date:20051211

Asterisk is really pissing me off.
Can someone tell me why this doesn't cause SRV lookups to be done on outbound 
calls:

[general]
srvlookup=yes
...

[proxy]
type=peer
host=pstn.voip.com
insecure=very
context=test
qualify=yes

exten = s,2,Dial(SIP/[EMAIL PROTECTED],20,rt)

NO SRV LOOKUP!

While the following DOES cause an SRV lookup to be done...

exten = s,2,Dial(SIP/[EMAIL PROTECTED],20,rt)

So... if I put the domain directly into the dial command an SRV lookup is done. 
If I reference it in sip.conf, an SRV lookup *IS NOT* done. WTF???
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Re: [Asterisk-Users] Dialing analog extensions from SIP?

2005-12-11 Thread Doug Lytle

 Is it possible to group all analog (regular phone) extensions so 
that you can dial it from a SIP extension?   i.e.  for use as an intercom


 I tried this:

 [default]
 exten = #3001,1,Dial(Zap/1,25,t,r)
 exten = #3001,2,Hangup


Change your dial to:

exten = #3001,1,Dial(ZAP/25,tr)

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] outgoing calls that last an unreasonably long time

2005-12-11 Thread Simone Cittadini


Warren Burstein ha scritto:




What is frustrating is that the cdr file shows the dst as T rather 
than as the phone number dialed.  I realize that AbsoluteTimout causes 
it to jump to the T extension, but it would help to know who the user 
dialed (asking a week later isn't going to get any useful information 
out of the user).  It's not in the log file, either - would increasing 
the log level help here?



I don't know how this AbsolutTimeout works, anyway I put all the info I 
need in variables before the actual Dial, then in the h extension I call 
SetUserField() (or whatever is called), helps me keeping track of 
reasons for non-terminated calls ...


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Re: [Asterisk-Users] Dialing analog extensions from SIP?

2005-12-11 Thread Robert La Ferla


Doug Lytle wrote:
 Is it possible to group all analog (regular phone) extensions so 
that you can dial it from a SIP extension?   i.e.  for use as an intercom


 I tried this:

 [default]
 exten = #3001,1,Dial(Zap/1,25,t,r)
 exten = #3001,2,Hangup


Change your dial to:

exten = #3001,1,Dial(ZAP/25,tr)

Doug


This didn't work.  I still get Call Failed followed by a fast busy tone.

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Re: [Asterisk-Users] Dialing analog extensions from SIP?

2005-12-11 Thread Eric \ManxPower\ Wieling

The phone's built in dialplan is prolly blocking the call.  Check the 
docs for your SIP device.  Remember SIP devices collect all digits, then 
pass them on to Asterisk as one packet.


Also what Zap port is your analog phone connected to?  What card are you 
using?


Robert La Ferla wrote:

Doug Lytle wrote:
 Is it possible to group all analog (regular phone) extensions so 
that you can dial it from a SIP extension?   i.e.  for use as an intercom


 I tried this:

 [default]
 exten = #3001,1,Dial(Zap/1,25,t,r)
 exten = #3001,2,Hangup


Change your dial to:

exten = #3001,1,Dial(ZAP/25,tr)

Doug


This didn't work.  I still get Call Failed followed by a fast busy tone.

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Re: [Asterisk-Users] Dialing analog extensions from SIP?

2005-12-11 Thread Robert La Ferla


Eric ManxPower Wieling wrote:
The phone's built in dialplan is prolly blocking the call.  Check the 
docs for your SIP device.  Remember SIP devices collect all digits, 
then pass them on to Asterisk as one packet.


Also what Zap port is your analog phone connected to?  What card are 
you using?

Thanks.  I'm using the Digium TDM11B card and an Aastra 9133i SIP phone.

The phone has a local dial plan set to this:

X+#|XX+*

I'm not sure what this regex means???

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Re: [Asterisk-Users] Dialing analog extensions from SIP?

2005-12-11 Thread Doug Lytle


Eric ManxPower Wieling wrote:
The phone's built in dialplan is prolly blocking the call.  Check the 
docs for your SIP device.  Remember SIP devices collect all digits, 
then pass them on to Asterisk as one packet.



I agree with Eric on this one.  On my Polycom IP501s, I had to change the digit 
map to allow for # and * matching.  For testing, remove the # and try again.

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] SRV Lookups ARRGH!

2005-12-11 Thread Philipp von Klitzing

Dear Douglas!

 Asterisk is really pissing me off.
 Can someone tell me why this doesn't cause SRV lookups to be done on
 outbound calls: 

In general: If you are missing documentation then you are warmly invited 
to write and enhance the existing one (e.g. the Wiki) wherever you see 
fit. In particular you might want to edit this and add what you (and me) 
have learned:

   http://www.voip-info.org/wiki/view/Asterisk+SIP+srvlookup

 So... if I put the domain directly into the dial command an SRV lookup
 is done. If I reference it in sip.conf, an SRV lookup *IS NOT* done.
 WTF??? 

I did some work for you and searched bugs.digium.com for srvlookup. And 
look at what I found:

  http://bugs.digium.com/view.php?id=1805
  http://bugs.digium.com/view.php?id=2081

Reading the bug notes you'll find that this is known - and probably even 
intended - behaviour. If you dislike it: File a bug report, write a 
patch, or find someone that's going to write it for you. Asterisk is an 
open source project, remember?

Cheers, Philipp


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RE: [Asterisk-Users] SRV Lookups ARRGH!

Phillip.

The link to the Wiki is woefully indadequate. I have no problem adding to the 
documentation, as soon as I bloody understand it myself. 

The two bug links you provided appear to be almost completely unrelated to what 
I asked about except they touch on the subject of SRV lookups. If you can't 
reference the proxy to dial in sip.conf, then you lose the ability to set a 
whole bunch of options (such as qualify which is required for detecting 
CONGESTION when the proxy is down etc).

If I stick with an IP/host and refer to what's in sip.conf, Asterisk ends URI's 
like sip:[EMAIL PROTECTED] which is but ugly and breaks SIP in general.

Oh, and you know what, why is it assumed that to use open source software I 
have to be a seasoned C programmer who can contribute to the code? Where is 
that requirement stipulated?

At this point I'm almost ready to throw Asterisk out the window and suggest we 
spend $300,000 on the Sylantro solution.

Doug.

-Original Message-
From: Philipp von Klitzing
[mailto:[EMAIL PROTECTED]
Sent: Sunday, December 11, 2005 10:13 AM
To: Douglas Garstang; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] SRV Lookups *ARRGH!*


Dear Douglas!

 Asterisk is really pissing me off.
 Can someone tell me why this doesn't cause SRV lookups to be done on
 outbound calls: 

In general: If you are missing documentation then you are warmly invited 
to write and enhance the existing one (e.g. the Wiki) wherever you see 
fit. In particular you might want to edit this and add what you (and me) 
have learned:

   http://www.voip-info.org/wiki/view/Asterisk+SIP+srvlookup

 So... if I put the domain directly into the dial command an SRV lookup
 is done. If I reference it in sip.conf, an SRV lookup *IS NOT* done.
 WTF??? 

I did some work for you and searched bugs.digium.com for srvlookup. And 
look at what I found:

  http://bugs.digium.com/view.php?id=1805
  http://bugs.digium.com/view.php?id=2081

Reading the bug notes you'll find that this is known - and probably even 
intended - behaviour. If you dislike it: File a bug report, write a 
patch, or find someone that's going to write it for you. Asterisk is an 
open source project, remember?

Cheers, Philipp


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Re: [Asterisk-Users] outgoing calls that last an unreasonably long time

2005-12-11 Thread Warren Burstein


Simone Cittadini wrote:


Warren Burstein ha scritto:




What is frustrating is that the cdr file shows the dst as T rather 
than as the phone number dialed.  I realize that AbsoluteTimout 
causes it to jump to the T extension, but it would help to know who 
the user dialed (asking a week later isn't going to get any useful 
information out of the user).  It's not in the log file, either - 
would increasing the log level help here?



I don't know how this AbsolutTimeout works, anyway I put all the info 
I need in variables before the actual Dial, then in the h extension I 
call SetUserField() (or whatever is called), helps me keeping track of 
reasons for non-terminated calls ...


I am not using the userfield for anything so that sounds like a good 
idea.  It's SetCDRUserField by the way.

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[Asterisk-Users] SRV Lookups

Anyone know when Asterisk is going to properly support DNS SRV Lookups?

We have Asterisk, OpenSER and Polycom phones here. The Polycom phones seem to 
have about the best implementation. They at least try a second system 
(round-robin based on equal weights is flaky tho) if the first doesn't respond 
unlike Asterisk and OpenSER. 

It's kinda hard to build a REDUNDANT VOIP network when more than 2/3 of it 
doesn't support SRV lookups! This also means you have to use IP addresses and 
hostnames a lot of the time, which makes routing a mess. Yuck!

Doug.
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[Asterisk-Users] SIP Qualify

Can someone tell me why qualify=yes is required in sip.conf, before you can 
detect the dialled status of a channel? If I don't have qualify=yes against a 
peer, than checking ${DIALSTATUS} has no effect. Asterisk will just keep trying 
until the dial timeout expires.

Why can't asterisk actually LOOK at the SIP response returned by the SIP proxy 
and do something with it? Why can't it detect a failure to connect (ie proxy 
down) and do something based on that? Do the Asterisk developers realise how 
damn hard this makes it to build any kind of reliability into Asterisk

Doug

-Original Message-
From: Warren Burstein [mailto:[EMAIL PROTECTED]
Sent: Sunday, December 11, 2005 11:17 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] outgoing calls that last an unreasonably
longtime


Simone Cittadini wrote:

 Warren Burstein ha scritto:



 What is frustrating is that the cdr file shows the dst as T rather 
 than as the phone number dialed.  I realize that AbsoluteTimout 
 causes it to jump to the T extension, but it would help to know who 
 the user dialed (asking a week later isn't going to get any useful 
 information out of the user).  It's not in the log file, either - 
 would increasing the log level help here?


 I don't know how this AbsolutTimeout works, anyway I put all the info 
 I need in variables before the actual Dial, then in the h extension I 
 call SetUserField() (or whatever is called), helps me keeping track of 
 reasons for non-terminated calls ...

I am not using the userfield for anything so that sounds like a good 
idea.  It's SetCDRUserField by the way.
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[Asterisk-Users] Asterisk Dynamic DNS

Hi everyone,

I am running two Asterisk servers on two machines that have dynamic DNS due
to ISP changing IP address daily. Both servers are registered on DynDns.org
and IP update scripts work fine on both machines. However, if one machine
changes IP address, other one (that has trunk pointing to machine that
changed address) starts displaying that trunk host is not reachable. O.k. I
thought, it is DNS propagation problem, but it is NOT! Even one hour after
IP change, machine A still points to old IP address and says that it is not
reachable.
Is there any solution?

Regards,
Branko

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[Asterisk-Users] C++ AGI debuggin

2005-12-11 Thread Danish Samad

Hi,

I am looking for ways to debug a custom C++ agi. By debugging I
imply inserting breakpoints in my code and stepping through it,
prefereably using tools such as ddd or kdevelop. I have been trying
different things but none seems to work. Looking forward to your
response.

Thanks.
Danish
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Re: [Asterisk-Users] Asterisk Dynamic DNS

2005-12-11 Thread Tom Vile

restart Asterisk as it cache's the IP when you start Asterisk and will
not re-read the new IP until you restart.  Thats why its important to
have a static IP address.

On 12/11/05, Branko Samardzic [EMAIL PROTECTED] wrote:
 Hi everyone,

 I am running two Asterisk servers on two machines that have dynamic DNS due
 to ISP changing IP address daily. Both servers are registered on DynDns.org
 and IP update scripts work fine on both machines. However, if one machine
 changes IP address, other one (that has trunk pointing to machine that
 changed address) starts displaying that trunk host is not reachable. O.k. I
 thought, it is DNS propagation problem, but it is NOT! Even one hour after
 IP change, machine A still points to old IP address and says that it is not
 reachable.
 Is there any solution?

 Regards,
 Branko

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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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Re: [Asterisk-Users] Asterisk Dynamic DNS

2005-12-11 Thread Roger Hill


Hope this helps.

I had a similar problem today, when I changed the machine that my single 
installation of Asterisk runs on, but keeping the same IP address. My 
Sipura ATA took about 10 minutes to pick up the new machine (even though 
the IP address had not changed). I deduced that it needed the 
registration to time out before it did the lookup again, and thus reset 
the ARP cache.


So, try changing the registration time out to something shorter...say 5 
minutes or so, see if that helps.


Roger

Branko Samardzic wrote:


Hi everyone,

I am running two Asterisk servers on two machines that have dynamic DNS due
to ISP changing IP address daily. Both servers are registered on DynDns.org
and IP update scripts work fine on both machines. However, if one machine
changes IP address, other one (that has trunk pointing to machine that
changed address) starts displaying that trunk host is not reachable. O.k. I
thought, it is DNS propagation problem, but it is NOT! Even one hour after
IP change, machine A still points to old IP address and says that it is not
reachable.
Is there any solution?

Regards,
Branko

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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] Asterisk Dynamic DNS

2005-12-11 Thread Schochet, Wes

Sounds like a DNS caching problem.  Can you tell if the machine is actually
going out to look up the address each time, or is it cached locally for some
period of time?

-Original Message-
From: Branko Samardzic [mailto:[EMAIL PROTECTED] 
Sent: Sunday, December 11, 2005 12:51 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Asterisk Dynamic DNS

Hi everyone,

I am running two Asterisk servers on two machines that have dynamic DNS due
to ISP changing IP address daily. Both servers are registered on DynDns.org
and IP update scripts work fine on both machines. However, if one machine
changes IP address, other one (that has trunk pointing to machine that
changed address) starts displaying that trunk host is not reachable. O.k. I
thought, it is DNS propagation problem, but it is NOT! Even one hour after
IP change, machine A still points to old IP address and says that it is not
reachable.
Is there any solution?

Regards,
Branko

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Re: [Asterisk-Users] Asterisk Dynamic DNS

2005-12-11 Thread Sergio Chersovani


Branko Samardzic wrote:


thought, it is DNS propagation problem, but it is NOT! Even one hour after
IP change, machine A still points to old IP address and says that it is not
reachable.
 

I bet it is a DNS cache problem. Probably the machine A uses a cache dns 
and the record is not up to date.
You have to run nslookup from the machine A to understand if the record 
was updated.

Set the /etc/resolv.conf to point to a ISP dns server.

Sergio
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Re: [Asterisk-Users] SRV Lookups

2005-12-11 Thread Leif Madsen

On 12/11/05, Douglas Garstang [EMAIL PROTECTED] wrote:
 Anyone know when Asterisk is going to properly support DNS SRV Lookups?

Well, luckily Asterisk is open source so you have the ability to code
this yourself. If you can't program in C (like myself), then you have
the option of either hiring someone directly. Another option is to
create a bounty and see if anyone else also requires this
functionality and is willing to contribute some money for development.

Leif Madsen
http://www.oreilly.com/catalog/asterisk
http://www.leifmadsen.com
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RE: [Asterisk-Users] Channel 0/1, span 1 got hangup request

2005-12-11 Thread Steve Totaro

What are you doing in between making changes and testing the changes?

Thanks,
Steve

 
 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.
 
 A do this config, but no effects
 
 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?
 
 No effect on settings:
 span=1,0,0,ccs,hdb3
 span=1,1,5,ccs,hdb3
 span=1,2,4,ccs,hdb3
 
 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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[Asterisk-Users] Asterisk Dynamic DNS

It is not problem with outdated DNS cache. It takes approximatelly 2 minutes
for DynDns entry to propagate and my router picks it up. However Asterisk
does reverse name lookup (instead of trying to resolve server.dyndns.org to
IP address and pick up new one, it does reverse lookup on old IP address
which happen to be valid ISP IP adress from some dynamic pool). I can't make
Asterisk to perform direct DNS lookup. I tried dnsmgr.conf to decrease
timeout to 60 sec, but it doesn't seem to have effect without enabling
managed dns. If I enable managed dns then instead of host name I get
UNSPECIFIED in IAX2 peer list and that is end of story.
Any other thoughts?
Regards,
Branko

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[Asterisk-Users] Attack dialing

2005-12-11 Thread Eric Bishop

Anyone have eny elegant dial plan config for attack dialing? Basically
I just want to automatically and continuously dial a busy until it is
answered and then hand it over to a SIP hanset.


Thanks.
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RE: [Asterisk-Users] SRV Lookups

Sounds like your saying that a serious limitation that effectively makes 
Asterisk unusable in a production environment isn't a priority for the 
'official' developers. Awesome...

-Original Message-
From: Leif Madsen [mailto:[EMAIL PROTECTED]
Sent: Sunday, December 11, 2005 12:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SRV Lookups

On 12/11/05, Douglas Garstang [EMAIL PROTECTED] wrote:
 Anyone know when Asterisk is going to properly support DNS SRV Lookups?

Well, luckily Asterisk is open source so you have the ability to code
this yourself. If you can't program in C (like myself), then you have
the option of either hiring someone directly. Another option is to
create a bounty and see if anyone else also requires this
functionality and is willing to contribute some money for development.

Leif Madsen
http://www.oreilly.com/catalog/asterisk
http://www.leifmadsen.com
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RE: [Asterisk-Users] SRV Lookups

I guess we can put that up there with the inability to share a common Realtime 
database between Asterisk servers for SIP peers too... another serious 
limitation.

-Original Message-
From: Douglas Garstang 
Sent: Sunday, December 11, 2005 12:49 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] SRV Lookups

Sounds like your saying that a serious limitation that effectively makes 
Asterisk unusable in a production environment isn't a priority for the 
'official' developers. Awesome...

-Original Message-
From: Leif Madsen [mailto:[EMAIL PROTECTED]
Sent: Sunday, December 11, 2005 12:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SRV Lookups

On 12/11/05, Douglas Garstang [EMAIL PROTECTED] wrote:
 Anyone know when Asterisk is going to properly support DNS SRV Lookups?

Well, luckily Asterisk is open source so you have the ability to code
this yourself. If you can't program in C (like myself), then you have
the option of either hiring someone directly. Another option is to
create a bounty and see if anyone else also requires this
functionality and is willing to contribute some money for development.

Leif Madsen
http://www.oreilly.com/catalog/asterisk
http://www.leifmadsen.com
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Re: [Asterisk-Users] Attack dialing

2005-12-11 Thread Gavin Hamill

On Mon, 2005-12-12 at 06:39 +1100, Eric Bishop wrote:
 Anyone have eny elegant dial plan config for attack dialing? Basically
 I just want to automatically and continuously dial a busy until it is
 answered and then hand it over to a SIP hanset.
 

Hi Eric :)

The terminology I've always understood for this feature is called 'Camp
on Hold' ... and google says this about 'asterisk camp on hold'

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

Good luck! :)

gdh


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RE: [Asterisk-Users] Re: Teliax experiences

2005-12-11 Thread gw




I also have had good experiences with Teliax. Also 
the CIDName beta function is way cool...

They also offer a pretty plans.

Greg


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Asterisk 
Users - DovidSent: Saturday, December 10, 2005 2:54 PMTo: 
[EMAIL PROTECTED]Cc: 
asterisk-users@lists.digium.comSubject: [Asterisk-Users] Re: Teliax 
experiences

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] SRV Lookups ARRGH!

2005-12-11 Thread Matt Riddell

Douglas Garstang wrote:

 At this point I'm almost ready to throw Asterisk out the window and suggest 
 we spend $300,000 on the Sylantro solution.

Um, if you have $300,000 to spend, you could instead put up a $10,000 bounty
on the feature you need and I can all but guarantee you will have lots of
developers knocking on your door...

-- 
Cheers,

Matt Riddell
___

http://www.sineapps.com/news.php (Daily Asterisk News - html)
http://freevoip.gedameurope.com (Free Asterisk Voip Community)
http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)

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Re: [Asterisk-Users] Attack dialing

2005-12-11 Thread Michael Welter

I think attack dialing means to dial all 10,000 number in an exchange, 
looking for modems and fax machines.  BTW, Colorado Springs, Colorado 
has made it illegal to dial a number without intending to have a 
conversation sigh  Probably something to do with NORAD or Space Command.


Eric Bishop wrote:
Anyone have eny elegant dial plan config for attack dialing? Basically I 
just want to automatically and continuously dial a busy until it is 
answered and then hand it over to a SIP hanset.



Thanks.




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--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
[EMAIL PROTECTED]
www.TelecomMatters.net
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Re: [Asterisk-Users] SRV Lookups


Douglas Garstang wrote:

I guess we can put that up there with the inability to share a common Realtime 
database between Asterisk servers for SIP peers too... another serious 
limitation.


Would you like your money back? Please tell us where to send it and 
we'll get it right over to you.


Whining about stuff that is not implemented (when it's clearly 
documented to not be implemented) does not do anyone any good, and it 
makes the rest of the community tend to ignore the remainder of whatever 
you have to say.


This is a volunteer-driven open source project; people write and test 
what they feel like writing and testing. If you want something that is 
not implemented, you can 'influence' what someone feels like writing and 
testing in whatever way is suitable... but whining at them usually has 
the opposite effect.

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Re: [Asterisk-Users] SRV Lookups

2005-12-11 Thread Paul Tinsley

I fail to understand your combative attitude towards an open source 
project...


Did you pay to download it?
Are you having to pay to use it?

If you really want to talk about throwing money around to fix your 
problems the bounty idea has already been thrown at you, but also keep 
in mind that Digium has a custom development service that you could pay 
for... I'll even give you the link for free!

http://www.digium.com/index.php?menu=service_categorycategory=development

Douglas Garstang wrote:


I guess we can put that up there with the inability to share a common Realtime 
database between Asterisk servers for SIP peers too... another serious 
limitation.

-Original Message-
From: Douglas Garstang 
Sent: Sunday, December 11, 2005 12:49 PM

To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] SRV Lookups


Sounds like your saying that a serious limitation that effectively makes 
Asterisk unusable in a production environment isn't a priority for the 
'official' developers. Awesome...

-Original Message-
From: Leif Madsen [mailto:[EMAIL PROTECTED]
Sent: Sunday, December 11, 2005 12:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SRV Lookups


On 12/11/05, Douglas Garstang [EMAIL PROTECTED] wrote:
 


Anyone know when Asterisk is going to properly support DNS SRV Lookups?
   



Well, luckily Asterisk is open source so you have the ability to code
this yourself. If you can't program in C (like myself), then you have
the option of either hiring someone directly. Another option is to
create a bounty and see if anyone else also requires this
functionality and is willing to contribute some money for development.

Leif Madsen
http://www.oreilly.com/catalog/asterisk
http://www.leifmadsen.com
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RE: [Asterisk-Users] SRV Lookups

What exactly do you mean by 'documented not to be implemented'? If you are 
referring to the fact it isn't implemented, yes I realise that. That's why I'm 
trying to get an idea for when these features will be. This isn't whining. 

If you are however, stating they are designed this way and there's no plan to 
implement them, I'm wondering why there's such resistance to putting redundant 
features into Asterisk?

-Original Message-
From: Kevin P. Fleming [mailto:[EMAIL PROTECTED]
Sent: Sunday, December 11, 2005 1:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SRV Lookups


Douglas Garstang wrote:
 I guess we can put that up there with the inability to share a common 
 Realtime database between Asterisk servers for SIP peers too... another 
 serious limitation.

Would you like your money back? Please tell us where to send it and 
we'll get it right over to you.

Whining about stuff that is not implemented (when it's clearly 
documented to not be implemented) does not do anyone any good, and it 
makes the rest of the community tend to ignore the remainder of whatever 
you have to say.

This is a volunteer-driven open source project; people write and test 
what they feel like writing and testing. If you want something that is 
not implemented, you can 'influence' what someone feels like writing and 
testing in whatever way is suitable... but whining at them usually has 
the opposite effect.
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Re: [Asterisk-Users] Attack dialing

2005-12-11 Thread Gary Reuter

On 12/11/05, Michael Welter [EMAIL PROTECTED] wrote:
I think attack dialing means to dial all 10,000 number in an exchange,looking for modems and fax machines.BTW, Colorado Springs, Coloradohas made it illegal to dial a number without intending to have a
conversation sighProbably something to do with NORAD or Space Command.

It's actually called 'war-dialing'. There were loads
of programs to do it using a modem back in the day -- they'd even
randomize and track the dialed numbers.

But Eric just seems to want an auto-redial-on-busy/congested -- which should be pretty simple...
Assuming you have Zap PRI as group 1 and the specific number you want to redial is 5551212:

exten = 5551212,1,Dial(Zap/g1/${EXTEN}
exten = 5551212,2,GotoIf($[x${DiALSTATUS} = xANSWER]?10)
exten = 5551212,3,Wait(3)
exten = 5551212,4,Goto(1)
exten = 5551212,10,NoOp(call was answered so do something with it)


If anyone knows a way to detect if a remote number becomes 'un-busy'
without actually dialing the number, it would make for an even more
elegant solution.


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Re: [Asterisk-Users] SRV Lookups

2005-12-11 Thread Aaron Daniel

HUH?  I better turn my servers off, they've been doing this for  
months now 0.o



On Dec 11, 2005, at 1:50 PM, Douglas Garstang wrote:

I guess we can put that up there with the inability to share a  
common Realtime database between Asterisk servers for SIP peers  
too... another serious limitation.


-Original Message-
From: Douglas Garstang
Sent: Sunday, December 11, 2005 12:49 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] SRV Lookups


Sounds like your saying that a serious limitation that effectively  
makes Asterisk unusable in a production environment isn't a  
priority for the 'official' developers. Awesome...


-Original Message-
From: Leif Madsen [mailto:[EMAIL PROTECTED]
Sent: Sunday, December 11, 2005 12:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SRV Lookups


On 12/11/05, Douglas Garstang [EMAIL PROTECTED] wrote:
Anyone know when Asterisk is going to properly support DNS SRV  
Lookups?


Well, luckily Asterisk is open source so you have the ability to code
this yourself. If you can't program in C (like myself), then you have
the option of either hiring someone directly. Another option is to
create a bounty and see if anyone else also requires this
functionality and is willing to contribute some money for development.

Leif Madsen
http://www.oreilly.com/catalog/asterisk
http://www.leifmadsen.com
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Re: [Asterisk-Users] mpg123 on x86_64 (Opteron MP)

2005-12-11 Thread Brad


Tzafrir Cohen wrote:

On Sun, Dec 11, 2005 at 09:56:45AM +1100, Brad wrote:


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



Don't bother. Unless you want to stream music, there are cheaper
alteratives to mpg123. Not to mention that mpg123 0.59r has some known
holes (as mentioned in http://mpg123.de/ ) if you consider streaming.


I am trying to install AMP for Asterisk, which requires mpg123. 
Apparently, they say, mpg321 does not work with their setup.




However, if you do want it compiled, why don't you file a bug report
against either http://bugs.debian.org/mpg123 or anything more
ubuntu-specific?


But if I am compiling from source, wouldn't this be a bug for the 
developer rather than the OS maintainers? Although i believe mpg123 is 
no longer maintained?


Brad




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Re: [Asterisk-Users] Change time when * is running

2005-12-11 Thread Julian Lyndon-Smith


I have done that, all is working fine now.

Many thanks for the help guys.

Julian.

Alejandro Vargas wrote:

2005/12/9, Julian Lyndon-Smith [EMAIL PROTECTED]:

Can I change the time when * is running ? I don't want to try just in
case it causes * some grief.


Set up an ntp client and let it work a few hours. It will adjust the
time by small junps avoiding problems of backward clock and will keep
it ok.

--
Alejandro Vargas
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[Asterisk-Users] Regexten




Before 
I play around with this again in 1.2.1, regexten is still essentially broken, 
correct?

The 
misconception seems to be that it allows you to execute a command upon 
registration from a SIP UA. Even the O'Reilly TFOT book erroneously states that 
this is what it is for. After reading the developer discussion though, it 
definitely seems to be broken. Is it fixed yet?

Doug.



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RE: [Asterisk-Users] SRV Lookups

2005-12-11 Thread Kerry Garrison

Dammit, I better go pull my servers out of all of my client's locations
because their production servers have just been rendered unusable. Time to
take their old Toshiba system out of mothballs an.wait a sec, it didn't
do it either, what now? I guess IP Telephony has just died today. Sad, and
it had so much promise.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Aaron Daniel
Sent: Sunday, December 11, 2005 12:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SRV Lookups

HUH?  I better turn my servers off, they've been doing this for months now
0.o


On Dec 11, 2005, at 1:50 PM, Douglas Garstang wrote:

 I guess we can put that up there with the inability to share a common 
 Realtime database between Asterisk servers for SIP peers too... 
 another serious limitation.

 -Original Message-
 From: Douglas Garstang
 Sent: Sunday, December 11, 2005 12:49 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: RE: [Asterisk-Users] SRV Lookups


 Sounds like your saying that a serious limitation that effectively 
 makes Asterisk unusable in a production environment isn't a priority 
 for the 'official' developers. Awesome...

 -Original Message-
 From: Leif Madsen [mailto:[EMAIL PROTECTED]
 Sent: Sunday, December 11, 2005 12:18 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] SRV Lookups


 On 12/11/05, Douglas Garstang [EMAIL PROTECTED] wrote:
 Anyone know when Asterisk is going to properly support DNS SRV 
 Lookups?

 Well, luckily Asterisk is open source so you have the ability to code 
 this yourself. If you can't program in C (like myself), then you have 
 the option of either hiring someone directly. Another option is to 
 create a bounty and see if anyone else also requires this 
 functionality and is willing to contribute some money for development.

 Leif Madsen
 http://www.oreilly.com/catalog/asterisk
 http://www.leifmadsen.com
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RE: [Asterisk-Users] SRV Lookups

I can't find the post right now, but if I do I will email it.

I distinctly remember reading a thread in the archives of this list where 
someone stated that this was ability was not implented yet, and there where 
several challanges in doing so.

Good luck!

-Original Message-
From: Aaron Daniel [mailto:[EMAIL PROTECTED]
Sent: Sunday, December 11, 2005 1:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SRV Lookups


HUH?  I better turn my servers off, they've been doing this for  
months now 0.o


On Dec 11, 2005, at 1:50 PM, Douglas Garstang wrote:

 I guess we can put that up there with the inability to share a  
 common Realtime database between Asterisk servers for SIP peers  
 too... another serious limitation.

 -Original Message-
 From: Douglas Garstang
 Sent: Sunday, December 11, 2005 12:49 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: RE: [Asterisk-Users] SRV Lookups


 Sounds like your saying that a serious limitation that effectively  
 makes Asterisk unusable in a production environment isn't a  
 priority for the 'official' developers. Awesome...

 -Original Message-
 From: Leif Madsen [mailto:[EMAIL PROTECTED]
 Sent: Sunday, December 11, 2005 12:18 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] SRV Lookups


 On 12/11/05, Douglas Garstang [EMAIL PROTECTED] wrote:
 Anyone know when Asterisk is going to properly support DNS SRV  
 Lookups?

 Well, luckily Asterisk is open source so you have the ability to code
 this yourself. If you can't program in C (like myself), then you have
 the option of either hiring someone directly. Another option is to
 create a bounty and see if anyone else also requires this
 functionality and is willing to contribute some money for development.

 Leif Madsen
 http://www.oreilly.com/catalog/asterisk
 http://www.leifmadsen.com
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RE: [Asterisk-Users] Regexten

2005-12-11 Thread Joshua Colp









It doesnt execute a
command upon registration. What it does is insert a simple noop into the
extension and context at priority 1 so the extension then becomes active. For
example:



Before I register:

Exten = 145,2,Dial(SIP/jcolp_cisco1)



When I register it then turns into:

Exten = 145,1,Noop()

Exten = 145,2,Dial(SIP/jcolp_cisco1)



This means when Im registered, a
person can call me but when Im not  the extension is useless and
calling it does nothing and doesnt attempt to call me.



Joshua Colp











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas Garstang
Sent: Sunday, December 11, 2005
4:38 PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: [Asterisk-Users] Regexten







Before I play around with this again in
1.2.1, regexten is still essentially broken, correct?











The misconception seems to be that it
allows you to execute a command upon registration from a SIP UA. Even the
O'Reilly TFOT book erroneously states that this is what it is for. After
reading the developer discussion though, it definitely seems to be broken. Is
it fixed yet?











Doug.


























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Re: [Asterisk-Users] RE:IConnecthere dial out problems

2005-12-11 Thread Dennis Gilmore

Once upon a time Wednesday 07 December 2005 8:42 pm, John Voss wrote:
 Your SIP.conf file looks much different than mine. I'll give it a try.

Hope mine helped

 [iconnect]
 type=friend
 secret=pass
 username=userid
 host=213.137.73.140 ;sipauth.deltathree.com
 permit=213.137.73.140/255.255.255.0
 permit=208.170.168.0/255.255.255.0
 disallow=all
 context=incoming
 allow=gsm
 allow=ulaw
 allow=alaw
 allow=G726
 insecure=very
 nat=Yes
 canreinvite=no

 I don't know what your register line looks like in your SIP.conf. This is
 mine.

 register = ph number:pass:userid@213.137.73.140:5060

 I was unable to receive calls until I added the insecure=very line.
mine is register = ph number:pass:userid@natrelay.deltathree.com

i can receive incomming calls for a little while after a reload  but after 
some timeouts incomming calls stop
-- 
Dennis Gilmore,  RHCE  
dennis AT ausil DOT us http://www.ausil.us


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RE: [Asterisk-Users] SRV Lookups

Yes, you better

http://lists.digium.com/pipermail/asterisk-users/2005-October/129384.html

Kevin P. Fleming kpfleming at digium.com 
Fri Oct 14 01:25:20 CDT 2005 

Marco Balmer wrote:

 Any ideas or hints?

Yes. Whatever documentation told you that you could share a Realtime SIP 
peer database between two Asterisk servers was in error (or at least 
very incomplete).

There are ways to do it right now, but it's not trivial and does not 
provide all the functionality that someone would want from such an 
arrangement.

There's no need to be nasty either.

-Original Message-
From: Kerry Garrison [mailto:[EMAIL PROTECTED]
Sent: Sunday, December 11, 2005 1:42 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] SRV Lookups


Dammit, I better go pull my servers out of all of my client's locations
because their production servers have just been rendered unusable. Time to
take their old Toshiba system out of mothballs an.wait a sec, it didn't
do it either, what now? I guess IP Telephony has just died today. Sad, and
it had so much promise.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Aaron Daniel
Sent: Sunday, December 11, 2005 12:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SRV Lookups

HUH?  I better turn my servers off, they've been doing this for months now
0.o


On Dec 11, 2005, at 1:50 PM, Douglas Garstang wrote:

 I guess we can put that up there with the inability to share a common 
 Realtime database between Asterisk servers for SIP peers too... 
 another serious limitation.

 -Original Message-
 From: Douglas Garstang
 Sent: Sunday, December 11, 2005 12:49 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: RE: [Asterisk-Users] SRV Lookups


 Sounds like your saying that a serious limitation that effectively 
 makes Asterisk unusable in a production environment isn't a priority 
 for the 'official' developers. Awesome...

 -Original Message-
 From: Leif Madsen [mailto:[EMAIL PROTECTED]
 Sent: Sunday, December 11, 2005 12:18 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] SRV Lookups


 On 12/11/05, Douglas Garstang [EMAIL PROTECTED] wrote:
 Anyone know when Asterisk is going to properly support DNS SRV 
 Lookups?

 Well, luckily Asterisk is open source so you have the ability to code 
 this yourself. If you can't program in C (like myself), then you have 
 the option of either hiring someone directly. Another option is to 
 create a bounty and see if anyone else also requires this 
 functionality and is willing to contribute some money for development.

 Leif Madsen
 http://www.oreilly.com/catalog/asterisk
 http://www.leifmadsen.com
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RE: [Asterisk-Users] Regexten

Ah ok.
Thanks. I was hoping to use it to 'replicate' registrations from one Asterisk
system to another.
Darn
it.

-Original Message-From: Joshua Colp
[mailto:[EMAIL PROTECTED]Sent: Sunday, December 11, 2005 1:47
PMTo: 'Asterisk Users Mailing List - Non-Commercial
Discussion'Subject: RE: [Asterisk-Users]
Regexten

It doesnt execute
a command upon registration. What it does is insert a simple noop into the
extension and context at priority 1 so the extension then becomes active. For
example:

Before I
register:
Exten =
145,2,Dial(SIP/jcolp_cisco1)

When I register it
then turns into:
Exten =
145,1,Noop()
Exten =
145,2,Dial(SIP/jcolp_cisco1)

This means when Im
registered, a person can call me but when Im not the extension is useless
and calling it does nothing and doesnt attempt to call
me.

Joshua
Colp

From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas GarstangSent: Sunday, December 11, 2005 4:38
PMTo: Asterisk Users Mailing
List - Non-Commercial DiscussionSubject: [Asterisk-Users]
Regexten

Before I play around
with this again in 1.2.1, regexten is still essentially broken,
correct?

The misconception
seems to be that it allows you to execute a command upon registration from a
SIP UA. Even the O'Reilly TFOT book erroneously states that this is what it is
for. After reading the developer discussion though, it definitely seems to be
broken. Is it fixed yet?

Doug.

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Re: [Asterisk-Users] Attack dialing

2005-12-11 Thread C F

Have you looked at retrydial?

On 12/11/05, Eric Bishop [EMAIL PROTECTED] wrote:
 Anyone have eny elegant dial plan config for attack dialing? Basically I
 just want to automatically and continuously dial a busy until it is answered
 and then hand it over to a SIP hanset.


  Thanks.

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Re: [Asterisk-Users] Asterisk Dynamic DNS

2005-12-11 Thread Wilson Pickett

 It is not problem with outdated DNS cache. It takes approximatelly 2 minutes
 Any other thoughts?

I guess you haven't read the earlier messages saying that asterisk has
chached the ip and needs to be restarted if that ip changes.
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Re: [Asterisk-Users] SRV Lookups

Douglas Garstang wrote:
What exactly do you mean by 'documented not to be implemented'? If you are referring to the fact it isn't implemented, yes I realise that. That's why I'm trying to get an idea for when these features will be. This isn't whining.

You did not ask when they would be implemented, you said that it not
being implemented was 'a major limitation'. That is not asking a
question, nor is it in any way constructive.

If you are however, stating they are designed this way and there's no plan to
implement them, I'm wondering why there's such resistance to putting redundant
features into Asterisk?

You are very good at putting words into others' mouths, apparently...
Since nobody has said anything of the kind, making a statement like this
is purely inflammatory. Where is this resistance that you speak of? Do
you have any evidence that someone provided a functional implementation
of this feature and it was rejected? Do you have any evidence that
someone provided even a functional design for others to implement and it
was rejected? If not, saying there is 'resistance' is purely
argumentative and only annoys everyone else.

Asterisk has the features it has because people with the skills to
implement them did so; features that are not present are not that way
because someone decided they would not ever be there (except in very
rare circumstances), they are that way because nobody has provided an
implementation that was merged into the source tree. Reading anything
more into the lack of a feature is wasting our time and yours :-)

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Re: [Asterisk-Users] Attack dialing

2005-12-11 Thread pdhales

In Australia, dialling a number more than 3 times in a row without waiting a
certain amount of time is not allowed to be programmed into a system.

Not against the law, but againt telecom reg's.

PaulH

- Original Message - 
From: Michael Welter [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, December 12, 2005 7:00 AM
Subject: Re: [Asterisk-Users] Attack dialing

 I think attack dialing means to dial all 10,000 number in an exchange,
 looking for modems and fax machines.  BTW, Colorado Springs, Colorado
 has made it illegal to dial a number without intending to have a
 conversation sigh  Probably something to do with NORAD or Space Command.

 Eric Bishop wrote:
  Anyone have eny elegant dial plan config for attack dialing? Basically I
  just want to automatically and continuously dial a busy until it is
  answered and then hand it over to a SIP hanset.

  Thanks.

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 -- 
 Michael Welter
 Telecom Matters Corp.
 Denver, Colorado US
 +1.303.414.4980
 [EMAIL PROTECTED]
 www.TelecomMatters.net
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[Asterisk-Users] Asterisk Dynamic DNS

That's fine. But there is, obviously, situations where such situation is not
welcome. Is it possible to force Asterisk to refresh cache every in a while.
Regards,
Branko

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Re: [Asterisk-Users] SRV Lookups

2005-12-11 Thread Eric \ManxPower\ Wieling


Douglas Garstang wrote:
What exactly do you mean by 'documented not to be implemented'? If you are referring to the fact it isn't implemented, yes I realise that. That's why I'm trying to get an idea for when these features will be. This isn't whining. 


If you are however, stating they are designed this way and there's no plan to 
implement them, I'm wondering why there's such resistance to putting redundant 
features into Asterisk?


There isn't.  However, so far nobody with the skills has cared enough to 
write that feature.

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RE: [Asterisk-Users] SRV Lookups ARRGH!

2005-12-11 Thread Philipp von Klitzing

Hey Douglas,

 The link to the Wiki is woefully indadequate. I have no problem adding
 to the documentation, as soon as I bloody understand it myself. 

Excellent, that's what I wanted to hear! :-)

  http://bugs.digium.com/view.php?id=1805
  http://bugs.digium.com/view.php?id=2081

 The two bug links you provided appear to be almost completely
 unrelated to what I asked about except they touch on the subject of
 SRV lookups.

Not really - if you read carefully enough you will find these two quotes 
in the two bug reports - but with almost all your energy spent with being 
upset you probably missed them... ;-

markster: ...Fortunately srv lookups are bypassed by peer 
declarations...

markster: If you have a sip friend/peer entry for budgetphone.nl it will 
not lookup the SRV record but will use the host you have specified. If 
you do not have srvlookup=yes in your sip.conf in the general section, 
SRV records will never be searched.

 Oh, and you know what, why is it assumed that to use open source
 software I have to be a seasoned C programmer who can contribute to
 the code? Where is that requirement stipulated? 

Nowhere - you just invented that. Why should you need to know C in order 
to put a bounty, or just formulate well documented bug report? Or where 
does the Wiki require C knowledge (talking about contributing, not 
using - you don't need C skills to at all to use asterisk).

Asterisk is not perfect, there is a lot of work in progress (sometimes 
too much), but it is the only one of its kind, it works, and it gets 
better day-by-day. And if you find a way to help with that (and 
preferably a way that doesn't step on people's collective toes): great!

Cheers, Philipp


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

2005-12-11 Thread snacktime


 I am trying to install AMP for Asterisk, which requires mpg123.
 Apparently, they say, mpg321 does not work with their setup.

I would think that you could just edit musiconhold.conf after AMP is
installed, and have it use something else like madplay.  Madplay has
worked very well for me so far.

Chris
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[Asterisk-Users] Realtime Subscribecontext

I don't see the sip.conf subscribecontext directive specified (on a per user 
basis) for use with Realtime. Does realtime allow it? What's the field called?

Thanks,
Doug
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[Asterisk-Users] Small / embedded system recommendations

2005-12-11 Thread Alistair Cunningham


I know it's been asked before, but this area moves rapidly.

Would anyone have recommendations for a small or embedded system 
suitable for running Asterisk on? Ideally, we'd like two boxes:


- One using compact flash, and is fanless, with rapid booting.

- One with a hard disk for voicemail, call recording, etc.

Preferably they would be capable of bridging 60 calls Zap-Zap or 
Zap-SIP, but we're willing to consider less powerful systems. The 
ability to take a single Digium card is desirable.


Being able to run MySQL, Apache, and SER as well are essential, as it's 
for our upcoming ITSP in an office product, which uses these heavily.


Speaking of which, this is moving forward. It will have all the end 
customer features of ITSP in a box 
(http://integrics.com/products/itsp/), but not billing, resellers, 
affiliates, calling cards, etc. We'll be looking for resellers, probably 
second quarter next year. If you're interested, feel free to email us, 
but we don't have much information yet, so will respond with a canned 
email for now. More information to follow.


--
Alistair Cunningham,
Integrics Ltd,
+44 (0)7870 699 479
http://integrics.com/
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Re: [Asterisk-Users] Asterisk Limitations

2005-12-11 Thread Eric \ManxPower\ Wieling


Philipp von Klitzing wrote:
Asterisk is not perfect, there is a lot of work in progress (sometimes 
too much), but it is the only one of its kind, it works, and it gets 
better day-by-day. And if you find a way to help with that (and 
preferably a way that doesn't step on people's collective toes): great!


Asterisk has several significant limitations.

SRV lookups only try the first host returned
SRV lookups are bypassed for peer/friend entries
No RTP Jitterbuffer
Outgoing RTP depends on incoming RTP
No VAD/CNG support
Using hostnames in config files is pretty useless
No useful CDR info for IAX2 transfers

I'm sure there are MANY more.

Many of these issues might be resolved in 1.4.  Many of these issues 
already have patches in the bug tracker to add that feature, etc.


You can do one of three things.  1) You can fight the limitations.  This 
is a losing battle.  2) You can accept these limitations and work around 
them.3) You can fix the limitations or convince someone else to fix 
these limitations.  Money works well for this.

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Re: [Asterisk-Users] Realtime Subscribecontext


Douglas Garstang wrote:

I don't see the sip.conf subscribecontext directive specified (on a per user 
basis) for use with Realtime. Does realtime allow it? What's the field called?


Any option that can be specified for a user/peer/friend in sip.conf can 
be specified in Realtime using the same option name as the column name.

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Re: [Asterisk-Users] Attack dialing

2005-12-11 Thread Rich Adamson


 In Australia, dialling a number more than 3 times in a row without waiting a
 certain amount of time is not allowed to be programmed into a system.
 
 Not against the law, but againt telecom reg's.
 
  I think attack dialing means to dial all 10,000 number in an exchange,
  looking for modems and fax machines.  BTW, Colorado Springs, Colorado
  has made it illegal to dial a number without intending to have a
  conversation sigh  Probably something to do with NORAD or Space Command.
 
   Anyone have eny elegant dial plan config for attack dialing? Basically I
   just want to automatically and continuously dial a busy until it is
   answered and then hand it over to a SIP hanset.

The OP just happened to use incorrect words for a pbx feature that
is rather well known as camp-on. The majority (if not all manufacturers) 
do not allow the camp-on feature to be used with outside numbers; only pbx
extensions.

The retrydial that someone suggested is sort of a first step, but 
doesn't come close to what commercial pbx manufacturers call camp-on.

In the commercial systems that I'm familiar with, if you call an extension
that is busy, you press a predefined key for the system and hang up. The
pbx then silently monitors for that extension to become available, and
when it does, calls both the caller and the callee back, bridging the
two together automatically. (Some systems use a destinctive ring for
such camp-on return calls, and include the logic to test both exensions
to be sure both are available before completing the camp-on return call.)

I'm not a programmer and couldn't code this up if I wanted to, but it is
a nice feature and all the basic components needed to develop the feature
already exist in * code. I suppose some could code an agi script to do it,
but it seems to me such a feature should be included in * code instead.


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RE: [Asterisk-Users] Realtime Subscribecontext

Thanks while we're on the topic of realtime. Can realtime sipusers be 
shared amongst multiple Asterisk boxes, to share a common location database? 
I'm sitting here on a Sunday jerking around with it, having problems. I'd like 
to know before I spend more Sundays doing the same thing if it's even supposed 
to work or not.

Thanks.

-Original Message-
From: Kevin P. Fleming [mailto:[EMAIL PROTECTED]
Sent: Sunday, December 11, 2005 3:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Realtime Subscribecontext


Douglas Garstang wrote:
 I don't see the sip.conf subscribecontext directive specified (on a per user 
 basis) for use with Realtime. Does realtime allow it? What's the field called?

Any option that can be specified for a user/peer/friend in sip.conf can 
be specified in Realtime using the same option name as the column name.
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Re: [Asterisk-Users] Realtime Subscribecontext


Douglas Garstang wrote:

Thanks while we're on the topic of realtime. Can realtime sipusers be 
shared amongst multiple Asterisk boxes, to share a common location database? 
I'm sitting here on a Sunday jerking around with it, having problems. I'd like 
to know before I spend more Sundays doing the same thing if it's even supposed 
to work or not.


Uhhh... you already quoted my previous message on that topic stating 
that it was not supported at this time. In any given situation, it may 
or may not work properly, depending on exactly what the servers and 
clients are doing.


Even if the code had been written, there will still be many issues 
involved in actually implementing it, including (but not limited to) NAT 
traversal, call limit handling, registration expiration and others. It 
also mandates that there can be _no_ caching of peer/user information in 
memory, which currently means there is no 'qualify' or MWI notification 
possible.

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Re: [Asterisk-Users] Attack dialing

2005-12-11 Thread trixter aka Bret McDanel

On Sun, 2005-12-11 at 16:25 -0600, Rich Adamson wrote:
Anyone have eny elegant dial plan config for attack dialing? Basically I
just want to automatically and continuously dial a busy until it is
answered and then hand it over to a SIP hanset.
 
 The OP just happened to use incorrect words for a pbx feature that
 is rather well known as camp-on. The majority (if not all manufacturers) 
 do not allow the camp-on feature to be used with outside numbers; only pbx
 extensions.
 
there is an asterisk agi (may not be an agi but is a program, I think it
was an agi) to do this for radio stations, perhaps a google for that, I
dont remember the exact name used but do remember that someone was
speaking about mass dialing to a radio contest line and bridging to
their phone once it was connected.  I am sure that it can be modified to
do just one chanenl if that is desired.  If it doesnt exist a timeout
could most likely be easily added so that it doesnt continue to dial
after some period has elapsed.  For radio contests you most likely dont
want it to dial all day as the call in parts are short lived.

 In the commercial systems that I'm familiar with, if you call an extension
 that is busy, you press a predefined key for the system and hang up. The
 pbx then silently monitors for that extension to become available, and
 when it does, calls both the caller and the callee back, bridging the
 two together automatically. (Some systems use a destinctive ring for
 such camp-on return calls, and include the logic to test both exensions
 to be sure both are available before completing the camp-on return call.)
 
That works internally but not outside the pbx unless you do infact
redial the number over and over.  Unless that was included in your
definition of 'monitor' which was not clear.


-- 
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
http://www.sacaug.org/ Sacramento Asterisk Users Group


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Re: [Asterisk-Users] Attack dialing

2005-12-11 Thread Alistair Cunningham


Rich Adamson wrote:


I'm not a programmer and couldn't code this up if I wanted to, but it is
a nice feature and all the basic components needed to develop the feature
already exist in * code. I suppose some could code an agi script to do it,
but it seems to me such a feature should be included in * code instead.


It's not too hard on Asterisk; we've done it in the past, including for 
external numbers. The hardest part is making it reliable with external 
numbers, as it has the bad habit of connecting you to voicemail, PBXs 
which answer then play ringing as they try to connect you to the 
extension, etc. For this reason, we generally recommend using it on 
internal numbers only were we know exactly how the call is going to be 
routed.


Alistair Cunningham,
Integrics Ltd,
+44 (0)7870 699 479
http://integrics.com/
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RE: [Asterisk-Users] Asterisk Limitations

2005-12-11 Thread James Sturges

Or.

You can go back to a Traditional PBX and really experience the meaning of
the phrase  { significant limitations. }

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
ManxPower Wieling
Sent: Monday, 12 December 2005 8:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk Limitations

Philipp von Klitzing wrote:
 Asterisk is not perfect, there is a lot of work in progress (sometimes 
 too much), but it is the only one of its kind, it works, and it gets 
 better day-by-day. And if you find a way to help with that (and 
 preferably a way that doesn't step on people's collective toes): great!

Asterisk has several significant limitations.

SRV lookups only try the first host returned
SRV lookups are bypassed for peer/friend entries
No RTP Jitterbuffer
Outgoing RTP depends on incoming RTP
No VAD/CNG support
Using hostnames in config files is pretty useless
No useful CDR info for IAX2 transfers

I'm sure there are MANY more.

Many of these issues might be resolved in 1.4.  Many of these issues 
already have patches in the bug tracker to add that feature, etc.

You can do one of three things.  1) You can fight the limitations.  This 
is a losing battle.  2) You can accept these limitations and work around 
them.3) You can fix the limitations or convince someone else to fix 
these limitations.  Money works well for this.
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RE: [Asterisk-Users] Asterisk Limitations

2005-12-11 Thread trixter aka Bret McDanel

On Mon, 2005-12-12 at 09:00 +1000, James Sturges wrote:
 Or.
 
 You can go back to a Traditional PBX and really experience the meaning of
 the phrase  { significant limitations. }
 

That is a really bad excuse for limitations however, and actually does
more harm than good.  While it may be true that asterisk has fewer
limitations than another product to say that your option is to use
asterisk or something else more limiting doesnt get any of the problems
fixed.  At least the person you replied to gave constructive answers to
remove some of the limitations, such as looking at CVS/bugtracker for
patches or paying money to get someone motivitated to fix them. 

-- 
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
http://www.sacaug.org/ Sacramento Asterisk Users Group


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Re: [Asterisk-Users] Regexten

2005-12-11 Thread Leif Madsen

On 12/11/05, Douglas Garstang [EMAIL PROTECTED] wrote:
 The misconception seems to be that it allows you to execute a command upon
 registration from a SIP UA. Even the O'Reilly TFOT book erroneously states
 that this is what it is for. After reading the developer discussion though,
 it definitely seems to be broken. Is it fixed yet?

And I quote from page 227, Asterisk will dynamically create and
destroy a NoOp at priority 1 for the extension. All actions to be
performed upon registration should start at priority 2.

Leif Madsen.
http://www.oreilly.com/catalog/asterisk
http://www.leifmadsen.com
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Re: [Asterisk-Users] Asterisk Dynamic DNS

2005-12-11 Thread John Biundo

If you're experiencing the same issue I am, there's a less painful 
solution than restarting asterisk.


Asterisk resolves the externip (and, I think, externhost) parameters in 
sip.conf at startup.  If the values are a domain name registered with 
dyndns.org, and the IP that these domain names point at changes, then 
you have a problem.


It turns out that sip reload will cause it to resolve externip (for 
sure, and I assume externhost as well) again.  So I run a cron job to do 
a sip reload periodically.


(Ideally, I want to trigger this job from my dyndns ip-change script, 
but I haven't gotten around to that yet.  Doing it every 5 minutes is a 
bit of an ugly hack, but it works for me.)


HTH.
john

Branko Samardzic wrote:

That's fine. But there is, obviously, situations where such situation is not
welcome. Is it possible to force Asterisk to refresh cache every in a while.
Regards,
Branko

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Re: [Asterisk-Users] Reboot stops TD400P cards from outgoing calls until first incoming call arrives - resolution

2005-12-11 Thread James B. MacLean





James MacLean wrote:

  
  
Rich Adamson wrote:
  

  Thanks for the heads up. More dissappointing is that the E/F card is 
the newer card purchased. Where can I go to see when certain revisions 
were released? Surprising that the newer card just purchased (to me) 
is the older rev :(.



You can probably search the -cvs list to find it, but that might be a
little time consuming. You should see the card's pic id's in dmesg and
then look in the zaptel src directory for matching entries, or, simply
call digium support.

It sounds like you are running an older version of zaptel/asterisk.

  
  
Thanks again Rich for the info. This is all from latest CVS though. I
have generated an e-mail support ticket with digium, so I am looking
forward to the answer. 
  
No doubt it will be too obvious :).
  
JES

My problem was that my lines were setup as kool start :

fxsks=3-4

But in our area, I should have set them to loopstart :

fxsls=3-4

Thanks to Kenny at Digium for working through it and finding the
problem :).

JES




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[Asterisk-Users] Re: Email to voice?

2005-12-11 Thread Steven

I do not  remember where I saw it, but it was for Asterisk and Nagios.
I believe they had festival and a sip client on the nagios server and would 
just place a sip call and have festival read the alert.
I do not have more info, sorry.

I am using the manager interface to get info from asterisk into my Nagios 
server, but I am only doing notifications via email and modem/TAP.




-- 
-- 
Steven

May you have the peace and freedom that come from abandoning all hope of 
having a better past.
----  ---  - - -   -- -   -   --  - - - --- - --   - 
 - --- - - -- -  -- --   -   --
Chris Mason (Lists) [EMAIL PROTECTED] wrote in message 
news:[EMAIL PROTECTED]

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.

 The Event would be a Nagios alert, and I can write a custom EventHandler 
 to send a message to Asterisk via a script over SSH, or just send an email 
 as it now does.

 -- 
 Chris Mason


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RE: [Asterisk-Users] Small / embedded system recommendations

2005-12-11 Thread Chris Bagnall

 Would anyone have recommendations for a small or embedded 
 system suitable for running Asterisk on? Ideally, we'd like two boxes:
 - One using compact flash, and is fanless, with rapid booting.
 - One with a hard disk for voicemail, call recording, etc.
 Preferably they would be capable of bridging 60 calls Zap-Zap 
 or Zap-SIP, but we're willing to consider less powerful 
 systems. The ability to take a single Digium card is desirable.

We've recently ordered a pair of these:
http://www.soekris.com/net4801.htm

Which have a standard PCI slot into which I'm hoping a TDM card will work.
Their Belgian distributor (kd85.com) appears to have a nice range of
expanded cases that might (hopefully) take a TDM card. I'll find out when
they arrive I guess.

I'm not sure whether a 266Mhz processor would stand a hope in hell of
running 60 calls though - I'll leave that one for someone else to answer.
Fortunately our requirement is only for 4-6 concurrent calls.

Regards,

Chris
-- 
C.M. Bagnall, Director, Minotaur I.T. Limited
This email is made from 100% recycled electrons


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[Asterisk-Users] Cleaning up CLID on incoming PRI lines

2005-12-11 Thread Adrian Carter


Hey guys,
   Does anyone know of a nice way to clean up CLID on all calls coming 
in a certain trunk? One of our PRI providers drops off the 0, which 
means that matching and 'one-click-call-return' are broken because it 
appends 'half' the Area Code to the CLID (In Australia, area codes are 
prefaced 0N).


   At the moment, I have a nasty patch to AMP's 'user-callerid' macro 
that re-writes teh CLIDNum variable to have the correct 12 digits with 
the 0. But because this happens after the initial call pickup, the 
ringing call itself reports the 'bad' CLID.


   Is there a patch or hook I could add to zapata.conf or some way to 
get asterisk to say Add 0 to the front of all CLID on this ZAP channel ?


Thanks all!

Adrian

--
Adrian Carter
Technical Manager
Leading Edge Internet

Web   http://www.lei.net.au http://support.lei.net.au
Direct+61 2 6163 6162  Support 1 300 662 415
E-mail[EMAIL PROTECTED]

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

2005-12-11 Thread Jeffery Chen

do festival support Chinese ?


On 12/12/05, Steven [EMAIL PROTECTED] wrote:
I do notremember where I saw it, but it was for Asterisk and Nagios.I believe they had festival and a sip client on the nagios server and would
just place a sip call and have festival read the alert.I do not have more info, sorry.I am using the manager interface to get info from asterisk into my Nagiosserver, but I am only doing notifications via email and modem/TAP.
StevenMay you have the peace and freedom that come from abandoning all hope ofhaving a better past. - - -- - - --- - - --- - -- -
- --- - - -- --- -- - --Chris Mason (Lists) [EMAIL PROTECTED] wrote in messagenews:[EMAIL PROTECTED]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 dealwith the event. The Event would be a Nagios alert, and I can write a custom EventHandler to send a message to Asterisk via a script over SSH, or just send an email
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http://lists.digium.com/mailman/listinfo/asterisk-users-- Jeffery 
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[Asterisk-Users] Originate action script causes Asterisk to hang

2005-12-11 Thread KRTorio

I have a customized dialer program tested for more than a week already.
I use the manager API Originate action to Originate a call from a
remote extension to an agent logged in the local Asterisk server. The
problem is that the Originate action causes Asterisk to hang (needs
restarting) everytime I run the following script:

in the manager API script:
Action: Originate
Channel: Local/[EMAIL PROTECTED] or Local/[EMAIL PROTECTED]
Application: Dial
Data: Agent/1XXX|30|tm

in extensions.conf:
[test]
exten = 325,1,Dial(IAX2/xxx:[EMAIL PROTECTED]/325,30,t)
exten = 325,2,Busy

exten = 400,1,Dial(IAX2/xxx:[EMAIL PROTECTED]/400,30,t)
exten = 400,2,Busy

exten = _3XX,1,Dial(SIP/${EXTEN},30,t)
exten = _3XX,2,Busy

exten = _4XX,1,Dial(SIP/${EXTEN},30,t)
exten = _4XX,2,Dial(SIP/${EXTEN},30,t)
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Re: [Asterisk-Users] Small / embedded system recommendations

2005-12-11 Thread Kristian Kielhofner


Chris Bagnall wrote:
Would anyone have recommendations for a small or embedded 
system suitable for running Asterisk on? Ideally, we'd like two boxes:

- One using compact flash, and is fanless, with rapid booting.
- One with a hard disk for voicemail, call recording, etc.
Preferably they would be capable of bridging 60 calls Zap-Zap 
or Zap-SIP, but we're willing to consider less powerful 
systems. The ability to take a single Digium card is desirable.



We've recently ordered a pair of these:
http://www.soekris.com/net4801.htm

Which have a standard PCI slot into which I'm hoping a TDM card will work.
Their Belgian distributor (kd85.com) appears to have a nice range of
expanded cases that might (hopefully) take a TDM card. I'll find out when
they arrive I guess.

I'm not sure whether a 266Mhz processor would stand a hope in hell of
running 60 calls though - I'll leave that one for someone else to answer.
Fortunately our requirement is only for 4-6 concurrent calls.

Regards,

Chris


Chris,

About the TDM card...  Several things:

- No FXS ports - the Soekris doesn't have the means to provide ringing 
voltage for the card.


- Even four concurrent calls might be tricky.  Even if you aren't doing 
transcoding, you will still probably have to do echo cancellation, which 
is CPU intensive (especially on the Soekris)!


- The case.  You already know about kd85.com.  But 30 euro is a little 
expensive for their full height PCI case...


	As far as 60 calls.  Maybe SIP to SIP, with re-invites and even that is 
pushing it.  Overall, a Net4801 would not be appropriate for what the OP 
is looking for.


P.S. - You should run AstLinux on your net4801:

http://www.astlinux.org

P.P.S. - I created AstLinux, and it rocks ;)!

--
Kristian Kielhofner
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Re: [Asterisk-Users] Cleaning up CLID on incoming PRI lines

2005-12-11 Thread Kristian Kielhofner


Adrian Carter wrote:

Hey guys,
   Does anyone know of a nice way to clean up CLID on all calls coming 
in a certain trunk? One of our PRI providers drops off the 0, which 
means that matching and 'one-click-call-return' are broken because it 
appends 'half' the Area Code to the CLID (In Australia, area codes are 
prefaced 0N).


   At the moment, I have a nasty patch to AMP's 'user-callerid' macro 
that re-writes teh CLIDNum variable to have the correct 12 digits with 
the 0. But because this happens after the initial call pickup, the 
ringing call itself reports the 'bad' CLID.


   Is there a patch or hook I could add to zapata.conf or some way to 
get asterisk to say Add 0 to the front of all CLID on this ZAP channel ?


Thanks all!

Adrian



Adrian,

I don't know how this will play with AMP, but here is goes:

in zapata.conf, create a new context for calls coming in the PRI in 
question, let's call it pri-cidfix


extensions.conf:

[pri-cidfix]
exten = _X.,1,SetCIDNUM(0${CALLERIDNUM})
exten = _X.,2,Dowhatever

or in 1.2:

[pri-cidfix]
exten = _X.,1,Set(CALLERID(number)=0${CALLERIDNUM})
exten = _X.,2,Dowhatever

	I'm being lazy with the extensions and the second priority, but 
hopefully you get the idea.


--
Kristian Kielhofner
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RE: [Asterisk-Users] Asterisk Dynamic DNS

Here is the solution.

On each and every server that has DynDns you can install Cron job that will
perform forced configuration updates of an running server. This updates will
pick up all the changes in iax.conf and IP address change!
So, put following in your cron job:

asterisk -r -x iax2 reload and you will have DynDns working nicely.
Cheers,
Branko

-Original Message-
From: Branko Samardzic [mailto:[EMAIL PROTECTED]
Sent: Sunday, December 11, 2005 3:25 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Asterisk Dynamic DNS


No problem.
Cheers,
Branko

-Original Message-
From: Manny A. Wise [mailto:[EMAIL PROTECTED]
Sent: Sunday, December 11, 2005 2:41 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Asterisk Dynamic DNS


I was told that the new version(1.2.1) of asterisk support FQDN on
(sip.conf) externip=name.dyndns.org... but we can't get it to work either,
if you find any solution to the problem, can you please let me know?
Thanks
Manny

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Branko
Samardzic
Sent: Sunday, December 11, 2005 1:51 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Asterisk Dynamic DNS
Hi everyone,
I am running two Asterisk servers on two machines that have dynamic DNS due
to ISP changing IP address daily. Both servers are registered on DynDns.org
and IP update scripts work fine on both machines. However, if one machine
changes IP address, other one (that has trunk pointing to machine that
changed address) starts displaying that trunk host is not reachable. O.k. I
thought, it is DNS propagation problem, but it is NOT! Even one hour after
IP change, machine A still points to old IP address and says that it is not
reachable.
Is there any solution?
Regards,
Branko


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RE: [Asterisk-Users] Regexten

No. It doesn't work that way. 

-Original Message- 
From: Leif Madsen [mailto:[EMAIL PROTECTED] 
Sent: Sun 12/11/2005 4:19 PM 
To: Asterisk Users Mailing List - Non-Commercial Discussion 
Cc: 
Subject: Re: [Asterisk-Users] Regexten

On 12/11/05, Douglas Garstang [EMAIL PROTECTED] wrote:
 The misconception seems to be that it allows you to execute a command 
upon
 registration from a SIP UA. Even the O'Reilly TFOT book erroneously 
states
 that this is what it is for. After reading the developer discussion 
though,
 it definitely seems to be broken. Is it fixed yet?

And I quote from page 227, Asterisk will dynamically create and
destroy a NoOp at priority 1 for the extension. All actions to be
performed upon registration should start at priority 2.

Leif Madsen.
http://www.oreilly.com/catalog/asterisk
http://www.leifmadsen.com
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RE: [Asterisk-Users] Realtime Subscribecontext

Oh dear. This is not good news.
 
The issues of NAT, call limit handling and registration expiration don't sound 
quite so bad. I think we can live with those, if we can in fact just get a 
central location database. Do you have any suggestions or ideas about how this 
can be implemented with Asterisk? Because, honestly, right now this current 
limitation is proving to be a real thorn in our side.
 
Doug.

-Original Message- 
From: Kevin P. Fleming [mailto:[EMAIL PROTECTED] 
Sent: Sun 12/11/2005 3:51 PM 
To: Asterisk Users Mailing List - Non-Commercial Discussion 
Cc: 
Subject: Re: [Asterisk-Users] Realtime Subscribecontext



Douglas Garstang wrote:
 Thanks while we're on the topic of realtime. Can realtime 
sipusers be shared amongst multiple Asterisk boxes, to share a common location 
database? I'm sitting here on a Sunday jerking around with it, having problems. 
I'd like to know before I spend more Sundays doing the same thing if it's even 
supposed to work or not.

Uhhh... you already quoted my previous message on that topic stating
that it was not supported at this time. In any given situation, it may
or may not work properly, depending on exactly what the servers and
clients are doing.

Even if the code had been written, there will still be many issues
involved in actually implementing it, including (but not limited to) NAT
traversal, call limit handling, registration expiration and others. It
also mandates that there can be _no_ caching of peer/user information in
memory, which currently means there is no 'qualify' or MWI notification
possible.
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RE: [Asterisk-Users] SRV Lookups

Hi Kerry.
 
Did you see Kevin's subsequent post about using Realtime to share SIP registry 
info?
 
Uhhh... you already quoted my previous message on that topic stating
that it was not supported at this time. In any given situation, it may
or may not work properly, depending on exactly what the servers and
clients are doing.

Even if the code had been written, there will still be many issues
involved in actually implementing it, including (but not limited to) NAT
traversal, call limit handling, registration expiration and others. It
also mandates that there can be _no_ caching of peer/user information in
memory, which currently means there is no 'qualify' or MWI notification
possible.
 
Now, I can be a real jerk and say I told you so, or I could inquire as to just 
how you got it working when it isn't supposed to? This limitation is proving to 
be a real thorn in our side and I would just die to get it to work.
 
Doug.

 
 

-Original Message- 
From: Kerry Garrison [mailto:[EMAIL PROTECTED] 
Sent: Sun 12/11/2005 1:41 PM 
To: 'Asterisk Users Mailing List - Non-Commercial Discussion' 
Cc: 
Subject: RE: [Asterisk-Users] SRV Lookups



Dammit, I better go pull my servers out of all of my client's locations
because their production servers have just been rendered unusable. Time 
to
take their old Toshiba system out of mothballs an.wait a sec, it 
didn't
do it either, what now? I guess IP Telephony has just died today. Sad, 
and
it had so much promise.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Aaron Daniel
Sent: Sunday, December 11, 2005 12:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SRV Lookups

HUH?  I better turn my servers off, they've been doing this for months 
now
0.o


On Dec 11, 2005, at 1:50 PM, Douglas Garstang wrote:

 I guess we can put that up there with the inability to share a common
 Realtime database between Asterisk servers for SIP peers too...
 another serious limitation.

 -Original Message-
 From: Douglas Garstang
 Sent: Sunday, December 11, 2005 12:49 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: RE: [Asterisk-Users] SRV Lookups


 Sounds like your saying that a serious limitation that effectively
 makes Asterisk unusable in a production environment isn't a priority
 for the 'official' developers. Awesome...

 -Original Message-
 From: Leif Madsen [mailto:[EMAIL PROTECTED]
 Sent: Sunday, December 11, 2005 12:18 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] SRV Lookups


 On 12/11/05, Douglas Garstang [EMAIL PROTECTED] wrote:
 Anyone know when Asterisk is going to properly support DNS SRV
 Lookups?

 Well, luckily Asterisk is open source so you have the ability to code
 this yourself. If you can't program in C (like myself), then you have
 the option of either hiring someone directly. Another option is to
 create a bounty and see if anyone else also requires this
 functionality and is willing to contribute some money for development.

 Leif Madsen
 http://www.oreilly.com/catalog/asterisk
 http://www.leifmadsen.com
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To

RE: [Asterisk-Users] Realtime Subscribecontext

Actually, you know, I can't STOP it from caching. I have rtcacheusers=no (or 
whatever it's called. I'm not in front of the system right now) and it's still 
populating astdb with registration entries. Hmmm.

-Original Message- 
From: Kevin P. Fleming [mailto:[EMAIL PROTECTED] 
Sent: Sun 12/11/2005 3:51 PM 
To: Asterisk Users Mailing List - Non-Commercial Discussion 
Cc: 
Subject: Re: [Asterisk-Users] Realtime Subscribecontext

Douglas Garstang wrote:
 Thanks while we're on the topic of realtime. Can realtime 
sipusers be shared amongst multiple Asterisk boxes, to share a common location 
database? I'm sitting here on a Sunday jerking around with it, having problems. 
I'd like to know before I spend more Sundays doing the same thing if it's even 
supposed to work or not.

Uhhh... you already quoted my previous message on that topic stating
that it was not supported at this time. In any given situation, it may
or may not work properly, depending on exactly what the servers and
clients are doing.

Even if the code had been written, there will still be many issues
involved in actually implementing it, including (but not limited to) NAT
traversal, call limit handling, registration expiration and others. It
also mandates that there can be _no_ caching of peer/user information in
memory, which currently means there is no 'qualify' or MWI notification
possible.
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RE: [Asterisk-Users] extensions and regular expressions ( probablyan easy question )

2005-12-11 Thread Rushowr

I just wanted to throw in here that I was intrigued by the question, so I
went through the Asterisk, The Future of Telephony book, all the notes
I've gathered, and every single link that looked even mildly promising. 
 
From what I see, you're asking if you can use a regular expression to match
against the dialed number. You _COULD_ if you wanted to do something like
this (on ALL calls being dialed by your users)
 
exten = _X.,1,GotoIf($[${REGEX(YOUR_REGEX_HERE ${EXTEN})} =
1]?800-in,s,1:2)
 
Easy-read format (how I design stuff before writing it fully)
 
exten = anynumber
If the number dialed matches the regex pattern, goto the 800-in context,
at the s extension, priority 1
Otherwise, continue on to priority 2 in this extension  context.
 
If you'd like to save a minute amount of cpu/mem/time, you can match against
10+ numbers only, using 
_XX. as the pattern to match against, which will match against any
set of numbers 10 or more digits long.
 
Hope this was somewhat helpful
 
SKM
 


From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sean Kennedy
Sent: Saturday, December 10, 2005 11:33 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] extensions and regular expressions (
probablyan easy question )


Rich,

It's kind of tough to truly understand what you are trying to accomplish 

Ack, sorry!  It's hard to post to the list on a saturday when my 2year old
is wanting to play with the keyboard as well.  Best I can do is half a mind,
most of the time that's enough.  

Not always, however.  :)

(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

I have an 800 number from teliax.  When my local users dial it, they will
dial 1866... instead of the 866 I have in my dial plan.  I do not want the
call to use one of my external sources to terminate the call ( in essence,
dialing out via voicepulse, and recieving the call via teliax ).  I know I
can do two seperate exten patterns, but I was hoping for a single pattern.
To that end, I was wondering if there was a way of saying Match this 0 or 1
times, something I'm used to in perl and the like.

If there isn't, there isn't.  Won't kill me to add the second exten match.

Sean

Rich Adamson wrote: 

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:

RE: [Asterisk-Users] SRV Lookups

2005-12-11 Thread Kerry Garrison

My comment was not specific to SRV lookups and more along the lines that
without it Asterisk is not usable in production environments. This type of
comment is a bit misleading at best. Without SRV lookups, Asterisk may not
be usable in YOUR environment but that hardly devalues Asterisk as a whole.
Not that it wouldnt make some things much easier (and possible) but it does
not preclude every possible scenerio.
-Kerry (sorry, having a bad day, didnt mean to be argumentative)

  _  

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Sunday, December 11, 2005 9:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] SRV Lookups


Hi Kerry.
 
Did you see Kevin's subsequent post about using Realtime to share SIP
registry info?
 
Uhhh... you already quoted my previous message on that topic stating
that it was not supported at this time. In any given situation, it may
or may not work properly, depending on exactly what the servers and
clients are doing.

Even if the code had been written, there will still be many issues
involved in actually implementing it, including (but not limited to) NAT
traversal, call limit handling, registration expiration and others. It
also mandates that there can be _no_ caching of peer/user information in
memory, which currently means there is no 'qualify' or MWI notification
possible.
 
Now, I can be a real jerk and say I told you so, or I could inquire as to
just how you got it working when it isn't supposed to? This limitation is
proving to be a real thorn in our side and I would just die to get it to
work.
 
Doug.

 
 

-Original Message- 
From: Kerry Garrison [mailto:[EMAIL PROTECTED] 
Sent: Sun 12/11/2005 1:41 PM 
To: 'Asterisk Users Mailing List - Non-Commercial Discussion' 
Cc: 
Subject: RE: [Asterisk-Users] SRV Lookups



Dammit, I better go pull my servers out of all of my client's locations
because their production servers have just been rendered unusable. Time to
take their old Toshiba system out of mothballs an.wait a sec, it didn't
do it either, what now? I guess IP Telephony has just died today. Sad, and
it had so much promise.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Aaron Daniel
Sent: Sunday, December 11, 2005 12:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SRV Lookups

HUH?  I better turn my servers off, they've been doing this for months now
0.o


On Dec 11, 2005, at 1:50 PM, Douglas Garstang wrote:

 I guess we can put that up there with the inability to share a common
 Realtime database between Asterisk servers for SIP peers too...
 another serious limitation.

 -Original Message-
 From: Douglas Garstang
 Sent: Sunday, December 11, 2005 12:49 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: RE: [Asterisk-Users] SRV Lookups


 Sounds like your saying that a serious limitation that effectively
 makes Asterisk unusable in a production environment isn't a priority
 for the 'official' developers. Awesome...

 -Original Message-
 From: Leif Madsen [mailto:[EMAIL PROTECTED]
 Sent: Sunday, December 11, 2005 12:18 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] SRV Lookups


 On 12/11/05, Douglas Garstang [EMAIL PROTECTED] wrote:
 Anyone know when Asterisk is going to properly support DNS SRV
 Lookups?

 Well, luckily Asterisk is open source so you have the ability to code
 this yourself. If you can't program in C (like myself), then you have
 the option of either hiring someone directly. Another option is to
 create a bounty and see if anyone else also requires this
 functionality and is willing to contribute some money for development.

 Leif Madsen
 http://www.oreilly.com/catalog/asterisk
 http://www.leifmadsen.com
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To

[Asterisk-Users] Mechanisms for Implementing a Common Contact Database

Well, I feel like I'm flogging a dead horse here, but I figured I'd at least do 
my bit and throw some ideas out there about how to implement a common contact 
database now that it's been confirmed that Realtime can't do this.
 
1. A script that runs on one or more boxes in the Asterisk farm and at (very) 
regular intervals replicates the differences in the /var/lib/asterisk/astdb 
file, such that each Asterisk box always knows wheve every phone is and can 
therefore terminate calls to all phones. Wouldn't be too hard to write. Major 
con is that Asterisk has to be reloaded to see the new astdb file and I don't 
think I'd want to be performing reloads on a production system every say, 10s 
or so.
 
2. It'd be cool if the regcontext command actually did something. There's a 
myth out there that it does something like execute a command upon registration. 
Even the O'Reilly The Future of Telephony seems to think this. After reading 
some posts in the developer discussion I can say it doesn't. It would be great 
though, if upon registration from a phone, Asterisk could perform some action, 
say for example copying the registration to another Asterisk system.
 
3. Phones register to OpenSER. Openser upon a REGISTER forwards the packets to 
N number of Asterisk boxes who then get a registration for the user. I've 
tested this and it works. Each Asterisk system then knows where every phone is 
and can terminate calls to any phone. However, I'm no SER expert (is anyone) 
and when the phones start sending SUBSCRIBE and receiving NOTIFY messages, with 
SER in the middle, it all gets very ugly.
 
4. Could all Asterisk systems be given access to a common file store? NFS or a 
SAN maybe? Does anyone know what what happen if multiple Asterisk boxes tried 
to use/write to the same astdb file? Anyone? Anyone?
 
5. DUNDi? Couldn't get it to work. Spent weeks on it and then realised that all 
the examples have RSA keys in what I thought was sip.conf. The sip.conf file 
doesn't support RSA keys.
 
I really hopes someone takes the time to reply to this message. Surely I'm not 
the only person in the universe who's trying to implement a HA Asterisk system. 
I would think it would be for the good of all to come up with some sort of a 
solution.
 
It amazes me that when I search for Asterisk redundancy on google, I keep 
coming up with my own posts!
 
I feel like I have been beating my head against a wall. Every time I think I'm 
close to coming up with some sort of hack or solution, some other limitation 
gets in the way and stops it dead in it's tracks.
 
 
 

 

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[Asterisk-Users] New Product ID.

2005-12-11 Thread Goran Donev









I am asking all the VOIP Gurus and any developers out there
if a product exists and if not if anyone would want to help me develop such
product. 



With the onslaught of new homes that are wired with
networking capabilities. I was wondering if there is a product out there
developed that can be used by Asterisk for intercom systems in homes, business
or multi-dwelling buildings. I want to know if there is a system that you can
install that will use SIP as the communication mechanism but install in every
room and dial the extension of the rooms or do an extension that does a
broadcast for all the intercoms. If this product exists can someone tell me who
makes it and point me out to the websites. If not if someone is interested in
developing such a product and cobranding it let me know.



This unit would be an all in one system wall mounted in
rooms that can be used inside or outside of entrance doors without a special
intercom system. 



I believe that such a device would allow better marketing for
Asterisk and VOIP systems to make their entrance in the residential field. This
would allow builders to further push VOIP in their new dwellings.



Thanks. 










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RE: [Asterisk-Users] Major Advantages of Asterisk vs Nortel PBX Systems

2005-12-11 Thread Rushowr




Depends on what Nortel you're using. But here's just a few 
I can say from being a Nortel PBX guy a while back:


  
  Cost (depends on the situation)
  
  Ability to write just about ANYTHING to interact with your phone 
  system. Just about anything you want can be done if you apply the time to do 
  it, or find someone that already has.
  
  RealTime Database storage of (this works reliably in my 
  experience):
  

 Call Detail Records

 SIP Users (phones)

 Voicemail Settings for the 
Users

I also use the DB to store other user options that are 
not part of the system.
  
  No need to purchase modules for extra 
functions
  
  Powerful dialplan and IVR control:
  

Match information against a regular _expression_, even if 
the information comes from an external source?

Query other PBXs to find a route to a number 
(DUNDi)

Supoprt Video (withoutlarge expense, Asterisk has 
SIP Video support already)

Loopa portion of the dialplan while a 
conditionis false
  
  Use on ANY hardware that's capable of handling the 
  load?
  
  You can use the PBX hardwareto do other things WHILE it's acting 
  as the PBX. (This can vary obviously)
  
  Management API for monitoring and 
controlling
I could go on and on... I'm no "Asterisk is the solution 
for everyone's PBX" guy, but it can work for just about any situation if it's 
done right. 

Thanks,
SKM



From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of 
DakotaSent: Friday, December 09, 2005 8:02 PMTo: 
Asterisk Users Mailing List - Non-Commercial DiscussionSubject: 
[Asterisk-Users] Major Advantages of Asterisk vs Nortel PBX 
Systems

What are the major advantages of Asterisk vs Nortel 
PBX Systems?
I am trying to justify whether it makes sense 
moving forward with an Asterisk installation.


Dakota
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RE: [Asterisk-Users] SRV Lookups

Kerry, thanks for the reply. The subject here is a bit misleading. I think I 
forgot to change it at some point. Ayway, I am REALLY curious how you got 
Realtime to allow sharing of SIP contact info is it a fluke of chance maybe?

-Original Message- 
From: [EMAIL PROTECTED] on behalf of Kerry Garrison 
Sent: Sun 12/11/2005 10:39 PM 
To: 'Asterisk Users Mailing List - Non-Commercial Discussion' 
Cc: 
Subject: RE: [Asterisk-Users] SRV Lookups

My comment was not specific to SRV lookups and more along the lines 
that without it Asterisk is not usable in production environments. This type of 
comment is a bit misleading at best. Without SRV lookups, Asterisk may not be 
usable in YOUR environment but that hardly devalues Asterisk as a whole. Not 
that it wouldnt make some things much easier (and possible) but it does not 
preclude every possible scenerio.
-Kerry (sorry, having a bad day, didnt mean to be argumentative)

  _  

From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas 
Garstang
Sent: Sunday, December 11, 2005 9:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] SRV Lookups

Hi Kerry.

Did you see Kevin's subsequent post about using Realtime to share SIP 
registry info?

Uhhh... you already quoted my previous message on that topic stating
that it was not supported at this time. In any given situation, it may
or may not work properly, depending on exactly what the servers and
clients are doing.

Even if the code had been written, there will still be many issues
involved in actually implementing it, including (but not limited to) NAT
traversal, call limit handling, registration expiration and others. It
also mandates that there can be _no_ caching of peer/user information in
memory, which currently means there is no 'qualify' or MWI notification
possible.

Now, I can be a real jerk and say I told you so, or I could inquire as 
to just how you got it working when it isn't supposed to? This limitation is 
proving to be a real thorn in our side and I would just die to get it to work.

Doug.

-Original Message- 
From: Kerry Garrison [mailto:[EMAIL PROTECTED] 
Sent: Sun 12/11/2005 1:41 PM 
To: 'Asterisk Users Mailing List - Non-Commercial Discussion' 
Cc: 
Subject: RE: [Asterisk-Users] SRV Lookups

Dammit, I better go pull my servers out of all of my client's 
locations
because their production servers have just been rendered 
unusable. Time to
take their old Toshiba system out of mothballs an.wait a 
sec, it didn't
do it either, what now? I guess IP Telephony has just died 
today. Sad, and
it had so much promise.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Aaron Daniel
Sent: Sunday, December 11, 2005 12:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SRV Lookups

HUH?  I better turn my servers off, they've been doing this for 
months now
0.o

On Dec 11, 2005, at 1:50 PM, Douglas Garstang wrote:

 I guess we can put that up there with the inability to share 
a common
 Realtime database between Asterisk servers for SIP peers 
too...
 another serious limitation.

 -Original Message-
 From: Douglas Garstang
 Sent: Sunday, December 11, 2005 12:49 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: RE: [Asterisk-Users] SRV Lookups

 Sounds like your saying that a serious limitation that 
effectively
 makes Asterisk unusable in a production environment isn't a 
priority
 for the 'official' developers. Awesome...

 -Original Message-
 From: Leif Madsen [mailto:[EMAIL PROTECTED]
 Sent: Sunday, December 11, 2005 12:18 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] SRV Lookups

 On 12/11/05, Douglas Garstang [EMAIL PROTECTED] wrote:
 Anyone

RE: [Asterisk-Users] Mechanisms for Implementing a Common ContactDatabase