 
making it through the list. Reading between the lines, it would appear 
from the 800-in that calls are coming in from some external source, and 
you trying to do something with them. Can you be a little more explicit.





Hi Dan,

Thanks for the info, but what I'm after is the ability to match a 
digit/character 0 or 1 times at the beginning of the string.  If I'm 
reading your example right, it'll match anything starting with 866, which 
doesn't work for me.  I am trying to match:


18661234567 and 8661234567

Sean

ps:  The pdf doesn't have a good explaination of this either, although it 
occurs to me that this might not be possible with * if I'm having such a 
hard time finding it.

Daniel Wright wrote:


Sean Kennedy wrote:


Hi all,

I'm having a hard time finding information related to the regular 
expressions that can be used in a dialplan, specifically as an 
extension.  For example, I have an 800 number which I'd like to jump 
directly to if my users dial it, instead of going over my pstn 
termination.  Currently, it looks like this:


exten = 8661234567,1,Goto(800-in)

However, I'd like 1866123456 to match as well.  I can't find in the 
wiki or sample configs how to say match this 0 or 1 times.
Can anybody provide a link that would go over this?  Again, I've been 
digging through the wiki, but I seem to be missing it.


Thanks

Sean


You could do it like this:

exten = _866.,1,GoTo(800-in)

The period means match one or more characters.

You can find reference to expressions and how they work  in this pdf 
book http://www.nufone.net/downloads/asteriskdocs/AsteriskTFOT.zip


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Re: [Asterisk-Users] extensions and regular expressions ( probably an easy question )