date:20090403

My old idea was to implement an option, since there are many people  
with different opinions
on how a PBX should behave when a channel is put on hold.

An option could control how we should handle the bridged channel when  
the caller or the callee
puts a call on hold. It could either be local hold, meaning we  
entertain the user with music,
or a remote hold, which means that we send the hold forward over ISDN  
or SIP and let the
other end handle the hold. This would also work well in larger  
Asterisk installations,
where you don't want to fill up trunks between Asterisk servers with  
music. The edge server
provides the music, no one else.

In SIP we could easily add a proprietary header for music class  
suggestion in these cases.

Asterisk admins should be able to set this option per call in the  
dialplan or per device in
channel configurations - or per PBX, also in channel configs.

local hold or remote hold might mean something else, coming to  
think of it. But it fitted
in nicely here :-)

/Olle

2 apr 2009 kl. 15.05 skrev Richard Brady:

 Furthermore, the following two IETF documents address the need to both
 signal the hold and provide the music:

 1. RFC 5359 (Session Initiation Protocol Service Examples)

 2. draft-worley-service-example-03 (Session Initiation Protocol
 Service Example -- Music on Hold)

 Unfortunately they both address more complex scenarios and solutions,
 but they do back me up on the fact that there are good reasons to both
 signal hold and provide music.

 R.

 On Wed, Apr 1, 2009 at 6:16 PM, Richard Brady rnbr...@gmail.com  
 wrote:
 Hi Tony

 I can see where you guys are coming from on this and have already
 enumerated your argument in my own email.

 But there are very real reasons for a PBX to signal the hold even  
 when
 it wants to send its own MOH:

 1. Bandwidth: under your scheme the PBX would continue to receive
 bandwidth-consuming media without using it.
 2. Privacy: the far-end has an expectation of privacy while on hold
 and should have the option to mute automatically when held.
 3. Feature richness: signalling the hold enables such innovative
 features such as reverse hold.
 4. ISDN interworking: ISDN supports this and SIP should be compatible
 with that (as per standard ITU-T Q.1912.5)

 Also, can you explain why the PBX would use a=sendonly but not
 dispatch media. Why not a=inactive for that case?

 IMHO, PBX-A would be broken if it passed this along the Hold  
 message to downstream and then started servicing the MOH itself

 Remember it is not a hold message, it is a media attribute and we are
 discussing how that should be interpreted within the context of the
 hold feature in traditional telephony.

 I would also like to point out in my defence that there are several
 telephone systems in the field which behave as I described (Nortel
 BCM50, Aastra Intelligate, Mitel 3300 to name a few).

 Regards,
 Richard


 I have to agree with Kevin on this one.

 I fail to understand how you have a PBX-A talking to Asterisk  
 talking to PBX-B and the PBX-A placing the call on hold.   
 Typically you should have a Client/Phone to PBX-A to Asterisk to  
 PBX-B to Client/Phone/VoiceMail.

 If the Client signals Hold, the PBX should NOT be passing that  
 Hold status on but transition audio stream from Client to MOH  
 (assuming MOH is handled).  Asterisk shouldn't notice a thing  
 except more RTP packets (or less if it is my teenage daughter on  
 the phone as the case may be).

 IMHO, PBX-A would be broken if it passed this along the Hold  
 message to downstream and then started servicing the MOH itself on  
 the RTP stream.  That just doesn't make sense.

 Now if PBX-A were not a PBX and were a SIP Router, and the SIP  
 Router was attempting this, I can see how it would Re-Invite, but  
 it shouldn't pass the hold status onto Asterisk.

 Need some clarity here.

 Tony Plack


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Re: [asterisk-users] SIP 183 progessl


2 apr 2009 kl. 17.40 skrev Danny Nicholas:

 Sipaddheader(180 Ringing) might do the trick.

This was very bad and propably untested advice. You're mixing a  
response with a header...

sipaddheader adds a header to the outbound INVITE. It does not handle  
any responses.

While I appreciate your efforts in trying to help, it would be better  
not shooting from the hip ;-)

/O


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Re: [asterisk-users] SIP vs RTP destination IP


2 apr 2009 kl. 17.45 skrev David Ruggles:

 Is it possible to have asterisk override the connection information  
 embedded
 in a SIP 200 packet with the registration information? I have  
 multihomed
 machines with softphones and they register just fine and sip works  
 fine, but
 the RTP packets get sent to the ip from the SIP connection  
 information and
 the softphones are sending the wrong ip. I can't find an option in the
 softphone to change ip it sends.

If you turn on NAT support, we will ignore all IP addresses in the 200  
OK and
just send our media directly to wherever the other end sends it from.

/O

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Re: [asterisk-users] Asterisk G729 codec...


2 apr 2009 kl. 20.42 skrev Kevin P. Fleming:

 Danny Nicholas wrote:
 You should not have a G729 command on the CLI.  Codecs are  
 addressed in
 sip.conf, dahdi.conf, etc.  restarting Asterisk might do the  
 trick.  You
 only need to reboot for a driver level change.

 This is incorrect. Digium's codec_g729a.so module does in fact add a
 'g729 show' command to the CLI, when it has found at least one valid
 license file. so that the user can see how many of their licensed
 channels are in use.

 If the 'g729 show' command is not available after you have loaded the
 module, then you need to look closely at your Asterisk log files  
 because
 the module was not able to find any valid license files.


Is this also available as a manager command?

I would really appreciate being able to check license status over  
manager.

/Olle

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Re: [asterisk-users] SIP Context Confusion

Or you could use the domain feature, where you set a default context  
per domain, that overrides the one in the general section.

/Olle

3 apr 2009 kl. 07.08 skrev Martin:

 Hi,

 It took me a while to understand what you were saying ... more clarity
 to your emails!

 I see where the code says  If we have a context defined, overwrite
 the original context and after consideration
 I agree with you ... the only problem is that even if you don't define
 the context=blah for the user... that user
 inherits the default context

 However since you did find it in the source code I'm sure you can fix
 it for yourself. Just check against the default_context
 and do not overwrite the user's context if it's default.

 Or add another flag to the user's definition for example
 is_context_set that would be NULL if no context keyword is processed
 from the sip.conf etc.
 That is easier to check instead of comparing against default_context

 Martin

 On Wed, Apr 1, 2009 at 2:45 PM, Anthony Plack t...@plack.net wrote:
 Okay, I am not understanding if I have this correct or not.

 I have a requirement to allow guests into a PBX from different  
 domains.  However, I can not allow the guests into the default  
 context because each domain has its own IVR.  So I end up setting  
 the domain context.  I also need to provide separate contexts for  
 different sip users (different dial groups).  Small system, few  
 users, so it doesn't make sense to create separate Asterisk boxes  
 (cost wise and support) and some of the prompts are similar.  Same  
 company, different micro departments and web domains.  Should need  
 to either.

 If I set the user context to user1 and have set a domain context  
 set to guests1 in sip.conf, the system is ignoring the user1  
 context.  An incoming call (from the code) will be force the  
 context to guests1 and not have the user1.  I quote:

/* If we have a context defined, overwrite the original  
 context */

 For example, in sip.conf:

[general]
context=fromsip
domain=domain1.tld,guests1
domain=domain2.tld,guests2

[userA]
context=user1

 It would seem to me, that if the context was NOT set in the SIP  
 entry, and a domain context was available, only then would you  
 replace the context.

 To me, I would go from micro to macro definition and not jump  
 around.  So we would have peer, domain, general in the SIP context  
 hierarchy.  Instead we have domain, peer, general.

 What am I missing about why this is setup this way (other than that  
 is the way it has always been)?

 Looking for some instruction here to wrap my head around this better.

 As stands now, I believe I have to set all the phones up to a  
 domain without a context to allow the local context to be used.  Is  
 that correct?

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[asterisk-users] VoIP Farm

2009-04-03 Thread Gabriel - IP Guys

Dear All,

 

Thank you for your comments and suggestions about my term 'VoIP Farm'.
It was done in jest, and I'm glad to see how the community reacted, with
off list emails with some with advice, some with warnings, some with
funny comments, and some sending me a promise of $30 for usage of the
word for three years! To lay it to rest, I know I can't trademark VoIP
Farm, but thanks to those who suggested I try!

 

I'm very happy with all our current asterisk installations, thanks to
this project, there are people able to run effective businesses using
VoIP, and I'm going to attempt to build a load balanced asterisk
implementation, and name it the voip farm! And yes, when I get stuck,
and google refuses to help me, I'll be running straight here.

 

Have a good day all, and thank  you for making this guy smile!

 

---

Mr Gabriel

 

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Re: [asterisk-users] Local music on hold -- mohinterpret=passthrough assymetrical ?

2009-04-03 Thread Richard Brady

Exvito

Did you ever make any progress on this?

Richard



On Mon, Mar 10, 2008 at 2:38 AM, Ex Vito ex.vitor...@gmail.com wrote:
  Hi list,

  I'm planning and testing a distributed asterisk deployment
  throughout several sites; each will be connected to the PSTN
  and all of them among themselves via IAX trunks. Phones
  will be SIP.

  I guess I already solved (worked-around, actually) asterisk's
  codec negotiation limitations regarding local G.711 utilization vs.
  remote G.729 while minimizing transcoding -- a bit of dial plan
  tweaking via the setting of SIP_CODEC variable seems to do
  the trick. But I digress... (with patch in issue 4825 things would
  be much nicer!)

  Now I'm still trying to improve bandwith usage with local music
  on hold; that is, when sip user A1, registered to server A puts
  sip caller B1, registered to server B, caller B1 gets server B's
  music on hold -- this removes the need of streaming audio from
  server A to server B while B1 is on hold, which in my scenario
  is a good thing.

  I post to the list trying to get peer feedback to my initial tests.
  The configurations I mention are always applied to both
  servers A and B.

  1. If I set mohinterpret=passthrough + mohsuggest=default
      in the [general] section of iax.conf the local music on hold
      never works. Results:

      bad - A1 calls B1, B1 puts A1 on hold, A1 gets B's music
      bad - A1 calls B1, A1 puts B1 on hold, B1 gets A's music
      bad - B1 calls A1, A1 puts B1 on hold, B1 gets A's music
      bad - B1 calls A1, B1 puts A1 on hold, A1 gets B's music

  2. If I set mohinterpret=passthrough + mohsuggest=default
      in the specific peer/user (friend, actually) section I get improved
      results but not perfect (or, at least, as I'd like them to be).
      Results:

      good - A1 calls B1, B1 puts A1 on hold, A1 gets A's music
      bad -   A1 calls B1, A1 puts B1 on hold, B1 gets A's music
      good - B1 calls A1, A1 puts B1 on hold, B1 gets B's music
      bad -   B1 calls A1, B1 puts A1 on hold, A1 gets B's music

  Fortunatelly, the good cases seem to be the most plausible ones.

  So, in my observation, the mohinterpret=passthrough behaviour
  is not symmetrical; that is, the hold signalling only seems to
  travel one way along the IAX trunk... From the side receiving the
  call to the side initiating it, and not the other way around.

  Can anyone verify this behaviour ? Am I doing something wrong
  or is this expected / by design behaviour ?

  Should I file a bug against 1. ? Against 2. ?


  Extra points question:

  Since the calls in this case are remote, from site A to site B, the
  codec in use is G.729 which, as you might well know, is really
  awfull at supporting music since it's been designed for voice only.

  How would one have the RTP stream renegotiated during call
  to G.711 when entering music on hold (local, of course, after fixing
  my issues above!) and back to G.729 when back to conversation ?

  (ok, this probably needs patching the source !... but maybe someone
  has an idea or has taken a different approach at this...)

  :-)


  Thanks a lot for any feedback,
 --
  exvito


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Re: [asterisk-users] Asterisk doesn't relay remote MOH during hold

2009-04-03 Thread Richard Brady

Agreed Olle, it would definitely have to be option driven, not least
for backward compatibility.

When you say old idea, is there any discussion we can refer to?

Exisiting variables include:

mohinterpret
mohsuggest
musicclass
musiconhold

The first step would be to clarify what each of these are for. Then
perhaps we can add options for those which cover the scenarios we are
interested in.

Of course we we need to understand those scenarios too. So, let's look
at that. For each channel in the call you need to know how it holds
and how it likes to be held.

Ways it may hold:
1.1. a=sendonly and sends its own MOH (most likely a PBX)
1.2. a= sendonly and expects MOH to be generated upstream (most likely
a handset)
1.3. a=inactive and expects MOH to be generated upstream (could be PBX
or handset)
1.4. No signalling, it will simply substitute media

Ways it may like to be held:
2.1. Send it a=sendonly and send it MOH (could be PBX or handset)
2.2. Send it a= sendonly and no media (inside a network as you mentioned)
2.3. Send it a=inactive and no media (could be PBX or handset)
2.4. No signalling, simply send it substituted media.

At first glance you would think that it would hold as it likes to be
held. But actually a handset could use 1.2. while expecting 2.4 as it
cannot generate hold music for either it's own user when put on hold
or the remote user when holding.

So do we need two variables with 4 values each? I don't think so. We
only need to disambiguate between 1.1 and 1.2, and to choose between
2.1 through 2.4. Hopefully there is some scope to narrow that down
further. I will think about it some more.

Giving chan_sip support for the mohinterpret=passthrough option would
would be a start. But that option itself is ambiguous: does it mean
media passthrough or signalling passthrough? This ambiguity is
highlighted in the unanswered message from exvito on this list in
March last year:

[asterisk-users] Local music on hold -- mohinterpret=passthrough assymetrical ?

So some thought definitely needs to go into this before it becomes a
feature request.

R.


On Fri, Apr 3, 2009 at 9:03 AM, Olle E. Johansson o...@edvina.net wrote:
 My old idea was to implement an option, since there are many people
 with different opinions
 on how a PBX should behave when a channel is put on hold.

 An option could control how we should handle the bridged channel when
 the caller or the callee
 puts a call on hold. It could either be local hold, meaning we
 entertain the user with music,
 or a remote hold, which means that we send the hold forward over ISDN
 or SIP and let the
 other end handle the hold. This would also work well in larger
 Asterisk installations,
 where you don't want to fill up trunks between Asterisk servers with
 music. The edge server
 provides the music, no one else.

 In SIP we could easily add a proprietary header for music class
 suggestion in these cases.

 Asterisk admins should be able to set this option per call in the
 dialplan or per device in
 channel configurations - or per PBX, also in channel configs.

 local hold or remote hold might mean something else, coming to
 think of it. But it fitted
 in nicely here :-)

 /Olle

 2 apr 2009 kl. 15.05 skrev Richard Brady:

 Furthermore, the following two IETF documents address the need to both
 signal the hold and provide the music:

 1. RFC 5359 (Session Initiation Protocol Service Examples)

 2. draft-worley-service-example-03 (Session Initiation Protocol
 Service Example -- Music on Hold)

 Unfortunately they both address more complex scenarios and solutions,
 but they do back me up on the fact that there are good reasons to both
 signal hold and provide music.

 R.

 On Wed, Apr 1, 2009 at 6:16 PM, Richard Brady rnbr...@gmail.com
 wrote:
 Hi Tony

 I can see where you guys are coming from on this and have already
 enumerated your argument in my own email.

 But there are very real reasons for a PBX to signal the hold even
 when
 it wants to send its own MOH:

 1. Bandwidth: under your scheme the PBX would continue to receive
 bandwidth-consuming media without using it.
 2. Privacy: the far-end has an expectation of privacy while on hold
 and should have the option to mute automatically when held.
 3. Feature richness: signalling the hold enables such innovative
 features such as reverse hold.
 4. ISDN interworking: ISDN supports this and SIP should be compatible
 with that (as per standard ITU-T Q.1912.5)

 Also, can you explain why the PBX would use a=sendonly but not
 dispatch media. Why not a=inactive for that case?

 IMHO, PBX-A would be broken if it passed this along the Hold
 message to downstream and then started servicing the MOH itself

 Remember it is not a hold message, it is a media attribute and we are
 discussing how that should be interpreted within the context of the
 hold feature in traditional telephony.

 I would also like to point out in my defence that there are several
 telephone systems in the field which behave as I

Re: [asterisk-users] Simple Queue question

2009-04-03 Thread Lenz Emilitri

You tried setting the call limit for the Agent's phone?

l.

2009/4/3 Steve Edwards asterisk@sedwards.com

 On Thu, 2 Apr 2009, Haim Dimer wrote:

  The issue is the that the agent needs to wait on the phone for a call to
  come in. I read
  http://www.voip-info.org/wiki/view/Asterisk+cmd+AgentCallbackLogin but
  it will be deprecated and the doc/queues-with-callback-members.txt means
  that I would have to convert to AEL (unless I can do extensions.conf and
  extensions.ael at the same time. Not sure)

 I'm a 1.2 Luddite, but you can use both extensions.conf and
 extensions.ael.

 You can load an ael and do a show dialplan to see how Asterisk
 converts AEL to conf.

 Thanks in advance,
 
 Steve Edwards  sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
 Newline Fax: +1-760-731-3000

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Re: [asterisk-users] Asterisk G729 codec...

2009-04-03 Thread Kevin P. Fleming

Olle E. Johansson wrote:

 Is this also available as a manager command?
 
 I would really appreciate being able to check license status over  
 manager.

It is not today, but I'll make a note to add it to the next builds,
which will probably happen next week.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem...@digium.com
Check us out at www.digium.com  www.asterisk.org

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[asterisk-users] agi no longer working with 1.4 svn 186229

2009-04-03 Thread John covici

The minute asterisk tries to execute an agi, it gets utils.c write
error broken pipe and so hangs up the call.

Anyone know what is going on?

I am using kernel 2.6.27 with dahdi trunk if that makes a difference.

thanks in advance for any ideas.

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

 John Covici
 cov...@ccs.covici.com

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Re: [asterisk-users] FXS Line Voltage When Dahdi/Zaptel is off?

2009-04-03 Thread Noah Miller

 You mean when the driver is not loaded ?
 It doesn't. The driver enables the current drawn.

 Well that is my guess. But since I have one card handy I'll confirm for you.
 CONFIRMED. No power without the driver loaded

Excellent.  Thanks, Martin!  I didn't have one to test with (yet).





 Martin

 On Thu, Apr 2, 2009 at 4:38 PM, Noah Miller noahisaacmil...@gmail.com wrote:
 Hi -

 Does anybody know if an FXS generates line voltage when Dahdi/Zaptel
 is disabled?


 Thanks,
 Noah

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[asterisk-users] Unichan wtih Te201p alarms

2009-04-03 Thread criptos

 
 
I'm using a Te201p card, with unichan, I want to know if my channels are 
ready or in alarm... but uc show channel o uc show channels, doesn't show 
me anything...

Any Ideas? 

thanks.


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Re: [asterisk-users] Please Advice SIP 183 progessl

2009-04-03 Thread Khaled W. Chehab

Dears


Kindly find my dial script below,I am trying to send the caller 180 ringing
but all tries were failed,
The caller always receive 183 session Progress
Even I add in the sip.conf 
progressinband=never

or if there any way to stop the music on hold and let the caller hear the
Ring Back Tone 

exten = _X.,1,Wait(1)
exten = _X.,n,SetMusicOnHold(English)
exten = _X.,n,WaitMusicOnHold(2)
exten = _X.,n,NoOp(Return-)
exten = _X.,n,Dial(SIP/Gateway1/8430${EXTEN}|300|m)
;exten = _X.,n,SIPAddHeader(Alert-Info: Ring Answer)
exten = _X.,n,Goto(y-${DIALSTATUS},1)   ; Jump based on status
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = y-NOANSWER,1,SetMusicOnHold(busy)
exten = y-NOANSWER,n,WaitMusicOnHold(18) ; If busy, Playback Busy /
NOANSWER announce
exten = y-BUSY,1,SetMusicOnHold(busy)
exten = y-BUSY,n,WaitMusicOnHold(18) ; If busy, Playback Busy / busy
announce
exten = _y-.,1,Goto(y-NOANSWER,1)   ; Treat anything else as no answer
exten = _X.,n,HangUp()

Please Advice







-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, April 02, 2009 6:40 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] SIP 183 progessl

Sipaddheader(180 Ringing) might do the trick.

If you are compiling your own asterisk, you could change chan_sip.c to
replace 183 Session Progress with 180 Ringing (line 3950 in my source)
but that might break something else.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Thursday, April 02, 2009 10:23 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] SIP 183 progessl

Can you please tell me how to Custom SIP header

Regards


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, April 02, 2009 6:16 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Xorcom and Doorbell

Custom SIP header?

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Thursday, April 02, 2009 10:02 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Xorcom and Doorbell

Dears

How can I send or force sending 180 Ringing instead of 183 back to the
caller ?or send both sequential if its impossible
I used progressinband=never but it did work .


Regards




*
No employee or agent is authorized to conclude any binding agreement on
behalf of Xplorium with another party by e-mail without express written
confirmation by an officer of Xplorium. Any views expressed by an individual
in this electronic message do not necessarily reflect views of Xplorium or
its subsidiaries and associates.

This electronic message and its attachments are solely addressed to the
addressee(s), and contain confidential information protected from disclosure
belonging to Xplorium.

If you are not the intended addressee of this electronic message and its
attachments, kindly delete it immediately from your system and notify the
sender by electronic mail. You must not copy this message or attachment or
disclose its content to any other person.

Xplorium does not guarantee the integrity of this electronic message and any
of its attachments, or that they are free from computer viruses or other
defects.
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No employee or agent is authorized to conclude any binding agreement on
behalf of Xplorium with another party by e-mail without express written
confirmation by an officer of Xplorium. Any views expressed by an individual
in this electronic message do not necessarily reflect views of Xplorium or
its subsidiaries and associates.

This electronic message and its attachments are solely addressed to the
addressee(s), and contain confidential information protected from disclosure
belonging to Xplorium.

If you are not the intended addressee of this electronic message and its
attachments, kindly delete it immediately from your system and notify the
sender by electronic mail. You must not copy this message or attachment or

Re: [asterisk-users] Please Advice SIP 183 progessl

2009-04-03 Thread David Gibbons

The fact that you sent this again (what is that -- 3 times now?) AND with high 
importance, will likely cause people to ignore your messages rather than trying 
to help you.

There are few things that annoy me more than messages sent with high importance 
(same category of annoyance as messages written in all caps). Let's have a 
little bit of intarweb etiquette.

-Dave

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W. Chehab
Sent: Friday, April 03, 2009 9:28 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Please Advice SIP 183 progessl
Importance: High

Dears


Kindly find my dial script below,I am trying to send the caller 180 ringing
but all tries were failed,
The caller always receive 183 session Progress
Even I add in the sip.conf
progressinband=never

or if there any way to stop the music on hold and let the caller hear the
Ring Back Tone

exten = _X.,1,Wait(1)
exten = _X.,n,SetMusicOnHold(English)
exten = _X.,n,WaitMusicOnHold(2)
exten = _X.,n,NoOp(Return-)
exten = _X.,n,Dial(SIP/Gateway1/8430${EXTEN}|300|m)
;exten = _X.,n,SIPAddHeader(Alert-Info: Ring Answer)
exten = _X.,n,Goto(y-${DIALSTATUS},1)   ; Jump based on status
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = y-NOANSWER,1,SetMusicOnHold(busy)
exten = y-NOANSWER,n,WaitMusicOnHold(18) ; If busy, Playback Busy /
NOANSWER announce
exten = y-BUSY,1,SetMusicOnHold(busy)
exten = y-BUSY,n,WaitMusicOnHold(18) ; If busy, Playback Busy / busy
announce
exten = _y-.,1,Goto(y-NOANSWER,1)   ; Treat anything else as no answer
exten = _X.,n,HangUp()

Please Advice







-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, April 02, 2009 6:40 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] SIP 183 progessl

Sipaddheader(180 Ringing) might do the trick.

If you are compiling your own asterisk, you could change chan_sip.c to
replace 183 Session Progress with 180 Ringing (line 3950 in my source)
but that might break something else.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Thursday, April 02, 2009 10:23 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] SIP 183 progessl

Can you please tell me how to Custom SIP header

Regards


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, April 02, 2009 6:16 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Xorcom and Doorbell

Custom SIP header?

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Thursday, April 02, 2009 10:02 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Xorcom and Doorbell

Dears

How can I send or force sending 180 Ringing instead of 183 back to the
caller ?or send both sequential if its impossible
I used progressinband=never but it did work .


Regards




*
No employee or agent is authorized to conclude any binding agreement on
behalf of Xplorium with another party by e-mail without express written
confirmation by an officer of Xplorium. Any views expressed by an individual
in this electronic message do not necessarily reflect views of Xplorium or
its subsidiaries and associates.

This electronic message and its attachments are solely addressed to the
addressee(s), and contain confidential information protected from disclosure
belonging to Xplorium.

If you are not the intended addressee of this electronic message and its
attachments, kindly delete it immediately from your system and notify the
sender by electronic mail. You must not copy this message or attachment or
disclose its content to any other person.

Xplorium does not guarantee the integrity of this electronic message and any
of its attachments, or that they are free from computer viruses or other
defects.
*



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Re: [asterisk-users] Simple Queue question

2009-04-03 Thread Haim Dimer

I have, and it only works for SIP. I also have IAX2 phones connected ...

On Fri, Apr 3, 2009 at 5:39 AM, Lenz Emilitri lenz.lo...@gmail.com wrote:
 You tried setting the call limit for the Agent's phone?

 l.

 2009/4/3 Steve Edwards asterisk@sedwards.com

 On Thu, 2 Apr 2009, Haim Dimer wrote:

  The issue is the that the agent needs to wait on the phone for a call to
  come in. I read
  http://www.voip-info.org/wiki/view/Asterisk+cmd+AgentCallbackLogin but
  it will be deprecated and the doc/queues-with-callback-members.txt means
  that I would have to convert to AEL (unless I can do extensions.conf and
  extensions.ael at the same time. Not sure)

 I'm a 1.2 Luddite, but you can use both extensions.conf and
 extensions.ael.

 You can load an ael and do a show dialplan to see how Asterisk
 converts AEL to conf.

 Thanks in advance,
 
 Steve Edwards      sedwa...@sedwards.com      Voice: +1-760-468-3867 PST
 Newline                                             Fax: +1-760-731-3000

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 --
 Loway - home of QueueMetrics - http://queuemetrics.com


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Re: [asterisk-users] Please Advice SIP 183 progessl

2009-04-03 Thread Steve Howes

And don't top post ;)

On 3 Apr 2009, at 14:38, David Gibbons wrote:

 The fact that you sent this again (what is that -- 3 times now?) AND  
 with high importance, will likely cause people to ignore your  
 messages rather than trying to help you.

 There are few things that annoy me more than messages sent with high  
 importance (same category of annoyance as messages written in all  
 caps). Let's have a little bit of intarweb etiquette.

 -Dave

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com 
 [mailto:asterisk-users-boun...@lists.digium.com 
 ] On Behalf Of Khaled W. Chehab
 Sent: Friday, April 03, 2009 9:28 AM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: Re: [asterisk-users] Please Advice SIP 183 progessl
 Importance: High

 Dears


 Kindly find my dial script below,I am trying to send the caller 180  
 ringing
 but all tries were failed,
 The caller always receive 183 session Progress
 Even I add in the sip.conf
 progressinband=never

 or if there any way to stop the music on hold and let the caller  
 hear the
 Ring Back Tone

 exten = _X.,1,Wait(1)
 exten = _X.,n,SetMusicOnHold(English)
 exten = _X.,n,WaitMusicOnHold(2)
 exten = _X.,n,NoOp(Return-)
 exten = _X.,n,Dial(SIP/Gateway1/8430${EXTEN}|300|m)
 ;exten = _X.,n,SIPAddHeader(Alert-Info: Ring Answer)
 exten = _X.,n,Goto(y-${DIALSTATUS},1)   ; Jump based on status
 (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
 exten = y-NOANSWER,1,SetMusicOnHold(busy)
 exten = y-NOANSWER,n,WaitMusicOnHold(18) ; If busy, Playback Busy /
 NOANSWER announce
 exten = y-BUSY,1,SetMusicOnHold(busy)
 exten = y-BUSY,n,WaitMusicOnHold(18) ; If busy, Playback Busy / busy
 announce
 exten = _y-.,1,Goto(y-NOANSWER,1)   ; Treat anything else as no  
 answer
 exten = _X.,n,HangUp()

 Please Advice







 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny  
 Nicholas
 Sent: Thursday, April 02, 2009 6:40 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: Re: [asterisk-users] SIP 183 progessl

 Sipaddheader(180 Ringing) might do the trick.

 If you are compiling your own asterisk, you could change chan_sip.c to
 replace 183 Session Progress with 180 Ringing (line 3950 in my  
 source)
 but that might break something else.

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled  
 W.
 Chehab
 Sent: Thursday, April 02, 2009 10:23 AM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: [asterisk-users] SIP 183 progessl

 Can you please tell me how to Custom SIP header

 Regards


 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny  
 Nicholas
 Sent: Thursday, April 02, 2009 6:16 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: Re: [asterisk-users] Xorcom and Doorbell

 Custom SIP header?

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled  
 W.
 Chehab
 Sent: Thursday, April 02, 2009 10:02 AM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: Re: [asterisk-users] Xorcom and Doorbell

 Dears

 How can I send or force sending 180 Ringing instead of 183 back to the
 caller ?or send both sequential if its impossible
 I used progressinband=never but it did work .


 Regards




 *
 No employee or agent is authorized to conclude any binding agreement  
 on
 behalf of Xplorium with another party by e-mail without express  
 written
 confirmation by an officer of Xplorium. Any views expressed by an  
 individual
 in this electronic message do not necessarily reflect views of  
 Xplorium or
 its subsidiaries and associates.

 This electronic message and its attachments are solely addressed to  
 the
 addressee(s), and contain confidential information protected from  
 disclosure
 belonging to Xplorium.

 If you are not the intended addressee of this electronic message and  
 its
 attachments, kindly delete it immediately from your system and  
 notify the
 sender by electronic mail. You must not copy this message or  
 attachment or
 disclose its content to any other person.

 Xplorium does not guarantee the integrity of this electronic message  
 and any
 of its attachments, or that they are free from computer viruses or  
 other
 defects.
 *



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 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] Please Advice SIP 183 progessl

2009-04-03 Thread David Gibbons

Lol.

I'm actually in the small minority who prefers top posting to bottom posting.

-d

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes
Sent: Friday, April 03, 2009 10:07 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Please Advice SIP 183 progessl

And don't top post ;)

On 3 Apr 2009, at 14:38, David Gibbons wrote:

 The fact that you sent this again (what is that -- 3 times now?) AND
 with high importance, will likely cause people to ignore your
 messages rather than trying to help you.

 There are few things that annoy me more than messages sent with high
 importance (same category of annoyance as messages written in all
 caps). Let's have a little bit of intarweb etiquette.

 -Dave

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com 
 [mailto:asterisk-users-boun...@lists.digium.com
 ] On Behalf Of Khaled W. Chehab
 Sent: Friday, April 03, 2009 9:28 AM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: Re: [asterisk-users] Please Advice SIP 183 progessl
 Importance: High

 Dears


 Kindly find my dial script below,I am trying to send the caller 180
 ringing
 but all tries were failed,
 The caller always receive 183 session Progress
 Even I add in the sip.conf
 progressinband=never

 or if there any way to stop the music on hold and let the caller
 hear the
 Ring Back Tone

 exten = _X.,1,Wait(1)
 exten = _X.,n,SetMusicOnHold(English)
 exten = _X.,n,WaitMusicOnHold(2)
 exten = _X.,n,NoOp(Return-)
 exten = _X.,n,Dial(SIP/Gateway1/8430${EXTEN}|300|m)
 ;exten = _X.,n,SIPAddHeader(Alert-Info: Ring Answer)
 exten = _X.,n,Goto(y-${DIALSTATUS},1)   ; Jump based on status
 (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
 exten = y-NOANSWER,1,SetMusicOnHold(busy)
 exten = y-NOANSWER,n,WaitMusicOnHold(18) ; If busy, Playback Busy /
 NOANSWER announce
 exten = y-BUSY,1,SetMusicOnHold(busy)
 exten = y-BUSY,n,WaitMusicOnHold(18) ; If busy, Playback Busy / busy
 announce
 exten = _y-.,1,Goto(y-NOANSWER,1)   ; Treat anything else as no
 answer
 exten = _X.,n,HangUp()

 Please Advice







 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny
 Nicholas
 Sent: Thursday, April 02, 2009 6:40 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: Re: [asterisk-users] SIP 183 progessl

 Sipaddheader(180 Ringing) might do the trick.

 If you are compiling your own asterisk, you could change chan_sip.c to
 replace 183 Session Progress with 180 Ringing (line 3950 in my
 source)
 but that might break something else.

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled
 W.
 Chehab
 Sent: Thursday, April 02, 2009 10:23 AM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: [asterisk-users] SIP 183 progessl

 Can you please tell me how to Custom SIP header

 Regards


 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny
 Nicholas
 Sent: Thursday, April 02, 2009 6:16 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: Re: [asterisk-users] Xorcom and Doorbell

 Custom SIP header?

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled
 W.
 Chehab
 Sent: Thursday, April 02, 2009 10:02 AM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: Re: [asterisk-users] Xorcom and Doorbell

 Dears

 How can I send or force sending 180 Ringing instead of 183 back to the
 caller ?or send both sequential if its impossible
 I used progressinband=never but it did work .


 Regards




 *
 No employee or agent is authorized to conclude any binding agreement
 on
 behalf of Xplorium with another party by e-mail without express
 written
 confirmation by an officer of Xplorium. Any views expressed by an
 individual
 in this electronic message do not necessarily reflect views of
 Xplorium or
 its subsidiaries and associates.

 This electronic message and its attachments are solely addressed to
 the
 addressee(s), and contain confidential information protected from
 disclosure
 belonging to Xplorium.

 If you are not the intended addressee of this electronic message and
 its
 attachments, kindly delete it immediately from your system and
 notify the
 sender by electronic mail. You must not copy this message or
 attachment or
 disclose its content to any other person.

 Xplorium does not guarantee the integrity of this electronic message
 and any
 of its attachments, or that they are free from computer viruses or
 other
 defects.

Re: [asterisk-users] Please Advice SIP 183 progessl

2009-04-03 Thread Singer XJ Wang

What's wrong with top posting?
David Gibbons wrote:
 Lol.

 I'm actually in the small minority who prefers top posting to bottom posting.

 -d

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com 
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes
 Sent: Friday, April 03, 2009 10:07 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Please Advice SIP 183 progessl

 And don't top post ;)

 On 3 Apr 2009, at 14:38, David Gibbons wrote:

   
 The fact that you sent this again (what is that -- 3 times now?) AND
 with high importance, will likely cause people to ignore your
 messages rather than trying to help you.

 There are few things that annoy me more than messages sent with high
 importance (same category of annoyance as messages written in all
 caps). Let's have a little bit of intarweb etiquette.

 -Dave

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com 
 [mailto:asterisk-users-boun...@lists.digium.com
 ] On Behalf Of Khaled W. Chehab
 Sent: Friday, April 03, 2009 9:28 AM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: Re: [asterisk-users] Please Advice SIP 183 progessl
 Importance: High

 Dears


 Kindly find my dial script below,I am trying to send the caller 180
 ringing
 but all tries were failed,
 The caller always receive 183 session Progress
 Even I add in the sip.conf
 progressinband=never

 or if there any way to stop the music on hold and let the caller
 hear the
 Ring Back Tone

 exten = _X.,1,Wait(1)
 exten = _X.,n,SetMusicOnHold(English)
 exten = _X.,n,WaitMusicOnHold(2)
 exten = _X.,n,NoOp(Return-)
 exten = _X.,n,Dial(SIP/Gateway1/8430${EXTEN}|300|m)
 ;exten = _X.,n,SIPAddHeader(Alert-Info: Ring Answer)
 exten = _X.,n,Goto(y-${DIALSTATUS},1)   ; Jump based on status
 (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
 exten = y-NOANSWER,1,SetMusicOnHold(busy)
 exten = y-NOANSWER,n,WaitMusicOnHold(18) ; If busy, Playback Busy /
 NOANSWER announce
 exten = y-BUSY,1,SetMusicOnHold(busy)
 exten = y-BUSY,n,WaitMusicOnHold(18) ; If busy, Playback Busy / busy
 announce
 exten = _y-.,1,Goto(y-NOANSWER,1)   ; Treat anything else as no
 answer
 exten = _X.,n,HangUp()

 Please Advice







 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny
 Nicholas
 Sent: Thursday, April 02, 2009 6:40 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: Re: [asterisk-users] SIP 183 progessl

 Sipaddheader(180 Ringing) might do the trick.

 If you are compiling your own asterisk, you could change chan_sip.c to
 replace 183 Session Progress with 180 Ringing (line 3950 in my
 source)
 but that might break something else.

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled
 W.
 Chehab
 Sent: Thursday, April 02, 2009 10:23 AM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: [asterisk-users] SIP 183 progessl

 Can you please tell me how to Custom SIP header

 Regards


 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny
 Nicholas
 Sent: Thursday, April 02, 2009 6:16 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: Re: [asterisk-users] Xorcom and Doorbell

 Custom SIP header?

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled
 W.
 Chehab
 Sent: Thursday, April 02, 2009 10:02 AM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: Re: [asterisk-users] Xorcom and Doorbell

 Dears

 How can I send or force sending 180 Ringing instead of 183 back to the
 caller ?or send both sequential if its impossible
 I used progressinband=never but it did work .


 Regards




 *
 No employee or agent is authorized to conclude any binding agreement
 on
 behalf of Xplorium with another party by e-mail without express
 written
 confirmation by an officer of Xplorium. Any views expressed by an
 individual
 in this electronic message do not necessarily reflect views of
 Xplorium or
 its subsidiaries and associates.

 This electronic message and its attachments are solely addressed to
 the
 addressee(s), and contain confidential information protected from
 disclosure
 belonging to Xplorium.

 If you are not the intended addressee of this electronic message and
 its
 attachments, kindly delete it immediately from your system and
 notify the
 sender by electronic mail. You must not copy this message or
 attachment or
 disclose its content to any other person.

 Xplorium does not guarantee the integrity of this electronic message
 and any
 of its attachments, or that they are free from

Re: [asterisk-users] Unichan wtih Te201p alarms

2009-04-03 Thread Moises Silva

Use dahdi_tool to see that.

On Fri, Apr 3, 2009 at 9:24 AM, criptos crip...@aullox.com wrote:


 I'm using a Te201p card, with unichan, I want to know if my channels are
 ready or in alarm... but uc show channel o uc show channels, doesn't show
 me anything...

 Any Ideas?

 thanks.


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-- 
I do not agree with what you have to say, but I’ll defend to the
death your right to say it. Voltaire

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Re: [asterisk-users] Please Advice SIP 183 progessl

2009-04-03 Thread Steve Howes

A: It destroys the conversation flow
Q: What is wrong with top posting?

On 3 Apr 2009, at 15:14, Singer XJ Wang wrote:

 What's wrong with top posting?
 David Gibbons wrote:

 Lol.

 I'm actually in the small minority who prefers top posting to  
 bottom posting.

 -d

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com 
 [mailto:asterisk-users-boun...@lists.digium.com 
 ] On Behalf Of Steve Howes
 Sent: Friday, April 03, 2009 10:07 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Please Advice SIP 183 progessl

 And don't top post ;)

 On 3 Apr 2009, at 14:38, David Gibbons wrote:


 The fact that you sent this again (what is that -- 3 times now?) AND
 with high importance, will likely cause people to ignore your
 messages rather than trying to help you.

 There are few things that annoy me more than messages sent with high
 importance (same category of annoyance as messages written in all
 caps). Let's have a little bit of intarweb etiquette.

 -Dave

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com 
 [mailto:asterisk-users-boun...@lists.digium.com
 ] On Behalf Of Khaled W. Chehab
 Sent: Friday, April 03, 2009 9:28 AM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: Re: [asterisk-users] Please Advice SIP 183 progessl
 Importance: High

 Dears


 Kindly find my dial script below,I am trying to send the caller 180
 ringing
 but all tries were failed,
 The caller always receive 183 session Progress
 Even I add in the sip.conf
 progressinband=never

 or if there any way to stop the music on hold and let the caller
 hear the
 Ring Back Tone

 exten = _X.,1,Wait(1)
 exten = _X.,n,SetMusicOnHold(English)
 exten = _X.,n,WaitMusicOnHold(2)
 exten = _X.,n,NoOp(Return-)
 exten = _X.,n,Dial(SIP/Gateway1/8430${EXTEN}|300|m)
 ;exten = _X.,n,SIPAddHeader(Alert-Info: Ring Answer)
 exten = _X.,n,Goto(y-${DIALSTATUS},1)   ; Jump based on status
 (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
 exten = y-NOANSWER,1,SetMusicOnHold(busy)
 exten = y-NOANSWER,n,WaitMusicOnHold(18) ; If busy, Playback Busy /
 NOANSWER announce
 exten = y-BUSY,1,SetMusicOnHold(busy)
 exten = y-BUSY,n,WaitMusicOnHold(18) ; If busy, Playback Busy /  
 busy
 announce
 exten = _y-.,1,Goto(y-NOANSWER,1)   ; Treat anything else as no
 answer
 exten = _X.,n,HangUp()

 Please Advice







 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny
 Nicholas
 Sent: Thursday, April 02, 2009 6:40 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: Re: [asterisk-users] SIP 183 progessl

 Sipaddheader(180 Ringing) might do the trick.

 If you are compiling your own asterisk, you could change  
 chan_sip.c to
 replace 183 Session Progress with 180 Ringing (line 3950 in my
 source)
 but that might break something else.

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled
 W.
 Chehab
 Sent: Thursday, April 02, 2009 10:23 AM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: [asterisk-users] SIP 183 progessl

 Can you please tell me how to Custom SIP header

 Regards


 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny
 Nicholas
 Sent: Thursday, April 02, 2009 6:16 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: Re: [asterisk-users] Xorcom and Doorbell

 Custom SIP header?

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled
 W.
 Chehab
 Sent: Thursday, April 02, 2009 10:02 AM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: Re: [asterisk-users] Xorcom and Doorbell

 Dears

 How can I send or force sending 180 Ringing instead of 183 back to  
 the
 caller ?or send both sequential if its impossible
 I used progressinband=never but it did work .


 Regards




 *
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 Xplorium or
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[asterisk-users] New ViciDial Call Center Suite Release: 2.0.5

2009-04-03 Thread Matt Florell

Hello,

We've released another update to our VICIDIAL/astGUIclient call center
suite: 2.0.5

http://astguiclient.sf.net/

The call center suite client applications run on most modern web
browsers on almost any GUI-capable operating system, and it includes
the VICIDIAL call center suite.
This package is free and AGPLv2.
This package is geared towards Asterisk installations with SIP,IAX or
Zap phones and Zaptel, IAX or SIP trunks.

For this release, we have added hundreds of new features including
Asterisk phone, trunk and DID configuration through the VICIDIAL web
interface. We have also tested the suite on Asterisk versions through
1.2.30.2 and 1.4.21.2.

All client web-apps and administration pages are available in English,
Spanish, Greek, German, Italian and French, with rough translations of
Polish, Portuguese, Brazillian Portuguese, Slovak, Russian and Dutch
for the client web-apps only.

Check out the project blog for more information:
http://astguiclient.blogspot.com

Let me know what you think.

Thanks,


MATT---

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Re: [asterisk-users] Please Advice SIP 183 progessl

2009-04-03 Thread Cary Fitch

FWIW, ($0.0007) I prefer top posting.  I follow the postings live, I know
the topics and comments, and don't want to read the stuff over and over to
get to the bottom new stuff.

YMMV.

Cary

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes
Sent: Friday, April 03, 2009 9:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Please Advice SIP 183 progessl

A: It destroys the conversation flow
Q: What is wrong with top posting?

On 3 Apr 2009, at 15:14, Singer XJ Wang wrote:

 What's wrong with top posting?
 David Gibbons wrote:

 Lol.

 I'm actually in the small minority who prefers top posting to  
 bottom posting.

 -d

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com 
 ] On Behalf Of Steve Howes
 Sent: Friday, April 03, 2009 10:07 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Please Advice SIP 183 progessl

 And don't top post ;)

 On 3 Apr 2009, at 14:38, David Gibbons wrote:


 The fact that you sent this again (what is that -- 3 times now?) AND
 with high importance, will likely cause people to ignore your
 messages rather than trying to help you.

 There are few things that annoy me more than messages sent with high
 importance (same category of annoyance as messages written in all
 caps). Let's have a little bit of intarweb etiquette.

 -Dave

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com
 ] On Behalf Of Khaled W. Chehab
 Sent: Friday, April 03, 2009 9:28 AM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: Re: [asterisk-users] Please Advice SIP 183 progessl
 Importance: High

 Dears


 Kindly find my dial script below,I am trying to send the caller 180
 ringing
 but all tries were failed,
 The caller always receive 183 session Progress
 Even I add in the sip.conf
 progressinband=never

 or if there any way to stop the music on hold and let the caller
 hear the
 Ring Back Tone

 exten = _X.,1,Wait(1)
 exten = _X.,n,SetMusicOnHold(English)
 exten = _X.,n,WaitMusicOnHold(2)
 exten = _X.,n,NoOp(Return-)
 exten = _X.,n,Dial(SIP/Gateway1/8430${EXTEN}|300|m)
 ;exten = _X.,n,SIPAddHeader(Alert-Info: Ring Answer)
 exten = _X.,n,Goto(y-${DIALSTATUS},1)   ; Jump based on status
 (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
 exten = y-NOANSWER,1,SetMusicOnHold(busy)
 exten = y-NOANSWER,n,WaitMusicOnHold(18) ; If busy, Playback Busy /
 NOANSWER announce
 exten = y-BUSY,1,SetMusicOnHold(busy)
 exten = y-BUSY,n,WaitMusicOnHold(18) ; If busy, Playback Busy /  
 busy
 announce
 exten = _y-.,1,Goto(y-NOANSWER,1)   ; Treat anything else as no
 answer
 exten = _X.,n,HangUp()

 Please Advice







 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny
 Nicholas
 Sent: Thursday, April 02, 2009 6:40 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: Re: [asterisk-users] SIP 183 progessl

 Sipaddheader(180 Ringing) might do the trick.

 If you are compiling your own asterisk, you could change  
 chan_sip.c to
 replace 183 Session Progress with 180 Ringing (line 3950 in my
 source)
 but that might break something else.

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled
 W.
 Chehab
 Sent: Thursday, April 02, 2009 10:23 AM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: [asterisk-users] SIP 183 progessl

 Can you please tell me how to Custom SIP header

 Regards


 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny
 Nicholas
 Sent: Thursday, April 02, 2009 6:16 PM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: Re: [asterisk-users] Xorcom and Doorbell

 Custom SIP header?

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled
 W.
 Chehab
 Sent: Thursday, April 02, 2009 10:02 AM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: Re: [asterisk-users] Xorcom and Doorbell

 Dears

 How can I send or force sending 180 Ringing instead of 183 back to  
 the
 caller ?or send both sequential if its impossible
 I used progressinband=never but it did work .


 Regards




 *
 No employee or agent is authorized to conclude any binding agreement
 on
 behalf of Xplorium with another party by e-mail without express
 written
 confirmation by an officer of Xplorium. Any views expressed by an
 individual
 in this electronic message do not necessarily reflect views of

[asterisk-users] Bridging Avaya IP systems and Cisco IP system

2009-04-03 Thread Gavin Henry

Hi all,

Has anyone put * in between an Avaya and Cisco system to connect two
offices together?

I was thinking about adding a SIP trunk on each side and getting
Asterisk to pass calls between them. There is a leased line for
bandwidth.

Any tips/ideas on whether this is possible or dumb?

Thanks.

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Re: [asterisk-users] FXS Line Voltage When Dahdi/Zaptel is off?

On Thu, Apr 02, 2009 at 11:28:04PM -0500, Martin wrote:
 You mean when the driver is not loaded ?
 It doesn't. The driver enables the current drawn.

Note: that's when the *driver* is loaded. Regardless of whether or not
the channel is configured with Asterisk.

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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Re: [asterisk-users] Ring group howto

2009-04-03 Thread Joseph L. Casale

The only cleaner way is to define the group in [globals] as follows:-

[globals]
group1 = SIP/3615221401SIP/3615221402SIP/3615221407SIP/52260014

...and then refer to this variable in the dial statement...

exten = 5226001454,1,Dial(${group1},20)

That certainly makes life easier, is there a way to associate this
to a context, or ring a context?

jlc

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Re: [asterisk-users] Asterisk + Cisco Call Manager

2009-04-03 Thread David Backeberg

On Thu, Apr 2, 2009 at 12:07 PM, Timothy Smith timotsm...@gmail.com wrote:
 In our office, we're migrating from a Cisco set up to Asterisk.

What is the goal of doing this migration?
Plenty of people do a blended environment with Cisco doing what Cisco
does well and Asterisk doing what Asterisk does well.

 Am now faced with the challenge relaying incoming calls from asterisk
 to call manager. Has anyone done that before?

I don't really have a good idea of what call manager is / does, nor
why you would want to relay incoming calls from asterisk to call
manager. If you're talking about reusing IVRs or other things that you
built in Cisco, those are straightforward to build in Asterisk. If you
really like the GUI for building IVRs I recommend trying out FreePBX
(or others), which provides a GUI on top of asterisk for tasks like
that.

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Re: [asterisk-users] meetme dahdi and zaptel

2009-04-03 Thread Dave Poirier

Thanks for the reply Martin. I'm pretty sure that we are running dahdi (see
below). Could the problem be that I didn't remove the Zaptel libraries prior
to compiling Asterisk? If that's the case I should be able to rerun the
./configure without the zaptel libs and see if that fixes it. I'm just not
sure what it checks for though. I did set the dahdichanname=no in the
asterisk.conf if that makes any difference. It seemed to in calling the
channel in the dialplan but didn't seem to effect the meetme app.

Thanks,
Dave

Relevent bits from lsmod

Module  Size  Used by
dahdi_dummy38984  0
dahdi_echocan_mg2  39048  0
xpp_usb52304  0
xpp   226468  1 xpp_usb
wctc4xxp   83392  0
dahdi_transcode42376  1 wctc4xxp
wcb4xxp   110756  0
wctdm  73804  0
wcfxo  47136  0
wctdm24xxp159332  0
wcte11xp   59936  0
wct1xxp48544  0
wcte12xp  102404  0
wct4xxp   349696  24
dahdi 232144  66
dahdi_dummy,dahdi_echocan_mg2,xpp,dahdi_transcode,wcb4xxp,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wcte12xp,wct4xxp


From the console...

asterisk*CLI dahdi show status
Description  Alarms IRQbpviol
CRC4
T2XXP (PCI) Card 0 Span 1OK 0  0
0
T2XXP (PCI) Card 0 Span 2RED0  0  0




On Thu, Apr 2, 2009 at 9:40 PM, Martin asteriskl...@callthem.info wrote:

 That's very strange ... the code when is compiling checks whether
 zaptel is present and then
 the #define HAVE_ZAPTEL is set.

 Since your error says No ZAP channel ...

 and the code says

 ast_log(LOG_WARNING, No %s channel available for conference, user
 introduction disabled\n, dahdi_chan_name);

 and

 in main/asterisk.c


 #ifdef HAVE_ZAPTEL
 static char _dahdi_chan_name[AST_CHANNEL_NAME] = Zap;
 #else
 static char _dahdi_chan_name[AST_CHANNEL_NAME] = DAHDI;
 #endif

 I deduct from that ... that you're still running zaptel and not dahdi.
 Because your log should say No DAHDI channel available ... UNLESS
 for some reason you only compiled
 chan_dahdi.so and copied it manually leaving the old app_meetme.so
 with HAVE_ZAPTEL flag...

 paste your lsmod output

 Martin



-- 
David Poirier
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Re: [asterisk-users] Asterisk G729 codec...


3 apr 2009 kl. 15.14 skrev Kevin P. Fleming:

 Olle E. Johansson wrote:

 Is this also available as a manager command?

 I would really appreciate being able to check license status over
 manager.

 It is not today, but I'll make a note to add it to the next builds,
 which will probably happen next week.


Thank you!

I would be happy to help out, but it's not open source, you know ;-)

Have a nice weekend!

/O

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[asterisk-users] Seg Fault after upgrade to Asterisk 1.6.0.8

2009-04-03 Thread M Hulber

Went from 1.6.0.6 to 1.6.0.8 and resulted in segmentation fault.  
Reverted to 1.6.0.6 and back to normal.

--

Linux asterisk.hulber.com 2.6.18-128.1.1.el5 #1 SMP Mon Jan 26 13:58:24 
EST 2009 x86_64 x86_64 x86_64 GNU/Linux

Apr  3 11:49:56 asterisk kernel: asterisk[3780]: segfault at 
2ce1ac0537a8 rip 003e980715a8 rsp 7fff5bf00c30 error 4
Apr  3 11:50:00 asterisk kernel: asterisk[3828]: segfault at 
0400 rip 003e980758d9 rsp 7fffd3138ef0 error 4
Apr  3 11:50:04 asterisk kernel: asterisk[3879]: segfault at 
0c00 rip 003e980758d9 rsp 7fffde4cf280 error 4
Apr  3 11:50:09 asterisk kernel: asterisk[3927]: segfault at 
1c00 rip 003e980758d9 rsp 7fff2fd65b10 error 4
Apr  3 11:50:13 asterisk kernel: asterisk[3973]: segfault at 
2ce1ac04f948 rip 003e980715a8 rsp 7fff6c283fb0 error 4
Apr  3 11:50:17 asterisk kernel: asterisk[4022]: segfault at 
2ce1ac0486e8 rip 003e980715a8 rsp 7fff4e1d0f00 error 4
Apr  3 11:50:21 asterisk kernel: asterisk[4069]: segfault at 
2ce1ac067e28 rip 003e980715a8 rsp 7fff2f3ee120 error 4

Asterisk ended with exit status 139
Asterisk exited on signal 11.
Automatically restarting Asterisk.
mpg123: no process killed
/usr/sbin/safe_asterisk: line 117:  5322 Segmentation fault  (core 
dumped) nice -n $PRIORITY ${ASTSBINDIR}/asterisk -f ${CLIARGS} 
${ASTARGS} /dev/${TTY}  /dev/${TTY}
Asterisk ended with exit status 139
Asterisk exited on signal 11.
Automatically restarting Asterisk.
mpg123: no process killed
/usr/sbin/safe_asterisk: line 117:  5372 Segmentation fault  (core 
dumped) nice -n $PRIORITY ${ASTSBINDIR}/asterisk -f ${CLIARGS} 
${ASTARGS} /dev/${TTY}  /dev/${TTY}
Asterisk ended with exit status 139
Asterisk exited on signal 11.
Automatically restarting Asterisk.
mpg123: no process killed
/usr/sbin/safe_asterisk: line 117:  5419 Segmentation fault  (core 
dumped) nice -n $PRIORITY ${ASTSBINDIR}/asterisk -f ${CLIARGS} 
${ASTARGS} /dev/${TTY}  /dev/${TTY}
Asterisk ended with exit status 139
Asterisk exited on signal 11.
Automatically restarting Asterisk.
mpg123: no process killed
/usr/sbin/safe_asterisk: line 117:  5467 Segmentation fault  (core 
dumped) nice -n $PRIORITY ${ASTSBINDIR}/asterisk -f ${CLIARGS} 
${ASTARGS} /dev/${TTY}  /dev/${TTY}
Asterisk ended with exit status 139
Asterisk exited on signal 11.
Automatically restarting Asterisk.
mpg123: no process killed
/usr/sbin/safe_asterisk: line 117:  5514 Segmentation fault  (core 
dumped) nice -n $PRIORITY ${ASTSBINDIR}/asterisk -f ${CLIARGS} 
${ASTARGS} /dev/${TTY}  /dev/${TTY}
Asterisk ended with exit status 139
Asterisk exited on signal 11.
Automatically restarting Asterisk.

-- 

MARK.

Hulber Technologies
asterisk-ad...@hulber.com

Read my blog :  http://mark.hulber.com
Follow @hulber on Twitter:  http://twitter.com/hulber


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Re: [asterisk-users] meetme dahdi and zaptel

On Fri, Apr 03, 2009 at 08:24:41AM -0700, Dave Poirier wrote:
 Thanks for the reply Martin. I'm pretty sure that we are running dahdi (see
 below). Could the problem be that I didn't remove the Zaptel libraries prior
 to compiling Asterisk? 

No. With Asterisk = 1.4.22 it will check for DAHDI rather than
ZAPTEL (and with Asterisk 1.4.x ,  x= 22, DAHDI can also be zaptel
if Asterisk was built vs. Zaptel).

 If that's the case I should be able to rerun the
 ./configure without the zaptel libs and see if that fixes it. 

Shouldn't be an issue.

 I'm just not
 sure what it checks for though. I did set the dahdichanname=no in the
 asterisk.conf if that makes any difference. It seemed to in calling the
 channel in the dialplan but didn't seem to effect the meetme app.
 
 Thanks,
 Dave
 
 Relevent bits from lsmod
 
 Module  Size  Used by
 dahdi_dummy38984  0
 dahdi_echocan_mg2  39048  0
 xpp_usb52304  0
 xpp   226468  1 xpp_usb
 wctc4xxp   83392  0
 dahdi_transcode42376  1 wctc4xxp
 wcb4xxp   110756  0
 wctdm  73804  0
 wcfxo  47136  0
 wctdm24xxp159332  0
 wcte11xp   59936  0
 wct1xxp48544  0
 wcte12xp  102404  0
 wct4xxp   349696  24
 dahdi 232144  66
 dahdi_dummy,dahdi_echocan_mg2,xpp,dahdi_transcode,wcb4xxp,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wcte12xp,wct4xxp
 
 
 From the console...
 
 asterisk*CLI dahdi show status
 Description  Alarms IRQbpviol
 CRC4
 T2XXP (PCI) Card 0 Span 1OK 0  0
 0
 T2XXP (PCI) Card 0 Span 2RED0  0  0

This indicates that Asterisk was built with DAHDI support.

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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Re: [asterisk-users] Seg Fault after upgrade to Asterisk 1.6.0.8

2009-04-03 Thread Mark Michelson

M Hulber wrote:
 Went from 1.6.0.6 to 1.6.0.8 and resulted in segmentation fault.  
 Reverted to 1.6.0.6 and back to normal.
 
 --
 
 Linux asterisk.hulber.com 2.6.18-128.1.1.el5 #1 SMP Mon Jan 26 13:58:24 
 EST 2009 x86_64 x86_64 x86_64 GNU/Linux
 
 Apr  3 11:49:56 asterisk kernel: asterisk[3780]: segfault at 
 2ce1ac0537a8 rip 003e980715a8 rsp 7fff5bf00c30 error 4
 Apr  3 11:50:00 asterisk kernel: asterisk[3828]: segfault at 
 0400 rip 003e980758d9 rsp 7fffd3138ef0 error 4
 Apr  3 11:50:04 asterisk kernel: asterisk[3879]: segfault at 
 0c00 rip 003e980758d9 rsp 7fffde4cf280 error 4
 Apr  3 11:50:09 asterisk kernel: asterisk[3927]: segfault at 
 1c00 rip 003e980758d9 rsp 7fff2fd65b10 error 4
 Apr  3 11:50:13 asterisk kernel: asterisk[3973]: segfault at 
 2ce1ac04f948 rip 003e980715a8 rsp 7fff6c283fb0 error 4
 Apr  3 11:50:17 asterisk kernel: asterisk[4022]: segfault at 
 2ce1ac0486e8 rip 003e980715a8 rsp 7fff4e1d0f00 error 4
 Apr  3 11:50:21 asterisk kernel: asterisk[4069]: segfault at 
 2ce1ac067e28 rip 003e980715a8 rsp 7fff2f3ee120 error 4
 
 Asterisk ended with exit status 139
 Asterisk exited on signal 11.
 Automatically restarting Asterisk.
 mpg123: no process killed
 /usr/sbin/safe_asterisk: line 117:  5322 Segmentation fault  (core 
 dumped) nice -n $PRIORITY ${ASTSBINDIR}/asterisk -f ${CLIARGS} 
 ${ASTARGS} /dev/${TTY}  /dev/${TTY}
 Asterisk ended with exit status 139
 Asterisk exited on signal 11.
 Automatically restarting Asterisk.
 mpg123: no process killed
 /usr/sbin/safe_asterisk: line 117:  5372 Segmentation fault  (core 
 dumped) nice -n $PRIORITY ${ASTSBINDIR}/asterisk -f ${CLIARGS} 
 ${ASTARGS} /dev/${TTY}  /dev/${TTY}
 Asterisk ended with exit status 139
 Asterisk exited on signal 11.
 Automatically restarting Asterisk.
 mpg123: no process killed
 /usr/sbin/safe_asterisk: line 117:  5419 Segmentation fault  (core 
 dumped) nice -n $PRIORITY ${ASTSBINDIR}/asterisk -f ${CLIARGS} 
 ${ASTARGS} /dev/${TTY}  /dev/${TTY}
 Asterisk ended with exit status 139
 Asterisk exited on signal 11.
 Automatically restarting Asterisk.
 mpg123: no process killed
 /usr/sbin/safe_asterisk: line 117:  5467 Segmentation fault  (core 
 dumped) nice -n $PRIORITY ${ASTSBINDIR}/asterisk -f ${CLIARGS} 
 ${ASTARGS} /dev/${TTY}  /dev/${TTY}
 Asterisk ended with exit status 139
 Asterisk exited on signal 11.
 Automatically restarting Asterisk.
 mpg123: no process killed
 /usr/sbin/safe_asterisk: line 117:  5514 Segmentation fault  (core 
 dumped) nice -n $PRIORITY ${ASTSBINDIR}/asterisk -f ${CLIARGS} 
 ${ASTARGS} /dev/${TTY}  /dev/${TTY}
 Asterisk ended with exit status 139
 Asterisk exited on signal 11.
 Automatically restarting Asterisk.
 

Thanks for the information. Could you open a bug report at 
http://bugs.digium.com and upload a backtrace from the core dumps? Instructions 
for uploading a backtrace can be found in doc/backtrace.txt in the Asterisk 
source.

I suspect this is a regression introduced between 1.6.0.6 and 1.6.0.7 since 
1.6.0.8 is exactly the same as 1.6.0.7, except for the security fix for 
AST-2009-003.

Mark Michelson

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Re: [asterisk-users] meetme dahdi and zaptel

Well maybe turn the dahdichanname=no to yes...
And check if you can open cat /dev/dahdi/pseudo ... or better yet
maybe you're running asterisk with user asterisk
and it doesn't have access to /dev/dahdi/pseudo ... ? Meetme tries to
open that for timing source.

Martin

On Fri, Apr 3, 2009 at 10:24 AM, Dave Poirier dpoir...@mesd.k12.or.us wrote:
 Thanks for the reply Martin. I'm pretty sure that we are running dahdi (see
 below). Could the problem be that I didn't remove the Zaptel libraries prior
 to compiling Asterisk? If that's the case I should be able to rerun the
 ./configure without the zaptel libs and see if that fixes it. I'm just not
 sure what it checks for though. I did set the dahdichanname=no in the
 asterisk.conf if that makes any difference. It seemed to in calling the
 channel in the dialplan but didn't seem to effect the meetme app.

 Thanks,
 Dave

 Relevent bits from lsmod

 Module  Size  Used by
 dahdi_dummy    38984  0
 dahdi_echocan_mg2  39048  0
 xpp_usb    52304  0
 xpp   226468  1 xpp_usb
 wctc4xxp   83392  0
 dahdi_transcode    42376  1 wctc4xxp
 wcb4xxp   110756  0
 wctdm  73804  0
 wcfxo  47136  0
 wctdm24xxp    159332  0
 wcte11xp   59936  0
 wct1xxp    48544  0
 wcte12xp  102404  0
 wct4xxp   349696  24
 dahdi 232144  66
 dahdi_dummy,dahdi_echocan_mg2,xpp,dahdi_transcode,wcb4xxp,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wcte12xp,wct4xxp


 From the console...

 asterisk*CLI dahdi show status
 Description  Alarms IRQ    bpviol
 CRC4
 T2XXP (PCI) Card 0 Span 1    OK 0  0
 0
 T2XXP (PCI) Card 0 Span 2    RED    0  0  0




 On Thu, Apr 2, 2009 at 9:40 PM, Martin asteriskl...@callthem.info wrote:

 That's very strange ... the code when is compiling checks whether
 zaptel is present and then
 the #define HAVE_ZAPTEL is set.

 Since your error says No ZAP channel ...

 and the code says

 ast_log(LOG_WARNING, No %s channel available for conference, user
 introduction disabled\n, dahdi_chan_name);

 and

 in main/asterisk.c


 #ifdef HAVE_ZAPTEL
 static char _dahdi_chan_name[AST_CHANNEL_NAME] = Zap;
 #else
 static char _dahdi_chan_name[AST_CHANNEL_NAME] = DAHDI;
 #endif

 I deduct from that ... that you're still running zaptel and not dahdi.
 Because your log should say No DAHDI channel available ... UNLESS
 for some reason you only compiled
 chan_dahdi.so and copied it manually leaving the old app_meetme.so
 with HAVE_ZAPTEL flag...

 paste your lsmod output

 Martin



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Re: [asterisk-users] SIP Context Confusion

He's already using domain feature but its logic is to override the
user's context even if it was predefined in sip.conf

Martin

On Fri, Apr 3, 2009 at 3:14 AM, Olle E. Johansson o...@edvina.net wrote:
 Or you could use the domain feature, where you set a default context
 per domain, that overrides the one in the general section.

 /Olle

 3 apr 2009 kl. 07.08 skrev Martin:

 Hi,

 It took me a while to understand what you were saying ... more clarity
 to your emails!

 I see where the code says  If we have a context defined, overwrite
 the original context and after consideration
 I agree with you ... the only problem is that even if you don't define
 the context=blah for the user... that user
 inherits the default context

 However since you did find it in the source code I'm sure you can fix
 it for yourself. Just check against the default_context
 and do not overwrite the user's context if it's default.

 Or add another flag to the user's definition for example
 is_context_set that would be NULL if no context keyword is processed
 from the sip.conf etc.
 That is easier to check instead of comparing against default_context

 Martin

 On Wed, Apr 1, 2009 at 2:45 PM, Anthony Plack t...@plack.net wrote:
 Okay, I am not understanding if I have this correct or not.

 I have a requirement to allow guests into a PBX from different
 domains.  However, I can not allow the guests into the default
 context because each domain has its own IVR.  So I end up setting
 the domain context.  I also need to provide separate contexts for
 different sip users (different dial groups).  Small system, few
 users, so it doesn't make sense to create separate Asterisk boxes
 (cost wise and support) and some of the prompts are similar.  Same
 company, different micro departments and web domains.  Should need
 to either.

 If I set the user context to user1 and have set a domain context
 set to guests1 in sip.conf, the system is ignoring the user1
 context.  An incoming call (from the code) will be force the
 context to guests1 and not have the user1.  I quote:

        /* If we have a context defined, overwrite the original
 context */

 For example, in sip.conf:

        [general]
        context=fromsip
        domain=domain1.tld,guests1
        domain=domain2.tld,guests2

        [userA]
        context=user1

 It would seem to me, that if the context was NOT set in the SIP
 entry, and a domain context was available, only then would you
 replace the context.

 To me, I would go from micro to macro definition and not jump
 around.  So we would have peer, domain, general in the SIP context
 hierarchy.  Instead we have domain, peer, general.

 What am I missing about why this is setup this way (other than that
 is the way it has always been)?

 Looking for some instruction here to wrap my head around this better.

 As stands now, I believe I have to set all the phones up to a
 domain without a context to allow the local context to be used.  Is
 that correct?

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 ---
 * Olle E Johansson - o...@edvina.net
 * Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden




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[asterisk-users] ISDN Timer T309

2009-04-03 Thread Afonso Zimmermann





Hi everione,

I'm make some test with pri timer T309. I'm using asterisk 1.4.23.1,
libpri 1.4.9, dahdi-linux 2.1.0.4 and dahdi-tools 2.1.0.2. But in my
tests, the timer fail with a telco link in this scenario:

Telco Phone -- Telco --- Asterisk
 Sip Phone

When i make a call from Telco Phone to Sip Phone, the call complete,
but when i disconnect the link and reconnect in few seconds, the
Asterisk clear call:

[Apr 3 10:44:40] WARNING[13081]: chan_dahdi.c:4385 handle_alarms:
Detected alarm on channel 1: Red Alarm
[Apr 3 10:44:40] NOTICE[21088]: chan_dahdi.c:9558 pri_dchannel: PRI
got event: Alarm (4) on Primary D-channel of span 1
 == Spawn extension (disc-from-trunk-TR001, 9800, 2) exited
non-zero on 'DAHDI/1-1'
[Apr 3 10:44:40] WARNING[21088]: chan_dahdi.c:3021 pri_find_dchan: No
D-channels available! Using Primary channel 16 as D-channel anyway!
[Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms:
Detected alarm on channel 2: Red Alarm
[Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec:
Unable to disable echo cancellation on channel 2: Invalid argument
[Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms:
Detected alarm on channel 3: Red Alarm
[Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec:
Unable to disable echo cancellation on channel 3: Invalid argument
[Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms:
Detected alarm on channel 4: Red Alarm
[Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec:
Unable to disable echo cancellation on channel 4: Invalid argument
[Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms:
Detected alarm on channel 5: Red Alarm
[Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec:
Unable to disable echo cancellation on channel 5: Invalid argument
[Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms:
Detected alarm on channel 6: Red Alarm
[Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec:
Unable to disable echo cancellation on channel 6: Invalid argument
[Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms:
Detected alarm on channel 7: Red Alarm
[Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec:
Unable to disable echo cancellation on channel 7: Invalid argument
[Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms:
Detected alarm on channel 8: Red Alarm
[Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec:
Unable to disable echo cancellation on channel 8: Invalid argument
[Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms:
Detected alarm on channel 9: Red Alarm
[Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec:
Unable to disable echo cancellation on channel 9: Invalid argument
[Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms:
Detected alarm on channel 10: Red Alarm
[Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec:
Unable to disable echo cancellation on channel 10: Invalid argument
[Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms:
Detected alarm on channel 11: Red Alarm
[Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec:
Unable to disable echo cancellation on channel 11: Invalid argument
[Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms:
Detected alarm on channel 12: Red Alarm
[Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec:
Unable to disable echo cancellation on channel 12: Invalid argument
[Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms:
Detected alarm on channel 13: Red Alarm
[Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec:
Unable to disable echo cancellation on channel 13: Invalid argument
[Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms:
Detected alarm on channel 14: Red Alarm
[Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec:
Unable to disable echo cancellation on channel 14: Invalid argument
[Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms:
Detected alarm on channel 15: Red Alarm
[Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec:
Unable to disable echo cancellation on channel 15: Invalid argument
[Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms:
Detected alarm on channel 17: Red Alarm
[Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec:
Unable to disable echo cancellation on channel 17: Invalid argument
[Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms:
Detected alarm on channel 18: Red Alarm
[Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec:
Unable to disable echo cancellation on channel 18: Invalid argument
[Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms:
Detected alarm on channel 19: Red Alarm
[Apr 3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec:
Unable to disable echo cancellation on channel 19: Invalid

Re: [asterisk-users] opermode=?

2009-04-03 Thread bilal ghayyad


Thanks Tzafrir;

But did not get where to find drivers? I have zaptel.

  Hi All;
  
  If I need to set the opermode to King Saudi Arabia,
 what the name I 
  have to use? For example, to set it for kuwait then I
 use 
  opermode=KUWAIT. So what will be for Saudi Arabia?
 
 $ grep -i saudi drivers/dahdi/fxo_modes.h
 { .name = SAUDIARABIA,
 
 -- 
Tzafrir Cohen
 icq#16849755  jabber:tzafrir.co...@xorcom.com
 +972-50-7952406   mailto:tzafrir.co...@xorcom.com
 http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir
 
 
 
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 End of asterisk-users Digest, Vol 57, Issue 7
 *


  

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[asterisk-users] Radio interfaces for Asterisk - ISO image distro

2009-04-03 Thread John Todd


I just ran across these guys - looks very interesting:

  http://xelatec.com/xippr/install

They distribute a self-installing ISO with Asterisk, FreePBX, and some  
pre-built software to do radio over IP.  You'll need to buy the USB  
radio hardware, but it looks really interesting as a pre-built system  
for radio trunking using some of the Asterisk capabilities.  There is  
a very narrow but highly interested group of people who use radio  
interfaces for exotic locations such as oil platforms, ships, or  
remote off-grid locations to connect Asterisk systems together for  
long-haul telecommunications access.

Hopefully there will be a speaker this year at Astricon who will be  
going into a very complex and unique system in detail using these  
components; I'll announce if/when they schedule their talk.

JT


---
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Digium, Inc. | Asterisk Open Source Community Director
445 Jan Davis Drive NW -  Huntsville AL 35806  -   USA
direct: +1-256-428-6083 http://www.digium.com/




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[asterisk-users] SIP Warnning Message

2009-04-03 Thread César García

Guys, when registering I am getting this error message, my question is
that if this could be the reason whay I am able to make calls but not
to recieve call ?


[Apr  3 11:24:31] WARNING[19578]: chan_sip.c:15104
handle_response_register: Got 423 Interval too brief for service
+506phonenum...@domain.co.cr@host.ip.addr, minimum is 3600 seconds


Thanks


-- 
http://celord.blogspot.com/

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Re: [asterisk-users] meetme dahdi and zaptel

2009-04-03 Thread Dave Poirier

Asterisk runs as root on this system. Verified by looking at the running
processes.

#cat /dev/dahdi/pseudo
Predictably spits out the same char over and over.

I did manage to fix the behavior by removing dahdichanname=no from
asterisk.conf. This did however break any reference to ZAP channels in my
extensions.conf but since I set that as a global variable it was easy to
fix. Just set it to refer to DAHDI instead of ZAP. I still think there is a
bug somewhere but I am unable to find it.
Thanks for the help.
Dave

On Fri, Apr 3, 2009 at 10:09 AM, Martin asteriskl...@callthem.info wrote:

Well maybe turn the dahdichanname=no to yes...
And check if you can open cat /dev/dahdi/pseudo ... or better yet
maybe you're running asterisk with user asterisk
and it doesn't have access to /dev/dahdi/pseudo ... ? Meetme tries to
open that for timing source.

Martin

On Fri, Apr 3, 2009 at 10:24 AM, Dave Poirier dpoir...@mesd.k12.or.us
wrote:
Thanks for the reply Martin. I'm pretty sure that we are running dahdi
(see
below). Could the problem be that I didn't remove the Zaptel libraries
prior
to compiling Asterisk? If that's the case I should be able to rerun the
./configure without the zaptel libs and see if that fixes it. I'm just
not
sure what it checks for though. I did set the dahdichanname=no in the
asterisk.conf if that makes any difference. It seemed to in calling the
channel in the dialplan but didn't seem to effect the meetme app.

Thanks,
Dave

Relevent bits from lsmod

Module Size Used by
dahdi_dummy38984 0
dahdi_echocan_mg2 39048 0
xpp_usb52304 0
xpp 226468 1 xpp_usb
wctc4xxp 83392 0
dahdi_transcode42376 1 wctc4xxp
wcb4xxp 110756 0
wctdm 73804 0
wcfxo 47136 0
wctdm24xxp159332 0
wcte11xp 59936 0
wct1xxp48544 0
wcte12xp 102404 0
wct4xxp 349696 24
dahdi 232144 66

dahdi_dummy,dahdi_echocan_mg2,xpp,dahdi_transcode,wcb4xxp,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wcte12xp,wct4xxp

From the console...

asterisk*CLI dahdi show status
Description Alarms IRQbpviol
CRC4
T2XXP (PCI) Card 0 Span 1OK 0 0
0
T2XXP (PCI) Card 0 Span 2RED0 0
0

On Thu, Apr 2, 2009 at 9:40 PM, Martin asteriskl...@callthem.info
wrote:

That's very strange ... the code when is compiling checks whether
zaptel is present and then
the #define HAVE_ZAPTEL is set.

Since your error says No ZAP channel ...

and the code says

ast_log(LOG_WARNING, No %s channel available for conference, user
introduction disabled\n, dahdi_chan_name);

and

in main/asterisk.c

#ifdef HAVE_ZAPTEL
static char _dahdi_chan_name[AST_CHANNEL_NAME] = Zap;
#else
static char _dahdi_chan_name[AST_CHANNEL_NAME] = DAHDI;
#endif

I deduct from that ... that you're still running zaptel and not dahdi.
Because your log should say No DAHDI channel available ... UNLESS
for some reason you only compiled
chan_dahdi.so and copied it manually leaving the old app_meetme.so
with HAVE_ZAPTEL flag...

paste your lsmod output

Martin

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Re: [asterisk-users] Radio interfaces for Asterisk - ISO image distro

On Fri, Apr 03, 2009 at 10:43:08AM -0700, John Todd wrote:
 
 I just ran across these guys - looks very interesting:
 
   http://xelatec.com/xippr/install
 
 They distribute a self-installing ISO with Asterisk, FreePBX, and some  
 pre-built software to do radio over IP.  You'll need to buy the USB  
 radio hardware, but it looks really interesting as a pre-built system  
 for radio trunking using some of the Asterisk capabilities.  There is  
 a very narrow but highly interested group of people who use radio  
 interfaces for exotic locations such as oil platforms, ships, or  
 remote off-grid locations to connect Asterisk systems together for  
 long-haul telecommunications access.
 
 Hopefully there will be a speaker this year at Astricon who will be  
 going into a very complex and unique system in detail using these  
 components; I'll announce if/when they schedule their talk.

Now, why would they need a separate ISO for that? What extra software do
they have?

One change is obvious: chan_usbradio is not build by default. 

The output from cppcheck for it:

[channels/xpmr/xpmr.c:160]: (style) Redundant condition. It is safe to 
deallocate a NULL pointer
[channels/chan_usbradio.c:391]: (style) struct or union member 'sound::desc' is 
never used
Checking channels/chan_usbradio.c: HAVE_SYS_IO_H...
Checking channels/chan_usbradio.c: RADIO_XPMRX...
Checking channels/chan_usbradio.c: HAVE_XPMRX...
Checking channels/chan_usbradio.c: __linux...
Checking channels/chan_usbradio.c: defined(__FreeBSD__)...
Checking channels/chan_usbradio.c: NEW_ASTERISK...
[channels/chan_usbradio.c:390]: (style) struct or union member 'sound::ind' is 
never used
[channels/chan_usbradio.c:394]: (style) struct or union member 'sound::samplen' 
is never used
[channels/chan_usbradio.c:395]: (style) struct or union member 
'sound::silencelen' is never used
[channels/chan_usbradio.c:396]: (style) struct or union member 'sound::repeat' 
is never used
Checking channels/chan_usbradio.c: __FreeBSD__...
Checking channels/chan_usbradio.c: defined(__OpenBSD__)||defined(__NetBSD__)...
Checking channels/chan_usbradio.c: MIN...
Checking channels/chan_usbradio.c: MAX...
Checking channels/chan_usbradio.c: DEBUG_FILETEST==1...
Checking channels/chan_usbradio.c: __BYTE_ORDER==__LITTLE_ENDIAN...
Checking channels/chan_usbradio.c: DEBUG_CAPTURES==1...
Checking channels/chan_usbradio.c: DEBUG_CAPTURES==1XPMR_DEBUG0==1...

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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Re: [asterisk-users] meetme dahdi and zaptel

On Thu, Apr 02, 2009 at 10:22:55AM -0700, Dave Poirier wrote:
 We recently updated our Asterisk (1.4.24) box from Zaptel (1.4.12.1) to
 Dahdi (2.1.0.4). 

Have you rebuilt Asterisk after that?

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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Re: [asterisk-users] meetme dahdi and zaptel

2009-04-03 Thread Dave Poirier

Yes that was on a fresh build. I updated from zaptel to dahdi at the same
time as moving from Asterisk 1.4.22 to 1.4.24.


On Fri, Apr 3, 2009 at 11:23 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:

 On Thu, Apr 02, 2009 at 10:22:55AM -0700, Dave Poirier wrote:
  We recently updated our Asterisk (1.4.24) box from Zaptel (1.4.12.1) to
  Dahdi (2.1.0.4).

 Have you rebuilt Asterisk after that?

 --
   Tzafrir Cohen
 icq#16849755  
 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.co...@xorcom.com
 +972-50-7952406   mailto:tzafrir.co...@xorcom.com
 http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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-- 
David Poirier
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[asterisk-users] Using multiple 'peer' identities on one phone with 1.4

2009-04-03 Thread Florian Hackenberger

Hi!

When using multiple identities on one physical phone (Snom 320), I get

check_auth: username mismatch, have 7705, digest has 7736

messages when placing a call from a different account than the first 
one. From reading the asterisk source, I can see that the problem is 
that peer authentication is not matched against username, but against 
ip/port.
I need to have multiple queues a user can be logged in, therefore I need 
to limit calls to phones (otherwise an agent would get multiple calls 
at the same time). Because of the requirement for call limits I cannot 
use friends which do not play well with call limits (I can't remember 
the exact problem I ran into, but I think it is quite well known).

Is there a way to solve this issue?

Thanks in advance,
Florian
-- 
DI Florian Hackenberger
flor...@hackenberger.at
www.hackenberger.at

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[asterisk-users] Eicon Diva 2.01 PCI Passive BRI ISDN card

2009-04-03 Thread Puskás Zsolt

Hi Guys!

I got a Diva passive ISDN card and I can't get it work with asterisk 1.4,
It is supported in the kernel as an isdn4linux device but I can't find Modem 
channel type when i type in: core show channeltypes. I'm guessing it is 
removed in asterisk 1.4. Tried with capi interface but it does not work :(

Anybody got some idea how can i make it work or got a link to a working 
how-to?

Thank you.


pc:~# capiinfo
capi not installed - No such device or address (6)

pc:~# lspci -v
00:0b.0 Network controller: Dialogic Corporation Diva 2.01 S/T PCI (rev 01)
Subsystem: Dialogic Corporation Diva 2.01 S/T PCI
Flags: bus master, medium devsel, latency 0, IRQ 9
Memory at fedfb000 (32-bit, non-prefetchable) [size=4K]
Memory at fedfc000 (32-bit, non-prefetchable) [size=4K]
Capabilities: [40] Power Management version 1

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[asterisk-users] conference calling

2009-04-03 Thread Danny Nicholas

Greetings listers.

 I'm running asterisk 1.4.21.2 on SUSE 11.0 using
Polycom 501 phones.  My outgoing connections are Zapata using a TDM401P.
For the most part I can make and receive calls fine except for these 3
issues:

1.   When I call an external conference, the call never bridges and
hangs up after 60-90 seconds.

2.   When I call another number there is a 2-4 second delay before the
callee can hear me.

3.   When I call an external conference and connect, the others cannot
hear me.

 

Zapata.conf

[trunkgroups]

 

[channels]

;context=from-zaptel

;context=line1

busydetect=yes

callprogress=yes

busycount=4

hanguponpolarityswitch=yes

answeronpolarityswitch=yes

usecallingpres=yes

priindication=outofband

pritimer=t305,5

signalling=fxs_ks

wink=50

useincomingcalleridonzaptransfer=yes

echocancel=yes

echocancelwhenbridged=yes

faxdetect=yes

rxgain=1.0

txgain=21.0

callgroup=1

group=1

usecallerid=yes

callerid=asreceived

cidstart=ring

hidecallerid=no

immediate=no

pickupgroup=1

;context=incoming

channel = 1-4

 

Sip.conf

[general]

srvlookup=yes ;allows DNS lookups of server names

naxexpirey=180

defaultexpirey=160

context=default ; Default context for incoming calls

allowoverlap=no ; Disable overlap dialing support. (Default is yes)

bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)

tos_sip=cs3

tos_audio=ef

 

; bindport is the local UDP port that Asterisk will

; listen on

bindaddr=192.168.xx.xx ; IP address to bind to (0.0.0.0 binds to all)

srvlookup=yes ; Enable DNS SRV lookups on outbound calls

limitonpeers=yes

notifyringing=yes

rtupdate=yes[authentication]

 

[104]

type=peer

context=phones

host=dynamic

fromuser=104

secret=xx

canreinvite=update

directrtpsetup=no

call-limit=3

nat=yes

qualify=yes

register=no

session-timers=accept

session-expires=90

session-minse=120

session-refresher=uac

register = 104:xx...@xx.com/104

defaultip=192.168.xx.xxx

mailbox=104

disallow=all

allow=ulaw,alaw

artcachefriends=yes

notifyhold=yes

incominglimit=1

call-limit=3

 

Other information will be provided as asked for.  

 

Thanks in advance for any help you can provide.

 

Danny Nicholas

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Re: [asterisk-users] Eicon Diva 2.01 PCI Passive BRI ISDN card

2009-04-03 Thread Puskás Zsolt

Oooh and i forgot to mention:

OS: Debian 5.0 Lenny
Kernel: 2.6.29 ( self compiled )
Asterisk: 1.4.23.1 ( self compiled, no misdn, no zaptel )

On Friday 03 April 2009 21.39.24 Puskás Zsolt wrote:
 Hi Guys!

 I got a Diva passive ISDN card and I can't get it work with asterisk 1.4,
 It is supported in the kernel as an isdn4linux device but I can't find
 Modem channel type when i type in: core show channeltypes. I'm guessing
 it is removed in asterisk 1.4. Tried with capi interface but it does not
 work :(

 Anybody got some idea how can i make it work or got a link to a working
 how-to?

 Thank you.


 pc:~# capiinfo
 capi not installed - No such device or address (6)

 pc:~# lspci -v
 00:0b.0 Network controller: Dialogic Corporation Diva 2.01 S/T PCI (rev 01)
 Subsystem: Dialogic Corporation Diva 2.01 S/T PCI
 Flags: bus master, medium devsel, latency 0, IRQ 9
 Memory at fedfb000 (32-bit, non-prefetchable) [size=4K]
 Memory at fedfc000 (32-bit, non-prefetchable) [size=4K]
 Capabilities: [40] Power Management version 1



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Re: [asterisk-users] Eicon Diva 2.01 PCI Passive BRI ISDN card

2009-04-03 Thread Darwin Solano

I think you need to install Zaptel or newer verisons and the DIVA Dialogic
drivers, before to make it work with Asterisk. The CAPI issue will be solved
with the Zaptel and Linux Dialogic drivers.

hope it helps


On Fri, Apr 3, 2009 at 12:54 PM, Puskás Zsolt erro...@gmail.com wrote:

 Oooh and i forgot to mention:

 OS: Debian 5.0 Lenny
 Kernel: 2.6.29 ( self compiled )
 Asterisk: 1.4.23.1 ( self compiled, no misdn, no zaptel )

 On Friday 03 April 2009 21.39.24 Puskás Zsolt wrote:
  Hi Guys!
 
  I got a Diva passive ISDN card and I can't get it work with asterisk 1.4,
  It is supported in the kernel as an isdn4linux device but I can't find
  Modem channel type when i type in: core show channeltypes. I'm guessing
  it is removed in asterisk 1.4. Tried with capi interface but it does not
  work :(
 
  Anybody got some idea how can i make it work or got a link to a working
  how-to?
 
  Thank you.
 
 
  pc:~# capiinfo
  capi not installed - No such device or address (6)
 
  pc:~# lspci -v
  00:0b.0 Network controller: Dialogic Corporation Diva 2.01 S/T PCI (rev
 01)
  Subsystem: Dialogic Corporation Diva 2.01 S/T PCI
  Flags: bus master, medium devsel, latency 0, IRQ 9
  Memory at fedfb000 (32-bit, non-prefetchable) [size=4K]
  Memory at fedfc000 (32-bit, non-prefetchable) [size=4K]
  Capabilities: [40] Power Management version 1



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Re: [asterisk-users] SIP vs RTP destination IP

2009-04-03 Thread David Ruggles

Thx! That did it.

Thanks,

David Ruggles
CCNA MCSE (NT) CNA A+
Network EngineerSafe Data, Inc.
(910) 285-7200  da...@safedatausa.com



-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olle E.
Johansson
Sent: Friday, April 03, 2009 4:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP vs RTP destination IP



2 apr 2009 kl. 17.45 skrev David Ruggles:

 Is it possible to have asterisk override the connection information  
 embedded
 in a SIP 200 packet with the registration information? I have  
 multihomed
 machines with softphones and they register just fine and sip works  
 fine, but
 the RTP packets get sent to the ip from the SIP connection  
 information and
 the softphones are sending the wrong ip. I can't find an option in the
 softphone to change ip it sends.

If you turn on NAT support, we will ignore all IP addresses in the 200  
OK and
just send our media directly to wherever the other end sends it from.

/O

---
* Olle E. Johansson - o...@edvina.net
* Asterisk/OpenSER/Kamailio Training http://edvina.net/training/




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Re: [asterisk-users] meetme dahdi and zaptel

2009-04-03 Thread Carlos Chavez

Last time I upgraded Zaptel to DAHDI I had a similar problem until I
erased the zaptel modules.  The problem is that the Zaptel modules load
before DAHDI and you have a conflict with Asterisk.  Delete everything
from /lib/modules/(Kernel Version)/misc and then reinstall DAHDI.

On Fri, 2009-04-03 at 11:53 -0700, Dave Poirier wrote:
 Yes that was on a fresh build. I updated from zaptel to dahdi at the
 same time as moving from Asterisk 1.4.22 to 1.4.24.
 
 
 On Fri, Apr 3, 2009 at 11:23 AM, Tzafrir Cohen
 tzafrir.co...@xorcom.com wrote:
 On Thu, Apr 02, 2009 at 10:22:55AM -0700, Dave Poirier wrote:
  We recently updated our Asterisk (1.4.24) box from Zaptel
 (1.4.12.1) to
  Dahdi (2.1.0.4).
 
 
 Have you rebuilt Asterisk after that?
 
 --
   Tzafrir Cohen
 icq#16849755  jabber:tzafrir.co...@xorcom.com
 +972-50-7952406   mailto:tzafrir.co...@xorcom.com
 http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir
 
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 David Poirier
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-- 
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001


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Re: [asterisk-users] Radio interfaces for Asterisk - ISO image distro

2009-04-03 Thread John Todd


On Apr 3, 2009, at 11:19 AM, Tzafrir Cohen wrote:
 On Fri, Apr 03, 2009 at 10:43:08AM -0700, John Todd wrote:

 I just ran across these guys - looks very interesting:

  http://xelatec.com/xippr/install

 They distribute a self-installing ISO with Asterisk, FreePBX, and  
 some
 pre-built software to do radio over IP.  You'll need to buy the USB
 radio hardware, but it looks really interesting as a pre-built system
 for radio trunking using some of the Asterisk capabilities.  There is
 a very narrow but highly interested group of people who use radio
 interfaces for exotic locations such as oil platforms, ships, or
 remote off-grid locations to connect Asterisk systems together for
 long-haul telecommunications access.

 Hopefully there will be a speaker this year at Astricon who will be
 going into a very complex and unique system in detail using these
 components; I'll announce if/when they schedule their talk.

 Now, why would they need a separate ISO for that? What extra  
 software do
 they have?

 One change is obvious: chan_usbradio is not build by default.

 The output from cppcheck for it:
 [snip]

Couldn't say why they have a separate ISO other than guessing that it  
is a more painless process for radio people who don't really want to  
mess around with installing things from source.  I'm always interested  
in distributions using Asterisk in any form, even though we here on  
these lists might consider the differences trivial or possibly even  
detrimental.  This particular case seems to be for encouraging  
Asterisk with hardware not typically associated with telephony, which  
makes it additionally interesting.   When there are four or five of  
these distros floating around with usbradio support, I'll start  
ignoring them.  :-)

JT

---
John Todd   email:jt...@digium.com
Digium, Inc. | Asterisk Open Source Community Director
445 Jan Davis Drive NW -  Huntsville AL 35806  -   USA
direct: +1-256-428-6083 http://www.digium.com/




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Re: [asterisk-users] Bridging Avaya IP systems and Cisco IP system

2009-04-03 Thread John Todd


On Apr 3, 2009, at 7:40 AM, Gavin Henry wrote:

 Hi all,

 Has anyone put * in between an Avaya and Cisco system to connect two
 offices together?

 I was thinking about adding a SIP trunk on each side and getting
 Asterisk to pass calls between them. There is a leased line for
 bandwidth.

 Any tips/ideas on whether this is possible or dumb?

 Thanks.


Gavin -
   The short answer is yes, this is possible, and is done quite  
often.  How exactly you configure it is of course the trick - there  
are many possible different methods by which you might accomplish this  
feat, depending on what your existing resources are and what your end  
goal is.  T1? PRI? H.323?  You may consider IAX2 for trunking and save  
a lot of bandwidth as compared to SIP, if bandwidth is a concern.  If  
you're using T1 or PRI, you'll need a hardware card to do this.

   I'd start with setting up a basic Asterisk server from source and  
getting two SIP phones working on it.  I'd not suggest using one of  
the GUI-enabled versions - that may be more layers of stuff than   
you're looking for given your stated goal.  Figure it out, read the  
O'Reilly Book (Asterisk: The Future of Telephony) and you'll probably  
figure out fairly quickly how to use Asterisk as a black-box trunking  
interface for your systems.


JT

---
John Todd   email:jt...@digium.com
Digium, Inc. | Asterisk Open Source Community Director
445 Jan Davis Drive NW -  Huntsville AL 35806  -   USA
direct: +1-256-428-6083 http://www.digium.com/




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[asterisk-users] Grandstream surveillance devices

2009-04-03 Thread Jeff LaCoursiere


I just got a spam from telephonydepot (which I invited to spam me, so I 
guess I have to call it legit marketing :) ), and they have some new 
device that is meant to be a surveillance camera with audio, but the 
interface is POE and SIP!  A cool idea.  Anyone playing with this toy yet?

I am trying to wrap my head around how asterisk might fit with the model 
of this camera being recorded 24x7 with a continuous video/audio stream. 
Does asterisk even support recording video streams?

Cheers,

j

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Re: [asterisk-users] Grandstream surveillance devices

2009-04-03 Thread Tilghman Lesher

On Friday 03 April 2009 16:41:34 Jeff LaCoursiere wrote:
 I just got a spam from telephonydepot (which I invited to spam me, so I
 guess I have to call it legit marketing :) ), and they have some new
 device that is meant to be a surveillance camera with audio, but the
 interface is POE and SIP!  A cool idea.  Anyone playing with this toy yet?

 I am trying to wrap my head around how asterisk might fit with the model
 of this camera being recorded 24x7 with a continuous video/audio stream.
 Does asterisk even support recording video streams?

Yes, Asterisk supports recording video streams, although the format might be
a little weird.  Most of the recorded video formats are merely frame dumps,
meant only for Asterisk to requeue to another destination.  They are not
generally useful otherwise.  I'm also not sure how well it'd work with
continuous streams.  There's also no support for transcoding of video, so what
is streamed is what is kept.

-- 
Tilghman

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Re: [asterisk-users] Grandstream surveillance devices

2009-04-03 Thread Cary Fitch

Me too.

Would it be you can dial it like a phone and get the video/audio from the?
Co-advertised phone?

We have an alarm system. Perhaps on alarm, we could dial and see the
premises?

Or the popular day care cam use.

Or any dial and see application?

Cary


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
LaCoursiere
Sent: Friday, April 03, 2009 4:42 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Grandstream surveillance devices


I just got a spam from telephonydepot (which I invited to spam me, so I 
guess I have to call it legit marketing :) ), and they have some new 
device that is meant to be a surveillance camera with audio, but the 
interface is POE and SIP!  A cool idea.  Anyone playing with this toy yet?

I am trying to wrap my head around how asterisk might fit with the model 
of this camera being recorded 24x7 with a continuous video/audio stream. 
Does asterisk even support recording video streams?

Cheers,

j

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Re: [asterisk-users] Radio interfaces for Asterisk - ISO image distro

On Fri, Apr 03, 2009 at 02:25:53PM -0700, John Todd wrote:

 Couldn't say why they have a separate ISO other than guessing that it  
 is a more painless process for radio people who don't really want to  
 mess around with installing things from source.  

I actually wondered why they built their own separate distro and not
used another one.

And why are other distributions not good enough for radio people. 

I wonder what the real bug is.

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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Re: [asterisk-users] meetme dahdi and zaptel

Hi did list his lsmod and it doesn't show dahdi modules ...
For me it seems to be that dahdichanname=no ...

Martin

On Fri, Apr 3, 2009 at 4:17 PM, Carlos Chavez cur...@telecomabmex.com wrote:
        Last time I upgraded Zaptel to DAHDI I had a similar problem until I
 erased the zaptel modules.  The problem is that the Zaptel modules load
 before DAHDI and you have a conflict with Asterisk.  Delete everything
 from /lib/modules/(Kernel Version)/misc and then reinstall DAHDI.

 On Fri, 2009-04-03 at 11:53 -0700, Dave Poirier wrote:
 Yes that was on a fresh build. I updated from zaptel to dahdi at the
 same time as moving from Asterisk 1.4.22 to 1.4.24.


 On Fri, Apr 3, 2009 at 11:23 AM, Tzafrir Cohen
 tzafrir.co...@xorcom.com wrote:
         On Thu, Apr 02, 2009 at 10:22:55AM -0700, Dave Poirier wrote:
          We recently updated our Asterisk (1.4.24) box from Zaptel
         (1.4.12.1) to
          Dahdi (2.1.0.4).


         Have you rebuilt Asterisk after that?

         --
                       Tzafrir Cohen
         icq#16849755              jabber:tzafrir.co...@xorcom.com
         +972-50-7952406           mailto:tzafrir.co...@xorcom.com
         http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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 Carlos Chávez Prats
 Director de Tecnología
 +52-55-91169161 ext 2001

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Re: [asterisk-users] meetme dahdi and zaptel

I meant zaptel modules ... no zaptel modules loaded on his system

Martin

On Fri, Apr 3, 2009 at 5:28 PM, Martin asteriskl...@callthem.info wrote:
 Hi did list his lsmod and it doesn't show dahdi modules ...
 For me it seems to be that dahdichanname=no ...

 Martin

 On Fri, Apr 3, 2009 at 4:17 PM, Carlos Chavez cur...@telecomabmex.com wrote:
        Last time I upgraded Zaptel to DAHDI I had a similar problem until I
 erased the zaptel modules.  The problem is that the Zaptel modules load
 before DAHDI and you have a conflict with Asterisk.  Delete everything
 from /lib/modules/(Kernel Version)/misc and then reinstall DAHDI.

 On Fri, 2009-04-03 at 11:53 -0700, Dave Poirier wrote:
 Yes that was on a fresh build. I updated from zaptel to dahdi at the
 same time as moving from Asterisk 1.4.22 to 1.4.24.


 On Fri, Apr 3, 2009 at 11:23 AM, Tzafrir Cohen
 tzafrir.co...@xorcom.com wrote:
         On Thu, Apr 02, 2009 at 10:22:55AM -0700, Dave Poirier wrote:
          We recently updated our Asterisk (1.4.24) box from Zaptel
         (1.4.12.1) to
          Dahdi (2.1.0.4).


         Have you rebuilt Asterisk after that?

         --
                       Tzafrir Cohen
         icq#16849755              jabber:tzafrir.co...@xorcom.com
         +972-50-7952406           mailto:tzafrir.co...@xorcom.com
         http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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Re: [asterisk-users] Bridging Avaya IP systems and Cisco IP system

2009-04-03 Thread Gavin Henry

BTW, what's the recommended production version of Asterisk source
you'd recommend, the latest 1.4 or 1.6?

In fact, nevermind. This is asked so many times I'll hit the archives.

Cheers.

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Re: [asterisk-users] Bridging Avaya IP systems and Cisco IP system

2009-04-03 Thread Gavin Henry

2009/4/3 John Todd jt...@digium.com:

 On Apr 3, 2009, at 7:40 AM, Gavin Henry wrote:

 Hi all,

 Has anyone put * in between an Avaya and Cisco system to connect two
 offices together?

 I was thinking about adding a SIP trunk on each side and getting
 Asterisk to pass calls between them. There is a leased line for
 bandwidth.

 Any tips/ideas on whether this is possible or dumb?

 Thanks.


 Gavin -
   The short answer is yes, this is possible, and is done quite
 often.  How exactly you configure it is of course the trick - there
 are many possible different methods by which you might accomplish this
 feat, depending on what your existing resources are and what your end
 goal is.  T1? PRI? H.323?  You may consider IAX2 for trunking and save
 a lot of bandwidth as compared to SIP, if bandwidth is a concern.  If
 you're using T1 or PRI, you'll need a hardware card to do this.

   I'd start with setting up a basic Asterisk server from source and
 getting two SIP phones working on it.  I'd not suggest using one of
 the GUI-enabled versions - that may be more layers of stuff than
 you're looking for given your stated goal.  Figure it out, read the
 O'Reilly Book (Asterisk: The Future of Telephony) and you'll probably
 figure out fairly quickly how to use Asterisk as a black-box trunking
 interface for your systems.

Thanks John. Yeah, we've done this for an Avaya system already using
H.323 and we can
just add a sip trunk to the CCM and do dialplans accordingly. Just
need to get some specs on
what each side is from the client.

We could put a simple box on each side and use IAX2 trunking, sure.

It's simple and I should have thought it through before posting ;-)

Cheers John.

Gavin.

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[asterisk-users] Live Support function?

2009-04-03 Thread Dean Collins

Hi guys,

 

I'd like to add a LIVE SUPPORT function to my website. 

 

Basically I want a client on my desktop that pops up when someone
request help BUT doesn't appear or says offline when I'm not available
or have logged out of this function.

 

When a person visiting my website has a question they hot the button to
cause a text popup chat to occur.

Anyone know of an open source solution? I know there are plenty of
commercial hosted options available for a monthly fee but seems like
such a simple requirement that something has to be available (especially
as I'm only looking for one support client and no need to round robin or
multiple agent support or agent cut and paste functions etc).

 

 

Just need basic text chat function - the terms I'm googling don't seem
to be bringing anything up.

 

(needs to be linux on the server end)

 

 

Regards,

Dean Collins
Cognation Inc
d...@cognation.net
mailto:d...@cognation.net +1-212-203-4357   New York
+61-2-9016-5642   (Sydney in-dial).
+44-20-3129-6001 (London in-dial).

 

 

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Re: [asterisk-users] Radio interfaces for Asterisk - ISO image distro

2009-04-03 Thread Steve Totaro

On Fri, Apr 3, 2009 at 6:23 PM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
 On Fri, Apr 03, 2009 at 02:25:53PM -0700, John Todd wrote:

 Couldn't say why they have a separate ISO other than guessing that it
 is a more painless process for radio people who don't really want to
 mess around with installing things from source.

 I actually wondered why they built their own separate distro and not
 used another one.

 And why are other distributions not good enough for radio people.

 I wonder what the real bug is.

 --
               Tzafrir Cohen
 icq#16849755              jabber:tzafrir.co...@xorcom.com
 +972-50-7952406           mailto:tzafrir.co...@xorcom.com
 http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir


I have fought this battle already.  It is what it is.  Apparently me
fighting the good fight has been removed from the tubes though..

From Here:  http://app-rpt.qrvc.com/faq.html

app_rpt is an application which comes bundled with Asterisk, however,
a later version may be available on our source repository. All you
need to do is go to asterisk.org, download asterisk, configure the
asterisk to build app_rpt by modifying the Makefile in the
asterisk/apps directory, and then compile and install it.  You can get
the latest version of app_rpt.c along with the sound files, and sample
configuration files from our repository at:
https://xelatec.com/viewvc/app_rpt 

As this is a small niche, all of six or seven people at this track
during Astricon 07, I am not surprised you didn't realize they have
already spoke at Astricon, nor am I really surprised that they have
their own repo.

I did get to meet Jim Dixon and Steven Rogers though, which was cool.

-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)

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Re: [asterisk-users] Live Support function?

2009-04-03 Thread Dean Collins

http://openwebim.org http://openwebim.org/ 

 

anyone using this one (was just emailed it from another channel) -
should have waited more than 5 mins before posting twice.

 

 

 

 

 

Regards,

Dean Collins
Cognation Inc
d...@cognation.net
mailto:d...@cognation.net +1-212-203-4357   New York
+61-2-9016-5642   (Sydney in-dial).
+44-20-3129-6001 (London in-dial).

 

 



From: Dean Collins 
Sent: Friday, April 03, 2009 7:27 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Live Support function?

 

Hi guys,

 

I'd like to add a LIVE SUPPORT function to my website. 

 

Basically I want a client on my desktop that pops up when someone
request help BUT doesn't appear or says offline when I'm not available
or have logged out of this function.

 

When a person visiting my website has a question they hot the button to
cause a text popup chat to occur.

Anyone know of an open source solution? I know there are plenty of
commercial hosted options available for a monthly fee but seems like
such a simple requirement that something has to be available (especially
as I'm only looking for one support client and no need to round robin or
multiple agent support or agent cut and paste functions etc).

 

 

Just need basic text chat function - the terms I'm googling don't seem
to be bringing anything up.

 

(needs to be linux on the server end)

 

 

Regards,

Dean Collins
Cognation Inc
d...@cognation.net
mailto:d...@cognation.net +1-212-203-4357   New York
+61-2-9016-5642   (Sydney in-dial).
+44-20-3129-6001 (London in-dial).

 

 

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Re: [asterisk-users] Radio interfaces for Asterisk - ISO image distro

2009-04-03 Thread Steve Totaro

On Fri, Apr 3, 2009 at 7:29 PM, Steve Totaro
stot...@asteriskhelpdesk.com wrote:
 On Fri, Apr 3, 2009 at 6:23 PM, Tzafrir Cohen tzafrir.co...@xorcom.com 
 wrote:
 On Fri, Apr 03, 2009 at 02:25:53PM -0700, John Todd wrote:

 Couldn't say why they have a separate ISO other than guessing that it
 is a more painless process for radio people who don't really want to
 mess around with installing things from source.

 I actually wondered why they built their own separate distro and not
 used another one.

 And why are other distributions not good enough for radio people.

 I wonder what the real bug is.

 --
               Tzafrir Cohen
 icq#16849755              jabber:tzafrir.co...@xorcom.com
 +972-50-7952406           mailto:tzafrir.co...@xorcom.com
 http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir


 I have fought this battle already.  It is what it is.  Apparently me
 fighting the good fight has been removed from the tubes though..

 From Here:  http://app-rpt.qrvc.com/faq.html

 app_rpt is an application which comes bundled with Asterisk, however,
 a later version may be available on our source repository. All you
 need to do is go to asterisk.org, download asterisk, configure the
 asterisk to build app_rpt by modifying the Makefile in the
 asterisk/apps directory, and then compile and install it.  You can get
 the latest version of app_rpt.c along with the sound files, and sample
 configuration files from our repository at:
 https://xelatec.com/viewvc/app_rpt 

 As this is a small niche, all of six or seven people at this track
 during Astricon 07, I am not surprised you didn't realize they have
 already spoke at Astricon, nor am I really surprised that they have
 their own repo.

 I did get to meet Jim Dixon and Steven Rogers though, which was cool.

 --
 Thanks,
 Steve Totaro
 +18887771888 (Toll Free)
 +12409381212 (Cell)


Correction, Steven Henke, apologies, a very helpful and downright nice guy.

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Re: [asterisk-users] conference calling

Turn off callprogres=yes or have it configured properly.
It should fix your problem.

regards
Martin

On Fri, Apr 3, 2009 at 2:42 PM, Danny Nicholas da...@debsinc.com wrote:
 Greetings listers.

  I’m running asterisk 1.4.21.2 on SUSE 11.0 using
 Polycom 501 phones.  My outgoing connections are Zapata using a TDM401P.
 For the most part I can make and receive calls fine except for these 3
 issues:

 1.   When I call an external conference, the call never bridges and
 hangs up after 60-90 seconds.

 2.   When I call another number there is a 2-4 second delay before the
 callee can hear me.

 3.   When I call an external conference and connect, the others cannot
 hear me.



 Zapata.conf

 [trunkgroups]



 [channels]

 ;context=from-zaptel

 ;context=line1

 busydetect=yes

 callprogress=yes

 busycount=4

 hanguponpolarityswitch=yes

 answeronpolarityswitch=yes

 usecallingpres=yes

 priindication=outofband

 pritimer=t305,5

 signalling=fxs_ks

 wink=50

 useincomingcalleridonzaptransfer=yes

 echocancel=yes

 echocancelwhenbridged=yes

 faxdetect=yes

 rxgain=1.0

 txgain=21.0

 callgroup=1

 group=1

 usecallerid=yes

 callerid=asreceived

 cidstart=ring

 hidecallerid=no

 immediate=no

 pickupgroup=1

 ;context=incoming

 channel = 1-4



 Sip.conf

 [general]

 srvlookup=yes ;allows DNS lookups of server names

 naxexpirey=180

 defaultexpirey=160

 context=default ; Default context for incoming calls

 allowoverlap=no ; Disable overlap dialing support. (Default is yes)

 bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)

 tos_sip=cs3

 tos_audio=ef



 ; bindport is the local UDP port that Asterisk will

 ; listen on

 bindaddr=192.168.xx.xx ; IP address to bind to (0.0.0.0 binds to all)

 srvlookup=yes ; Enable DNS SRV lookups on outbound calls

 limitonpeers=yes

 notifyringing=yes

 rtupdate=yes[authentication]



 [104]

 type=peer

 context=phones

 host=dynamic

 fromuser=104

 secret=xx

 canreinvite=update

 directrtpsetup=no

 call-limit=3

 nat=yes

 qualify=yes

 register=no

 session-timers=accept

 session-expires=90

 session-minse=120

 session-refresher=uac

 register = 104:xx...@xx.com/104

 defaultip=192.168.xx.xxx

 mailbox=104

 disallow=all

 allow=ulaw,alaw

 artcachefriends=yes

 notifyhold=yes

 incominglimit=1

 call-limit=3



 Other information will be provided as asked for.



 Thanks in advance for any help you can provide.



 Danny Nicholas

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Re: [asterisk-users] Using multiple 'peer' identities on one phone with 1.4

Hi

On Fri, Apr 3, 2009 at 1:57 PM, Florian Hackenberger
f.hackenber...@chello.at wrote:
 Hi!

 When using multiple identities on one physical phone (Snom 320), I get

 check_auth: username mismatch, have 7705, digest has 7736
The SNOM evidently has a bug. When it originates the call as user 7705
then it should also authenticate
as user 7736. Asterisk doesn't like it. You'd have to patch your
asterisk to remove that check.


 messages when placing a call from a different account than the first
 one. From reading the asterisk source, I can see that the problem is
 that peer authentication is not matched against username, but against
 ip/port.
It's matched against both IHMO in some order of priority.

Martin

 I need to have multiple queues a user can be logged in, therefore I need
 to limit calls to phones (otherwise an agent would get multiple calls
 at the same time). Because of the requirement for call limits I cannot
 use friends which do not play well with call limits (I can't remember
 the exact problem I ran into, but I think it is quite well known).

 Is there a way to solve this issue?

 Thanks in advance,
        Florian
 --
 DI Florian Hackenberger
 flor...@hackenberger.at
 www.hackenberger.at

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Re: [asterisk-users] SIP Warnning Message

You're trying to register with the service but Asterisk is using the
default expiry value of 120 seconds (1.6.x version)
And your provider wants you to use minimum of 3600 seconds (1 hr)

add defaultexpiry=3600 to [general] section of sip.conf

That should help register...

Martin

On Fri, Apr 3, 2009 at 12:45 PM, César García cel...@gmail.com wrote:
 Guys, when registering I am getting this error message, my question is
 that if this could be the reason whay I am able to make calls but not
 to recieve call ?


 [Apr  3 11:24:31] WARNING[19578]: chan_sip.c:15104
 handle_response_register: Got 423 Interval too brief for service
 +506phonenum...@domain.co.cr@host.ip.addr, minimum is 3600 seconds


 Thanks


 --
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Re: [asterisk-users] ISDN Timer T309

What is the specification for T309 ? I'm too lazy to look it up.

The default behaviour when the alarm of layer 1 (electrical T1/E1) is
detected is to assume
all calls dropped on both sides and that's what Asterisk does.

The timer is simply deactivated since all the calls are supposed to
drop. I believe that agrees with Q921/Q931 specs.

Martin

On Fri, Apr 3, 2009 at 12:14 PM, Afonso Zimmermann afo...@disc-os.org wrote:
 Hi everione,

 I'm make some test with pri timer T309. I'm using asterisk 1.4.23.1, libpri
 1.4.9, dahdi-linux 2.1.0.4 and dahdi-tools 2.1.0.2. But in my tests, the
 timer fail with a telco link in this scenario:

 Telco Phone -- Telco --- Asterisk  Sip
 Phone

 When i make a call from Telco Phone to Sip Phone, the call complete, but
 when i disconnect the link and reconnect in few seconds, the Asterisk clear
 call:

 [Apr  3 10:44:40] WARNING[13081]: chan_dahdi.c:4385 handle_alarms: Detected
 alarm on channel 1: Red Alarm
 [Apr  3 10:44:40] NOTICE[21088]: chan_dahdi.c:9558 pri_dchannel: PRI got
 event: Alarm (4) on Primary D-channel of span 1
   == Spawn extension (disc-from-trunk-TR001, 9800, 2) exited non-zero on
 'DAHDI/1-1'
 [Apr  3 10:44:40] WARNING[21088]: chan_dahdi.c:3021 pri_find_dchan: No
 D-channels available!  Using Primary channel 16 as D-channel anyway!
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
 alarm on channel 2: Red Alarm
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
 to disable echo cancellation on channel 2: Invalid argument
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
 alarm on channel 3: Red Alarm
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
 to disable echo cancellation on channel 3: Invalid argument
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
 alarm on channel 4: Red Alarm
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
 to disable echo cancellation on channel 4: Invalid argument
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
 alarm on channel 5: Red Alarm
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
 to disable echo cancellation on channel 5: Invalid argument
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
 alarm on channel 6: Red Alarm
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
 to disable echo cancellation on channel 6: Invalid argument
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
 alarm on channel 7: Red Alarm
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
 to disable echo cancellation on channel 7: Invalid argument
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
 alarm on channel 8: Red Alarm
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
 to disable echo cancellation on channel 8: Invalid argument
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
 alarm on channel 9: Red Alarm
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
 to disable echo cancellation on channel 9: Invalid argument
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
 alarm on channel 10: Red Alarm
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
 to disable echo cancellation on channel 10: Invalid argument
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
 alarm on channel 11: Red Alarm
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
 to disable echo cancellation on channel 11: Invalid argument
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
 alarm on channel 12: Red Alarm
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
 to disable echo cancellation on channel 12: Invalid argument
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
 alarm on channel 13: Red Alarm
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
 to disable echo cancellation on channel 13: Invalid argument
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
 alarm on channel 14: Red Alarm
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
 to disable echo cancellation on channel 14: Invalid argument
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
 alarm on channel 15: Red Alarm
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
 to disable echo cancellation on channel 15: Invalid argument
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:4385 handle_alarms: Detected
 alarm on channel 17: Red Alarm
 [Apr  3 10:44:40] WARNING[21089]: chan_dahdi.c:2012 dahdi_disable_ec: Unable
 to

Re: [asterisk-users] opermode=?

if you use zaptel then it seems you can pass opermode as the argument
to modprobe or hardcode it in /etc/modprobe.conf
it depends though what card you have since wctdm uses a character
opermode while wcfxo doesn't support that at all (it has its
own opermode definitions which are incomplete)

Martin

On Fri, Apr 3, 2009 at 12:19 PM, bilal ghayyad bilmar...@yahoo.com wrote:

 Thanks Tzafrir;

 But did not get where to find drivers? I have zaptel.

  Hi All;
 
  If I need to set the opermode to King Saudi Arabia,
 what the name I
  have to use? For example, to set it for kuwait then I
 use
  opermode=KUWAIT. So what will be for Saudi Arabia?

 $ grep -i saudi drivers/dahdi/fxo_modes.h
         { .name = SAUDIARABIA,

 --
                Tzafrir Cohen
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Re: [asterisk-users] Radio interfaces for Asterisk - ISO image distro