Re: [asterisk-users] Audiocodes MP-114 noise

[Jason Kim]

Thank you Jessee,

Firmware seems to be recent(4.80A.025.004).
For 'noisy', I mean IP Phone <--> * <--> MP-114 side.
Audio quality of MP-114 <--> PSTN <--> Analog phone is
good.
I think it can be power ground or gain problem.
Any experience?

Thanks,
Jason

--- Jessee J Holmes <[EMAIL PROTECTED]> wrote:

> Dear Jason,
> 
> Please define better noisy? You talking echo issues?
> Is it on just  
> your side or on the called party's side as well?
> This start happening immediately, or was the box
> working before and  
> the problem just started?
> 
> Also, a quick heads up, make sure before even
> beginning to  
> troubleshoot an issue like this you do a factory
> reset to the unit  
> and get the latest available firmware on it. Usually
> that fixes  
> annoying issues like this.
> 
> Thanks,
> 
> 
> Jessee Holmes
> Atacomm / Ataractic Corporation
> www.atacomm.com
> V: 1-877-700-VOIP
> [EMAIL PROTECTED]
> 
> Looking for voice over IP products?  Visit our VoIP
> store at http:// 
> voipstore.atacomm.com/
> 
> 
> On Oct 30, 2006, at 10:36 PM, Jason Kim wrote:
> 
> > It's noisy while talking.
> > Any idea?
> >
> > Thanks in advance.
> > Jason
> >
> >
> >
> >
>
__
> 
> > __
> > Cheap Talk? Check out Yahoo! Messenger's low
> PC-to-Phone call rates
> > (http://voice.yahoo.com)
> >
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Low, Low, Low Rates! Check out Yahoo! Messenger's cheap PC-to-Phone call rates 
(http://voice.yahoo.com)

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Re: [asterisk-users] S(x) - Hang up the call after 'x' seconds - Not working from queue

[Rajkumar S]

On 11/1/06, Leo Ann Boon <[EMAIL PROTECTED]> wrote:

Rajkumar S wrote:
> On 10/31/06, Leo Ann Boon <[EMAIL PROTECTED]> wrote:
>> Someone correct me if I'm wrong: The Dial string is missing a '/n'
>> parameter for the Local channel. Without /n, Asterisk will do a native
>> transfer to SIP/1001 and lose the timeout value defined earlier.
>
> What does '/n' refer here? There is no mention about this in the wiki.
>
It's in the wiki, see this:
http://www.voip-info.org/wiki/index.php?page=Asterisk+local+channels


Thanks Leo.

I went though the code of the app_queue to find out if the cutoff
value I gave in the dialplan is indeed being passed though when
bridging is happening and it's not. The actual line where bridging is
happening is

bridge = ast_bridge_call(qe->chan,peer, &bridge_config);

The bridge_config is of type ast_bridge_config and holds the options
to use for this bridging and it has a field called timelimit, which
holds the timelimit of the call. This variable is not set in
app_queue. This is the reason why the timelimit was not working when
called from queue.

I edited  the code and put a sample value (in milliseconds) and the
call cutoff is working fine. I am not sure if this introduces any side
effects, but it's so far so good.

Another advantage of this method is that the call cutoff will work
only when the call is bridged from queue and not from directly called
calls.

Thanks for your help, Leo and Lenz.

raj
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[asterisk-users] wrong password on authentication for INVITE

[Klaverstyn, David C]

I have a Snom 360 phone that will not work on an Asterisk
server but it will on another server.  This phone has been working for
over 4 months or so.  I can not figure it out.  This is the only Snom
phone that I have so I can check it against another one.  The PBX that
fails, fails with any extension number.  Replacing the phone with a SPA
has no problems.

 

I don’t understand how it can work on one server but
not another.

 

The error that is occurring is:

Nov  1 14:31:42 WARNING[32190]: chan_sip.c:9720
handle_response_invite: Forbidden - wrong password on authentication for INVITE
to '"DavidR (Perth)"
sip:[EMAIL PROTECTED]>;tag=as11bbecc0'

 

Please help.






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[asterisk-users] wrong password on authentication for INVITE

[Klaverstyn, David C]

I have a Snom 360 phone that will not work on an Asterisk
server but it will on another server.  This phone has been working for over 4
months or so.  I can not figure it out.  This is the only Snom phone that I
have so I can check it against another one.  The PBX that fails, fails with any
extension number.  Replacing the phone with a SPA has no problems.

 

I don’t understand how it can work on one server but
not another.

 

The error that is occurring is:

Nov  1 14:31:42 WARNING[32190]: chan_sip.c:9720
handle_response_invite: Forbidden - wrong password on authentication for INVITE
to '"DavidR (Perth)"
sip:[EMAIL PROTECTED]>;tag=as11bbecc0'

 

Please help.






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Re: [asterisk-users] S(x) - Hang up the call after 'x' seconds - Not working from queue

[Leo Ann Boon]

Rajkumar S wrote:

On 10/31/06, Leo Ann Boon <[EMAIL PROTECTED]> wrote:

Someone correct me if I'm wrong: The Dial string is missing a '/n'
parameter for the Local channel. Without /n, Asterisk will do a native
transfer to SIP/1001 and lose the timeout value defined earlier.


What does '/n' refer here? There is no mention about this in the wiki.


It's in the wiki, see this:
http://www.voip-info.org/wiki/index.php?page=Asterisk+local+channels

Leo

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Re: [asterisk-users] Re: Opinions on the best wholesale origination/term providers

[Brad Templeton]

On Tue, Oct 31, 2006 at 08:03:56PM -0800, Martin Joseph wrote:
> On 2006-10-31 17:29:47 -0800, Brad Templeton <[EMAIL PROTECTED]> 
> said:
> 
> >
> >I've been losing patience with my current provider, a small company
> >called Sellvoip.  Their termination is good, and they are
> >asterisk based, but they are understaffed and have no concept
> >of customer service.  So I'm shopping.
> I also use Sellvoip and I am close to them (Seattle).  They by FAR 
> produce the best call quality for me, when compared to nufone and 
> Teliax, although both of those companies do ok, my routes to them 
> aren't nearly as clean.
> 
> I recommend Teliax for good support.
> 

Their DIDs ($5/month plus 2 cents/minute) are much too high,
their termination is 2 cents which is tolerable but in general
too high for a wholesale service.  But thanks for the comment.

The sellvoip guys (guy?) are indeed producing good quality.  Another
thing they are doing, which I really like, is processing
termination quickly, in that when I do the invite it's ringing
within a fraction of a second.   A few other termination providers
I have tried are taking 3-4 seconds to ring after invite.

You thought I wrote a lot and I didn't even put that on
my list.

We just have to convince Jed at Sellvoip to hire some
some support techs, even if he has to add a couple of tenths
per minute.
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Re: [asterisk-users] Asterisk both behind a NAT and outside at the same time

[Brad Templeton]

On Wed, Nov 01, 2006 at 08:10:29AM +0800, Leo Ann Boon wrote:
> Brad Templeton wrote:
> >  
> The way I understand it, externalip and localnet work hand-in-hand. I do 
> agree with you that this is commonly used for Asterisk behind a NAT. I 
> believe these parameter just helps asterisk determine what to do. In 
> your case, you don't lose anything - the external IP would still have to 
> be written into every outbound packet.

It's called externip I think, not externalip.  I have set both externip
to be my external IP address, and localnet to be the natwork, and even
set canreinvite=no and nat=yes and the SDP I get back from an invite to
[EMAIL PROTECTED]  still has 192.186.* in it.


> 
> >It uses bindaddr=0.0.0.0 and listens to both addresses.  
> >  
> externalip doesn't affect the bindaddr.

Would not expect it to.  Just trying to be clear to people that
the machine has two ethernets.   I was hoping Asterisk would
just automatically say, "Wait a minute, I'm taking an SDP with
addresses in the localnet, and sending it out to a peer on the
outside internet.  That's not going to work!"   


Now one of my tests has a SIP program I have written attempt to
call Asterisk.  It sits on port 5061 invites to Asterisk on 5160
of the machine with the external address as follows:


INVITE sip:[EMAIL PROTECTED]:5160;transport=udp SIP/2.0^M
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE^M
From: "Voxable" ;tag=3445^M
To: "Party Leg1" ^M
Via: SIP/2.0/UDP 
198.144.201.82:5061;branch=z9hG4bK6563eba5fd430b5af93579617a44450e^M
Max-Forwards: 12^M
Contact: "Caller App" ^M
Date: Wed, 01 Nov 2006 04:52:07 GMT^M
User-Agent: Voxable 0.1^M
Content-Type: application/sdp^M
Content-Length: 154^M
^M
v=0
o=capp 1162356727 1 IN IP4 198.144.201.82
s=CApp3PCC
c=IN IP4 198.144.201.82
t=0 0
m=audio 5308 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PMCA/8000

Asterisk sends this on to the phone, but rewrites the SDP
to present a local one:
o=root 26391 26391 IN IP4 192.168.123.10
s=session
c=IN IP4 192.168.123.10
t=0 0
m=audio 10856 RTP/AVP 0 97 8 101
 

I answer the phone and it responds to this SDP with an OK

o=brad 8000 8000 IN IP4 192.168.123.18
s=SIP Call
c=IN IP4 192.168.123.18
t=0 0
m=audio 5004 RTP/AVP 0 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11 
 


Asterisk then sends back this SDP without rewriting it.
Is it only doing that because it knows the traffic came from
the same machine?

Asterisk forwards this OK back.  Note the SDP

SIP/2.0 200 OK^M
Via: SIP/2.0/UDP 
198.144.201.82:5061;branch=z9hG4bK94e379b15dbd22f8594fa6e88a4cfcc0;received=198.144.201.82^M
From: "Voxable" ;tag=3445^M
To: "Party Leg1" ;tag=as62de8d32^M
Call-ID: [EMAIL PROTECTED]
CSeq: 1 INVITE^M
User-Agent: Caller Asterisk^M
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY^M
Supported: replaces^M
Contact: ^M
Content-Type: application/sdp^M
Content-Length: 199^M
^M
v=0^M
o=root 26391 26391 IN IP4 192.168.123.18^M
s=session^M
c=IN IP4 192.168.123.18^M
t=0 0^M
m=audio 5004 RTP/AVP 0 8^M
a=rtpmap:0 PCMU/8000^M
a=rtpmap:8 PCMA/8000^M
a=silenceSupp:off - - - -^M
a=sendrecv^M

My software then forwards that SDP on to an outside location, where the 
SDP is useless.

It works if the outside provider I forward the SDP to has my asterisk
box set with some flags (nat=yes I presume?) though I can't figure why.
That box is presumably, seeing the internal address, routing the
audio to some port on the * box, and asterisk is forwarding it but I
can't see how this is happening.  Odd.
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Re: [asterisk-users] Example Polycom function key config

[Noah Miller]

Hi Jamie -


Has anyone here reprogrammed their Polycom features keys using
sip/ipmid.cfg?

If so I would be really grateful if someone could send me an
example


Here's the "keys" line that I use for one of my clients:



I've used this with all versions of the firmware since 1.4.1.

- Noah



On 10/31/06, Jamie Heckford <[EMAIL PROTECTED]> wrote:


Hi,

Has anyone here reprogrammed their Polycom features keys using
sip/ipmid.cfg?

If so I would be really grateful if someone could send me an example as
I have tried various entries for hours now and don't seem to be getting
anywhere.

Any help appreciated.


Kind regards

Jamie Heckford
Technical Consultant


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Re: [asterisk-users] S(x) - Hang up the call after 'x' seconds - Not working from queue

[Rajkumar S]

On 10/31/06, Leo Ann Boon <[EMAIL PROTECTED]> wrote:

Rajkumar S wrote:
>
>-- Executing Queue("SIP/1002-74e9", "Auth-Enq|t") in new stack
>-- Started music on hold, class 'default', on channel 'SIP/1002-74e9'
>-- outgoing agentcall, to agent '1001', on
> 'Local/[EMAIL PROTECTED],1'
>-- Executing Dial("Local/[EMAIL PROTECTED],2",
> "SIP/1001||tS(30)") in new stack
>-- Setting call duration limit to 30 seconds.
>-- Called 1001
>-- Called Agent/1001
>-- SIP/1001-d43c is ringing
>-- Agent/1001 is ringing
>-- SIP/1001-d43c answered Local/[EMAIL PROTECTED],2
>-- Agent/1001 answered SIP/1002-74e9
>-- Stopped music on hold on SIP/1002-74e9
>  == Spawn extension (from-sip, 1001, 1) exited non-zero on
> 'Local/[EMAIL PROTECTED],2'
>  == Spawn extension (from-sip, 99, 1) exited non-zero on 'SIP/1002-74e9'
>
Someone correct me if I'm wrong: The Dial string is missing a '/n'
parameter for the Local channel. Without /n, Asterisk will do a native
transfer to SIP/1001 and lose the timeout value defined earlier.


What does '/n' refer here? There is no mention about this in the wiki.

raj
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[asterisk-users] Architecture for Asterisk

[jezzzz .]

Thank you for you responses re: my question on the architecture of Asterisk. Olle, your explanation was especially useful. I still feel like I'm missing a crucial part here. If Asterisk is an endpoint, and according to the example you gave in your response (if u1 hangs up then Asterisk decides to hang up with u2) it is then for a 'bye' message, where would Asterisk come in? Would u1 send a bye message to Asterisk who would then send that message to u2? I'm still a little puzzled as to where it comes in the picture when we're talking about the flow of messages between two users. Perhaps more specifically, why is Asterisk required at all? Is it just to locate users?
 
Thank you so much
 
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[asterisk-users] Re: Opinions on the best wholesale origination/term providers

[Martin Joseph]

On 2006-10-31 17:29:47 -0800, Brad Templeton <[EMAIL PROTECTED]> said:



I've been losing patience with my current provider, a small company
called Sellvoip.  Their termination is good, and they are
asterisk based, but they are understaffed and have no concept
of customer service.  So I'm shopping.
I also use Sellvoip and I am close to them (Seattle).  They by FAR 
produce the best call quality for me, when compared to nufone and 
Teliax, although both of those companies do ok, my routes to them 
aren't nearly as clean.


I recommend Teliax for good support.

Man! You wrote a lot!
Marty


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Re: [asterisk-users] Newbie Questions

[Lacy Moore - Aspendora]

You can put the Asterisk system in front (i.e., between the PSTN and your Comdial system).  This will let Asterisk choose whether the call should go out over the PSTN or the Internet using VoIP.
 
You would use the same for the second location, provided that is a complete Comdial system.  You could not, however, just put Comdial phones over there and expect it to work.  You also would not be "on the same phone system."  But, if you are looking at tying two offices together using VoIP (and not paying long distance), then yes, this would work.

 
With the right dial plan, you could possibly dial direct if the Comdial has an autoattendant.  In this case, Asterisk would dial into the remote Comdial, wait, then dial the extension number and complete the call.  On the local COmdial, you would most problably have to dial a 9 to get to the Asterisk system.  I imagine, you may be able to use speeddials for the remote extensions which would automatically dial the 9.

 
The possibilities are endless.
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AW: [asterisk-users] NAT issue ? [More Info]

[Christian Stredicke]

Your router might have a problem if there are several 
devices behind NAT with the same port number. Either explicitly set the ports on 
the phone (SIP, RTP, and risk that other ports like DNS, NTP, ... will have 
the same problem) or buy another router that implements NAT/PAT 
properly.
 
CS


Von: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] Im Auftrag von Dovid 
BGesendet: Dienstag, 31. Oktober 2006 22:31An: 
asterisk-users@lists.digium.comBetreff: [asterisk-users] NAT issue ? 
[More Info]

Also when I do sip show peers I get
 
sip show 
peersName/username  
Host    Dyn Nat 
ACL Port Status    
sipmedia/XX   
69.1.236.33 
5060 
Unmonitored10307/10307    
65.8.212.215 D   
N  60414    OK (147 
ms)10305/10305    
(Unspecified)    D   N  
0    UNKNOWN   
10306/10306    
65.8.212.215 D   
N  60414    OK (135 
ms)10320/10320    
(Unspecified)    D   N  
0    UNKNOWN   
10325/10325    
(Unspecified)    D   N  
0    UNKNOWN   
10315/10315    
(Unspecified)    D   N  
0    UNKNOWN   
10310/10310    
69.33.224.23 D   
N  3120 OK (104 ms)8 
sip peers [4 online , 4 offline]
 
307 is the SNOM 300 and 306 is the SNOM 
360
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Re: [asterisk-users] Re: Newbie Questions

[Dovid B]

Please see my previous email in regards to connecting the Comdial system to 
asterisk. In regards to connecting Asterisk to the internet you do not need an 
FXP. All you need is a NIC and to have a good connection to the internet. If you 
can get Asterisk to talk to the Comdial system you should not have a problem 
with your set up ( Comdial Phone -> Comdial System -> Asterisk PBX (FXO?) -> 
Internet -> Asterisk PBX (FXO?) -> Comdial Phone 
).

  - Original Message - 
  From: 
  Ken Williams 
  To: Asterisk Users Mailing List - 
  Non-Commercial Discussion 
  Sent: Wednesday, November 01, 2006 2:49 
  AM
  Subject: [asterisk-users] Re: Newbie 
  Questions
  
  
  I knew I should've waited til tomorrow to send the e-mail so I 
  could have a nights thought on the subject.
   
  That being said, scratch the FXO/FXS thing, what I really picture 
  is someway of passing proprietary information through the Asterisk PBX's on 
  both ends to get remote locations on our phone system through a VOIP 
  connection.  That is:
   
  Comdial Phone -> Comdial System -> Asterisk PBX (FXO?) 
  -> Internet -> Asterisk PBX (FXO?) -> Comdial Phone
   
  I realize this isn't likely an option, but before I try pitching 
  new hardware for everything, thought I'd see if a cheaters option was 
  available.  
   
  Thanks for any help.
  
  

  
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Re: [asterisk-users] Newbie Questions

[Dovid B]

- Original Message - 
From: "Ken Williams" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 


Sent: Wednesday, November 01, 2006 2:10 AM
Subject: [asterisk-users] Newbie Questions


I've been doing a lot of reading over the last few weeks on Asterisk,
and will be implementing a test system this week to play with.

I've got two questions in regards to the ideal implementation for our
company.  First, has anyone written any drivers to interface with
proprietary phones?  Specifically we have a comdial system and if we
could use our existing 35 phones instead of having to buy all new
there'd be huge savings there.  I can't find anywhere that anyone has
written any type of interface for proprietary (no reverse hacks or
anything anywhere from what I can find), so I figure this is a no.

If they are SIP phones and they support SIP then most likely yes. If they 
are POTS phones then you can use them with a voice card or a channel bank. 
If they are proprietary phones from a different PBX then most likely not. To 
cut down costs you may want to look at selling your current systems and your 
phones on eBay.




Now for the more complicated question, that I have my doubts on the
ability to perform.  Would it be possible to throw an Asterisk PBX
system between our Comdial system & the Internet, and then throw another
Asterisk PBX system at a remote location with Comdial phones to tie in
to our system that way?  I'm imagining using a TDM400 or the likes,
connecting to the Comdial via FXO and connecting the to Asterisk PBX's
via FXS.

I have never used this system so I cant comment on it. However if you can 
connect to it with POTS lines it shouldnt be too hard. Also if the system 
can handle a T1 card you may want to connect it to Asterisk that way.



Rereading on the FXO & FXS I think I'm misunderstanding how FXS works
and this won't work at all.

Basicluy an FXO port connects to a phone line (i.e. the line coming in from 
the telco) and the FXS connects to a device (such as a POTS phone or fax 
machine).



Any suggestions for what I'd like to do aside from scrap everything and
start over with IP phones?
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Re: [asterisk-users] Snom or Cisco Phones?

[mitcheloc]

My vote is definitely for Snom, I've worked with Cisco phones for
years, but the Snom is much better integrated, and the feature buttons
can be retooled for any environment, making custom installs very easy.

On 10/31/06, Conrad Wood <[EMAIL PROTECTED]> wrote:

On Tue, 2006-10-31 at 13:29 -0600, Joe Dennick wrote:
> Comparing Snom to Cisco phones is sort of like comparing Mercedes to Kia
> cars

Not really. Both are very good phones.

* My Clients prefer cisco because it looks more business-like. - The new
snom phones do look better though and the side car rules.
* The Cisco phone 'feels' very good in your hand, and the voicequality
is superb. (I'd say slightly better than that of the snom 360)

* Technically, I find the snom phone more advanced and I can do more
cool stuff with it - Cisco doesn't seem to like giving features away in
SIP.
* Snom phones, for example, have freely programmable buttons that can
park/retrieve/transfer calls, show line status etc. I can't get that to
work with Cisco phones at all.
* Putting custom ringtones (and choosing which ones to use) is a
no-brainer with snoms and real trouble with ciscos.
* On ciscos, I find the "upgrade" path from sccp to sip a totally
unnecessary annoyance.



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--

Mitchel Constantin
Snap - A desktop user interface for Asterisk
www.snapanumber.com
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Re: [asterisk-users] Server Recommendations

[Andrew Latham]

Joe

While having done this before I am rather unhappy with current
offerings.  A 4U or bigger HP DL5XX for example will run you upwards
of 10k.  While they are some nice machines I am currently building my
own.   You may want to contact Rhino about there new servers, I feel
that they are filling a niche that their customers have asked for.

http://www.rhinoequipment.com/


Andrew

On 10/30/06, Joe Dennick <[EMAIL PROTECTED]> wrote:

We have a number of clients who will be needing a server to host
Asterisk on.  Many of these clients use analog (FXO) lines that will
need to be connected to Asterisk via Sangoma cards.  Can anyone
recommend an industry-standard server (like IBM, Dell, HP, etc.) that
has enough open PCI slots to handle up to six of the Sangoma cards?  We
would like to be able to tell the customer to just go purchase this
model server from this manufacturer and it will work.  Suggestions?

Thank you!

Joe Dennick
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--
---
Andrew Latham - AKA: LATHAMA (lay-th-ham-eh)
[EMAIL PROTECTED] - [EMAIL PROTECTED]
If any of the above are down we have bigger problems than my email!
Hind sight is most always 20/20 or better.
---
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Re: [asterisk-users] FXO Cards vs. Channel bank with T1

[Rich Adamson]

You'll find the cost of a PRI varies dramatically from one telco to 
another. I've heard numbers in one case where three analog pstn lines 
cost the same as a PRI, another case where 16 analog pstn lines cost the 
same as a PRI. And, having worked in the telecomm industry for many 
years, there are still a very large number of telco's that do not 
support PRI's at all.


Rich


Dovid B wrote:

Looking at the number's now it seems that a T1 will be more.
Anyone here sell PRI's ?

- Original Message - From: "Jay R. Ashworth" <[EMAIL PROTECTED]>
To: 
Sent: Tuesday, October 31, 2006 9:38 PM
Subject: Re: [asterisk-users] FXO Cards vs. Channel bank with T1



On Tue, Oct 31, 2006 at 08:20:57PM +0200, Dovid B wrote:

   Is there any advantage of getting a T1 card with a channel bank
   over 2-3 FXO cards ?


If you need enough ports to make a T-1 card cost-efficient, then you
might oughtta be looking at an Ethernet to FXO media gateway instead --
assuming you need analog interfaces.  FXO side, why not just go T-1 or
PRI?

Cheers,
-- jra
--
Jay R. Ashworth [EMAIL PROTECTED]
Designer  Baylink 
RFC 2100
Ashworth & AssociatesThe Things I Think
'87 e24
St Petersburg FL USA  http://baylink.pitas.com +1 727 
647 1274


"That's women for you; you divorce them, and 10 years later,
  they stop having sex with you."  -- Jennifer Crusie; _Fast_Women_
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Re: [asterisk-users] app_meetme not loading

[Will Roy]

On Sun, Oct 29, 2006 at 04:47:48PM +0800, Will Roy wrote:> I originally built my Asterisk server without installing the Zaptel package> as it was going to be a purely SIP based system. However when I went to
> setup conferencing using meetme I found out that app_meetme is dependant on> the ztdummy for timing. I have now installed the zaptel package and I> believe the ztdummy module is 
loading ok>> [EMAIL PROTECTED] asterisk-1.4.0-beta2]# lsmod> Module                  Size  Used by> ztdummy                 5672  0> zaptel                207908  9
> ztdummy,wcusb,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wct4xxp,tor2>> crc_ccitt               2497  1 zaptel>> I have tried to recompile asterisk again by doing the following>
> make clean> make> make install>> However Asterisk still does not compile  app_meetme. Is there soemthing else
> I should be doing? 
> hat version of zaptel do you have? Asterisk 1.4 requires zaptel 1.4 tobuild chan_zap.> (or a small tweak to zaptel.h and some symlinks to provide the new> cations, if you don't really want to bother your system with zaptel
> 4. Though zaptel 1.4 should hopefully be fully compatible with older> an_zap versions).
 
 
The version of zaptel is  zaptel-1.4.0-beta2. 
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Re: [asterisk-users] Compatability

[Tom Vile]

http://www.digium.com/en/docs/misc/compatibility_notes.phpServer Compatibility The following list of servers are known
to be partially incompatible with Digium® hardware. We do not recommend
using the following computers to set up an Asterisk® server: Dell PowerEdge 1600Dell PowerEdge SC 420Dell PowerEdge SC 1420Dell PowerEdge 650Dell PowerEdge 700Dell PowerEdge 750
Dell PowerEdge 2650IBM eSeries 336IBM xSeries 345IBM xSeries 360/365
			Motherboard Chipset Compatibility
			The following list of motherboard chipsets are known to be partially incompatible with Digium hardware:
			Intel 915 (all variations)Intel E7221Intel E7525
			Digium Hardware Motherboard Compatibility
Some server motherboards utilize an onboard Intel e1000 Ethernet
controller that can interfere with the operation of Digium's cards. The
recommended action for this server is to disable the onboard Ethernet
controller and use a PCI-based solution. Also, the MS-7032 (K8T
Neo-V/K8M Neo-V) motherboard is incompatible with the TE4XXP using the
firmware ending in 164. The problem is that the card will randomly
receive interrupts.On 10/31/06, Joel Hill <[EMAIL PROTECTED]> wrote:
Hi All,I have a new client who has an existing  Asterisk  PABX and is lookingfor us to install a TE110P for him, However he has a Dell SC420 and Ihave never used one before.I have had no problems with any other Dell servers which we use almost
exclusively.Has anyone had any good/bad experiences with the SC420 in relation withDigium cards?Thanks for your help.JoelAsterisk ITwww.asteriskit.com.au
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   http://lists.digium.com/mailman/listinfo/asterisk-users-- Tom VileBaldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephonywww.baldwintechsolutions.comPhone: 518-631-2855 x205Fax: 518-631-2856
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Re: [asterisk-users] No ring tone when using IAX

[Time Bandit]

Then what would be a better solution?

Usually the IAX phone will play you a ring tone until the other end
answer. If you're phone doesn't do it, then it is a flaw in that
phone.

What phone is this ?
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[asterisk-users] Re: anti ex-girlfriend

[Steve Murphy]

On Mon, 2006-10-30 at 02:28 -0800, Pezhman Lali <[EMAIL PROTECTED]>
wrote:
> Hi Dear
> 
> I want to use asterisk(1.2.7.1) as a router by caller
> id.
> 
> I have only a DID number, I want to map this number to
> some ip-phones , base on received Caller-id.
> it is my database's view:
> 
>  456 | DID | 14193016880  |2 | hangup |   
> |
>  455 | DID | 14193016880  |1 | Dial   |
> H323/[EMAIL PROTECTED]|60 | didx.org for
> test by pezhman 
> 
> it's work good.
> 
> but for routing by caller id:
>  456 | DID | 14193016880/2085838  |2 |
> hangup ||
>  455 | DID | 14193016880/2085838  |1 |
> Dial   | H323/[EMAIL PROTECTED]|60 |
> didx.org for test by pezhman   
> 
> this extension does not work , with a call from
> 2085838
> 
> 
> please help me
> tanx 
> Pezhman

Pezhman--

I see someone else has already stated that that the CID qualifier for an
extension
is fixed at the time the dialplan is loaded, and there is no evaluation
done on
the value at run time.

However, this does not prevent you from writing some dialplan code to
query a database,
check the result, and take appropriate actions, without using the
builtin matchcid feature
for an extension.

murf



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Re: [asterisk-users] DTMF Tones

[Eric \"ManxPower\" Wieling]

Andrew Joakimsen wrote:
Where are these DTMF tones going? From where? Be specific, post the 
relevant

config file sections I can't read minds and I'd be surprised if 0.1% of
the people who read this can either

On 10/31/06, Jason Walker <[EMAIL PROTECTED]> wrote:


I have tried beta2, beta3 and now back to 1.2.12.1 and I have correct
DTMF tones 25% of the time.  I have to call several times to enter an
extension.  I have a router and a packet shaper and some other stuff.
Anyone have any other ideas why this might happen.  I do not have any
Zap channels but I am running CentOS4. I also do not have any cards
installed. Thanks


My psychic friend says he's using a compressed codec (G726 maybe) and 
using inband DTMF.  Obviously he should be using RFC2833 if he is using 
a compressed codec.  Heck, I use RFC2833 even for phones that use ulaw. 
 It does not hurt as long as both the phone and Asterisk know what DTMF 
mode to use.

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Re: [asterisk-users] Compatability

[Time Bandit]

I have a new client who has an existing  Asterisk  PABX and is looking
for us to install a TE110P for him, However he has a Dell SC420 and I
have never used one before.
I have had no problems with any other Dell servers which we use almost
exclusively.

Has anyone had any good/bad experiences with the SC420 in relation with
Digium cards?

According to Digium this model is partially incompatible :
http://www.digium.com/en/docs/misc/compatibility_notes.php

I would suggest a Sangoma card if you want to avoid problem with that server
http://www.sangoma.com/datasheets/p_aft-et1-specs

hth
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[asterisk-users] Re: AEL2 and the variables

[Steve Murphy]

On Sun, 2006-10-29 at 22:41 +0100, Dominique Dartois <[EMAIL PROTECTED]>
wrote:
> Hi,
> I am using Asterisk 1.2.12.1 + the AEL2 patch.
> If I use a variable instead of the extension itself, an
> incoming call cannot
> be connected.
> ${ID-FST1} => Dial(SIP/gs|15|r);   <== NON ok
> sip debug shows :
>  Looking for 6674262730 in interne (domain 192.168.1.14)
>  SIP/2.0 404 Not Found
> Is it a bug or am I doing something wrong?
> 
> Thank you.

I believe that extension numbers are not meant to be variable
references. Asterisk
provides no mechanism to evaluate the extension number.

However, you **can** use patterns like _667426XXX to activate the
extension
for such numbers. To activate on ANY number, use "_." as a pattern.

I hope this helps.

murf

> 
> //===
> // extensions.ael2
> globals {
> ID-FST1=6674262730;
> GS=SIP/gs;
> }
> 
> context entrant {
> //6674262730 => Dial(SIP/gs,15,r);  <== OK
> ${ID-FST1} => Dial(SIP/gs|15|r);   <== NON ok
> }
> 
> context interne {
> includes {
> entrant;
> }
> }
> 
> -
> Dominique Dartois


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Re: [asterisk-users] Cisco 7960 Skinny calling SIP phone

[Will Roy]

 
I am running 1.4.0-beta2
 
Date: Tue, 31 Oct 2006 10:57:06 -0600 (CST)From: Anthony LaMantia <[EMAIL PROTECTED]>Subject: Re: [asterisk-users] Cisco 7960 Skinny calling SIP phone
To: Asterisk Users Mailing List - Non-Commercial Discussion       
Message-ID:       <[EMAIL PROTECTED]
>Content-Type: text/plain; charset=utf-8Which asterisk release are you running chan_skinny under?- Original Message -From: Will Roy <
[EMAIL PROTECTED]>To: asterisk-users@lists.digium.comSent: Monday, October 30, 2006 7:52:01 PM GMT-0600 US/Central
Subject: [asterisk-users] Cisco 7960 Skinny calling SIP phoneBefore I got down the path of converting a Cisco 7960 I have over to SIP I wanted to try and set it up using Skinny.The phone registers ok with Asterisk. When I call a SIP softphone extension on my network the call is made and I can answering it. However no voice is heard over the call.
When I debug Skinny on the console after the call has connected I see the following messag:Recieved Alarm Message: DSP Keepalive Timeout [0x3, 0x10, 0x0, 0x7]What additional information would be required to troubleshoot this? or should I stop wasting time and just convert the phone to SIP? :)
regardsWil 
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Re: [asterisk-users] Asterisk Call Statistics

[Moises Silva]

of course you can always use http://cacti.net/download_cacti.php

On 10/31/06, Andrew Joakimsen <[EMAIL PROTECTED]> wrote:

Check out voip-info.org, there are quite a few GUIS some even generate nice
graphs!


On 10/31/06, omar parihuana < [EMAIL PROTECTED]> wrote:
> Hi Folks,
>
> I would like to recover all information about the calls, incoming
> calls, call time, call history, etc in a Web Format,  are  there some
> open source aplication for Asterisk that be easier for use. Pls
> anything suggestion will be very appreciate.
>
> Thanks
>
> Rgds.
> --
> Omar E.P.T
> -
> Certified Networking Professionals make better Connections!
>
> http://omarept.blogspot.com/
>
>   Usysnet Corp
> Open Source Solutions
> www.usysnet.com.pe
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"Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org";
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[asterisk-users] Opinions on the best wholesale origination/term providers

[Brad Templeton]

I've been losing patience with my current provider, a small company
called Sellvoip.  Their termination is good, and they are
asterisk based, but they are understaffed and have no concept
of customer service.  So I'm shopping.

I am interested in the opinions of others on the providers they
work with.

Here are my criteria, roughly in order

a) Decent quality, low latency. 
In particular, this means they probably tie into the PSTN at
multiple points, definitely east and west coast and also in
Europe.  I don't want a California caller calling California
to have to send their packets to the east coast and back.

(This made me discard RNKVoIP, which was high on my list)

b) Fair pricing.  I've seen blended rates down to a penny, and
non-blended down to half/cent in the big city Tier-1s.  I
don't expect the lowest possible price but I don't want to
see 100% markup either.   For a blended rate, let's see
under 1.5 cents to the USA and Canada.  (Canada is actually
down to .8 cents at some providers now, others charge more
for it.)

c) Origination, also at a fair price
Which seems to be about $1/month for DID in USA, $2 in
Canada, and close to 1 cent/minute.  But I can pay more
to get other factors.  I guess I can go to another firm
for origination outside the USA in a pinch.

d) Reliability very high.  Duh.

e) Decent customer service.  If things go down you fix them and
   I can reach you to fix them.  I don't need handholding, I
   know my tools, but I do need you to fix problems.
   If you know your Asterisk, linux and SIP even better.

f) Decent automated interface.
   So I can get DIDs, configure IPs, billing etc.

g) Static IP authentication
   It's faster.  Though dynamic IP registration as a backup is
   handy.

h) Global termination
   I don't want to have to manage and support too many different
   providers.  That's work for me.  So give me good global
   termination prices too.   That knocked out termination.com/icall
   Though if I can't get all I want, I guess I'll buy global from
   one company and domestic from another.

i) No high minimums
   I am just testing my software apps right now so I'm not
   going to bill minutes until much later when they ship.
   So I can't give you tons of minutes per month.  I don't
   mind prepaying.

j) SIP, and decently implemented.  Asterisk/SER is fine.

Now we get to my "nice to have" list

o) IAX as well as SIP.   Makes testing stuff easier.

o) DTMF via SIP-INFO.
This lets me have native bridge for the voice but still
hear the DTMFs at my server, which would be handy.

o) Origination worldwide

o) Toll free origination

o) Cheap toll free termination.  (Why does this cost money anyway?)

o) Don't want E911 service now.  Might want it in future.
   Don't want to pay now.



So here's what I have found that come close

sellvoip -- good quality, low latency, good price.  Online tools suck,
customer service nonexistent

rnkvoip -- most of what I want but east coast gateways only.  Good
customer service but som unreliability in equipment

telcommone.net -- Looks fairy good so far.  $2/DID in small
quantities, but comes down eventually.  Very good term prices.
Claims to enforce instate calling prices.  (Old world thinking)

termination.com -- very good prices but USA only

terravon -- 1.7 / minute.

trxtelecom -- offers free 800 termination, they claim, and pay-you
origination in rural latas if that's your style.  (Great if
you expect most calls to come from cell phones or other people
with bundled long distance blended rates.)

unlimitel -- for canada

netiqsys.net -- no origination but good prices


Any views on these or other providers?
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[asterisk-users] Re: Newbie Questions

[Ken Williams]

I knew I should've waited til tomorrow to send the e-mail so I could have a nights thought on the subject.
 
That being said, scratch the FXO/FXS thing, what I really picture is someway of passing proprietary information through the Asterisk PBX's on both ends to get remote locations on our phone system through a VOIP connection.  That is:
 
Comdial Phone -> Comdial System -> Asterisk PBX (FXO?) -> Internet -> Asterisk PBX (FXO?) -> Comdial Phone
 
I realize this isn't likely an option, but before I try pitching new hardware for everything, thought I'd see if a cheaters option was available.  
 
Thanks for any help.
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[asterisk-users] NAT issue ? [More Info]

[Dovid B]

Also when I do sip show peers I get
 
sip show 
peersName/username  
Host    Dyn Nat 
ACL Port Status    
sipmedia/XX   
69.1.236.33 
5060 
Unmonitored10307/10307    
65.8.212.215 D   
N  60414    OK (147 
ms)10305/10305    
(Unspecified)    D   N  
0    UNKNOWN   
10306/10306    
65.8.212.215 D   
N  60414    OK (135 
ms)10320/10320    
(Unspecified)    D   N  
0    UNKNOWN   
10325/10325    
(Unspecified)    D   N  
0    UNKNOWN   
10315/10315    
(Unspecified)    D   N  
0    UNKNOWN   
10310/10310    
69.33.224.23 D   
N  3120 OK (104 ms)8 
sip peers [4 online , 4 offline]
 
307 is the SNOM 300 and 306 is the SNOM 
360
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[asterisk-users] Newbie Questions

[Ken Williams]

I've been doing a lot of reading over the last few weeks on Asterisk,
and will be implementing a test system this week to play with.

I've got two questions in regards to the ideal implementation for our
company.  First, has anyone written any drivers to interface with
proprietary phones?  Specifically we have a comdial system and if we
could use our existing 35 phones instead of having to buy all new
there'd be huge savings there.  I can't find anywhere that anyone has
written any type of interface for proprietary (no reverse hacks or
anything anywhere from what I can find), so I figure this is a no.

Now for the more complicated question, that I have my doubts on the
ability to perform.  Would it be possible to throw an Asterisk PBX
system between our Comdial system & the Internet, and then throw another
Asterisk PBX system at a remote location with Comdial phones to tie in
to our system that way?  I'm imagining using a TDM400 or the likes,
connecting to the Comdial via FXO and connecting the to Asterisk PBX's
via FXS.  

Rereading on the FXO & FXS I think I'm misunderstanding how FXS works
and this won't work at all.  

Any suggestions for what I'd like to do aside from scrap everything and
start over with IP phones?
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Re: [asterisk-users] simultaneous ring - call groups or queues orsomething else?

[Dovid B]

- Original Message - 
From: "Stephen Bosch" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 


Sent: Tuesday, October 31, 2006 10:06 PM
Subject: Re: [asterisk-users] simultaneous ring - call groups or queues 
orsomething else?




Brian Rogan wrote:

You can just seperate multiple phones with "&" in the Dial command,
as the voip-info wiki page shows:

http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial


Thanks! It's not always clear where to look first for these things.

I'm repeatedly blown away by the ease of configuration and flexibility
of Asterisk.

-Stephen-

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Read the book Asterisk: The future of Telephony
http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11

It will teach you a lot.




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Re: [asterisk-users] Asterisk both behind a NAT and outside at the same time

[Brad Templeton]

On Tue, Oct 31, 2006 at 04:51:44PM -0500, C F wrote:
> >The correct behaviour, as I see it is:
> >
> >a) Native bridge when connecting two external channels -- everybody is 
> >on the real internet
> 
> It might not work if one of them is NATed. Therefore the correct way
> to do this is to use canreinvite=no

Of course, if the external peer or user is natted, you would want to
turn on nat and canreinvite=no for such channels.   However, I do not
wish to set canreinvite=no for an external peer like another asterisk
box with an external IP, or a SIP termination provider.

More than do not wish -- this is the most important case.

> 
> >b) Native bridge when connecting two internal channels -- everybody is 
> >on the 192.168.* network
> 
> canreinvite=yes will take care of this.

Of course, but the point is that the internal channels (local SIP phones)
are involved in connections to both local phones, and to external peers.
> 
> >c) Route RTP through Asterisk when connecting internal and external
> 
> Again by adding canreinvite=no to externals you have this.

But this defeats the entire purpose.  Here are two situations where you would
most definitely not want to have canreinvite=no

a) Call comes in via SIP origination, and you direct it back out to a PSTN
phone via your SIP termination.   You want the RTP to go directly from
the originating point to the termination point, not to hairpin through
your asterisk box, which would just add latency and eat bandwidth.

b) Click to call APP I have where I connect two PSTN endpoints.  Again,
it's necessary not to hairpin the RTP.

c) Double all this with some advanced providers who, once they figure
where the call is actually being terminated, do their own native bridging
and direct your RTP to the actual PSTN entry point.  There it's possible
to get the RTP to go by the shortest (and lowest latency) path it can.

But not if you hairpin it.
> 
> >d) When a channel is to a device behind a remote NAT, the usual rules 
> >apply
> >   (either use STUN or other smart NAT, or route RTP through Asterisk)
> 
> How will asterisk know? The correct *setting* (not behavior) is
> canreinvite=no for the external devices.

I would have to differ.  That's the right setting for external user devices
behind NAT. Do you believe it's correct for devices not behind NAT?

Asterisk can tell if a device is behind NAT if the device has been made
in the last few years, because such devices support a variety of techniques
to inform the server they call that they are behind NAT, and even what
their external IP is if need be.

However, reinvite is not safe with a symmetric nat unless there is really good
cooperation.  So I understand turning off reinvite for any external device
behind NAT.

(Of course a clever box can notice that it sees two devices that have
NAT addresses on the same subnet and both are using the same external
IP.  In that case, it can tell them to send their RTP directly to one
another which is very much the best thing to do.  This allows things
like a branch office using a head office Asterisk server and calls
within the branch staying on the LAN.

This is what the ICE protocol is supposed to solve, of course.)

> 
> Why are you so against having the RTP go thru asterisk?

For external connections?  There are a ton of reasons, some outlined above.
An asterisk box with proper use of native bridging can handle a virtually
unlimited number of calls.Put the RTP through it and it can handle only
a modest number, and reduces the quality of those calls significantly.

Having internal calls go through the asterisk box is not as much of a problem,
but I have noticed latencies because of it even on my internal LAN, though I
have not pieced together exactly why, it's almost certainly because of the
RTP bridging.   Which I always get when I call IAX to SIP of course.  It's
not because of load. 

Anyway, my main question is, has anybody figured out how to make Asterisk
do the right thing here.  I am surprised if my configuration is that
unusual.  Having both an internal and external network is pretty common
at a lot of places.   And a server that's on both is, I think, quite common,
so I had not expected this to be a difficult thing to figure out how to do.
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Re: [asterisk-users] Asterisk both behind a NAT and outside at the same time

[Leo Ann Boon]

Brad Templeton wrote:

On Tue, Oct 31, 2006 at 07:40:35PM +0800, Leo Ann Boon wrote:
  
 
  

Have you tried setting the externalip and localnet parameters?



Localnet makes some sense, and is set (should be the default anyway, no?)
  
I don't think it's set by default. Anyone know how we can see which 
localnets are in use from the CLI? sip show settings doesn't work even 
if I explicitly defined a localnet.

externalip, as I understand it, is for an Asterisk which is behind
a NAT.  This asterisk is not behind a NAT to anybody.  The
phones are behind a NAT to the outside world but not to the
Asterisk box, which has two ethernets on it, one for the internal
natwork and one for the real internet.
  
The way I understand it, externalip and localnet work hand-in-hand. I do 
agree with you that this is commonly used for Asterisk behind a NAT. I 
believe these parameter just helps asterisk determine what to do. In 
your case, you don't lose anything - the external IP would still have to 
be written into every outbound packet.


It uses bindaddr=0.0.0.0 and listens to both addresses.  
  

externalip doesn't affect the bindaddr.

Leo.

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Re: [asterisk-users] simultaneous ring - call groups or queues orsomething else?

[Dovid B]

I dont know the name of the file, but you can do it customly in asterisk

Exten => X,1,Dial(SIP/1234&ZAP/1/18005551212)


- Original Message - 
From: "Stephen Bosch" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 


Sent: Tuesday, October 31, 2006 7:28 PM
Subject: [asterisk-users] simultaneous ring - call groups or queues 
orsomething else?




Hi, folks:

I need to be able to have a single DID ring multiple remote (IP and
PSTN) extensions, and then pass the call to whichever picks up first.
I'm sure this is old hat -- lots of providers offer it.

I see that Trixbox will do it, but it's not clear how it's doing it.
They use different terminology -- a "ring group" and "hunt strategy"

How can it be done with a straight Asterisk server?

Thanks for the help!

-Stephen-
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RE: [SPAM HEADER] - Re: [asterisk-users] Snom or Cisco Phones? - Email found in subject

[Cory Andrews]

I concur with Conrad.  Cisco phones were retrofitted for SIP, whereas
Snom phone are built around, and expressly for, the SIP standard.  To be
in compliance with Cisco regs, you are also supposed to have a SIP User
license and a Smartnet contract for each phone if you abide by their
program.  I realize it is not difficult to obtain the SIP firmware, but
if you were dealing with a Cisco authorized VAR, they are supposed to
sell you the SIP User license and Smartnet, which allows you to obtain a
CCO login and access Cisco firmware files including SIP.  Legitimately
licensing and Smartnetting your phones adds $150-$200 per seat depending
on how far up the food chain you are. 

Cisco is the industry leader in terms of market penetration with their
IP handsets, and it's not a coincidence that you see Cisco phones and
other products prominently features in television shows and movies.  

I think Snom made great strides in terms of aesthetics with their
product line moving from the Snom 190/200 series, which had a look and
feel I would describe as "european", to their newer 300/320/360 series
handsets, which have a look, feel and ballast more suitable for the US
marketplace.


Cory Andrews
e - [EMAIL PROTECTED]
m - 716.907.4059
aim - B2Cory

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Conrad
Wood
Sent: Tuesday, October 31, 2006 4:43 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: [SPAM HEADER] - Re: [asterisk-users] Snom or Cisco Phones? -
Email found in subject

On Tue, 2006-10-31 at 13:29 -0600, Joe Dennick wrote:
> Comparing Snom to Cisco phones is sort of like comparing Mercedes to 
> Kia cars

Not really. Both are very good phones. 

* My Clients prefer cisco because it looks more business-like. - The new
snom phones do look better though and the side car rules.
* The Cisco phone 'feels' very good in your hand, and the voicequality
is superb. (I'd say slightly better than that of the snom 360)

* Technically, I find the snom phone more advanced and I can do more
cool stuff with it - Cisco doesn't seem to like giving features away in
SIP.
* Snom phones, for example, have freely programmable buttons that can
park/retrieve/transfer calls, show line status etc. I can't get that to
work with Cisco phones at all.
* Putting custom ringtones (and choosing which ones to use) is a
no-brainer with snoms and real trouble with ciscos.
* On ciscos, I find the "upgrade" path from sccp to sip a totally
unnecessary annoyance.



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[asterisk-users] NAT issue ?

[Dovid B]

I have a SNOM 360 and a SNOM 300 on a lan. They 
both are connecting to a server with a public IP outside of the lan (dedicated 
server). When the 300 is on alone it works inboud and outbound. When they are 
both plugged in then the 360 will call out and in and the 300 will only allow 
inbound. When the 360 calls the 300 asterisk shows that the call was bridged as 
well as the 360 shows the call in progress however the 300 shows nothing. Any 
idea as to what this may be ?
 
Thanks a lot.
 
Dovid
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Re: [asterisk-users] Re: DTMF Tones

[Zeeshan Zakaria]

And yes, you need to do the same for the phones or adapters you are using. They also have the various options for DTMF setup.
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Re: [asterisk-users] Re: DTMF Tones

[Zeeshan Zakaria]

In my experience DTMF works reliably only when sent over RTP using rfc2833. If you are using SIP, put this line under [general] section in sip.conf: dtmfmode = rfc2833. If you don't want to put this in [general], you can also put dtmfmode = rfc2833 in the declaration of each individual extension which you want to send DTMF using rfc2833.

 
 
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Re: [asterisk-users] Asterisk Call Statistics

[Doug Lytle]

omar parihuana wrote:

Hi Folks,

I would like to recover all information about the calls, incoming
calls, call time, call history, etc in a Web Format,  are  there some
open source aplication for Asterisk that be easier for use. Pls
anything suggestion will be very appreciate.


http://www.areski.net/asterisk-stat-v2/about.php

Doug


-- Ben Franklin quote: "Those who would give up Essential Liberty to 
purchase a little Temporary Safety, deserve neither Liberty nor Safety."


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[asterisk-users] SIP with Qualify and NAT

[David Bath]

Hi guys,

 

I’m having a really strange problem, which I’m
pretty sure has only appeared since my last upgrade (1.2.12.1) .

 

It’s about NAT and Qualify.  I’m using
Asterisk to register with some external SIP providers.  However, they’re
always marked as UNREACHABLE, when they weren’t before! 

 

A typical debug looks like this: 

 

hera*CLI> sip reload

 Reloading SIP

  == Parsing '/etc/asterisk/sip.conf': Found

  == Parsing '/etc/asterisk/sip_notify.conf': Found

 



 

Reliably Transmitting (no NAT) to 195.189.173.10:5060:

OPTIONS sip:sip.voipfone.co.uk SIP/2.0

Via: SIP/2.0/UDP
87.194.194.249:5060;branch=z9hG4bK07c29ff6;rport

From: "asterisk"
;tag=as38a9e906

To: 

Contact: 

Call-ID: [EMAIL PROTECTED]

CSeq: 102 OPTIONS

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 31 Oct 2006 23:22:19 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY

Content-Length: 0

 

---

hera*CLI>

<-- SIP read from 195.189.173.10:5060:

SIP/2.0 200 OK

Via: SIP/2.0/UDP
87.194.194.249:5060;branch=z9hG4bK07c29ff6;received=10.0.0.8;rport=65509

Record-Route: 

From: "asterisk"
;tag=as38a9e906

To: ;tag=as7165a192

Call-ID: [EMAIL PROTECTED]

CSeq: 102 OPTIONS

User-Agent: Voipfone Sip Network

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER

Contact: 

Accept: application/sdp

Content-Length: 0

 



 

--- (12 headers 0 lines)---

Destroying call '[EMAIL PROTECTED]'

Retransmitting #4 (no NAT) to 195.189.173.10:5060:

OPTIONS sip:sip.voipfone.co.uk SIP/2.0

Via: SIP/2.0/UDP
87.194.194.249:5060;branch=z9hG4bK07c29ff6;rport

From: "asterisk"
;tag=as38a9e906

To: 

Contact: 

Call-ID: [EMAIL PROTECTED]

CSeq: 102 OPTIONS

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Tue, 31 Oct 2006 23:22:19 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY

Content-Length: 0

 

 

---

Oct 31 23:22:23 NOTICE[30434]: chan_sip.c:11613
sip_poke_noanswer: Peer 'duncVF_proxy-out' is now UNREACHABLE!  Last
qualify: 0

Destroying call
'[EMAIL PROTECTED]'

hera*CLI>

<-- SIP read from 195.189.173.10:5060:

SIP/2.0 200 OK

Via: SIP/2.0/UDP
87.194.194.249:5060;branch=z9hG4bK07c29ff6;received=10.0.0.8;rport=65509

Record-Route: 

From: "asterisk"
;tag=as38a9e906

To: ;tag=as300cbe8d

Call-ID: [EMAIL PROTECTED]

CSeq: 102 OPTIONS

User-Agent: Voipfone Sip Network

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER

Contact: 

Accept: application/sdp

Content-Length: 0

 

 

--- (12 headers 0 lines)---

Destroying call '[EMAIL PROTECTED]'

hera*CLI> sip no debug

 

 

As you can see, the 200 OK’s appear to be being
ignored… and no amount of fiddling seems to fix it…

 

The SIP config is as follows:

 

type=peer

username=**

fromuser==**

secret==**

fromdomain=sip.voipfone.co.uk

host=sip.voipfone.co.uk

call-limit=5

insecure=very

dtmfmode=rfc2833

nat=yes

qualify=yes

canreinvite=no

context=voipfone-in

disallow=all

allow=g729

allow=ulaw

 

 

Any insight would be very much appreciated.

 

Cheers,

 

Dave

 

 

 

 






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Re: [asterisk-users] No ring tone when using IAX

[Andrew Joakimsen]

Then what would be a better solution?On 10/30/06, Pavel Jezek <[EMAIL PROTECTED]> wrote:
this is really ugly workaround, because using "r" option in dial youlose any other progress tones, including busy, congestion, and you willalways hear ring tone even in case of congestion...PJ
Michiel van Baak wrote:>> check your Dial call. You can add a r to the options. That> way it will generate ring tone while waiting for the other> side to pickup.>> exten => s,n,Dial(IAX2/sometrunk/${NR_TO_DIAL},45,r)
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Re: [asterisk-users] Server Recommendations

[Andrew Joakimsen]

Like Carlos said, the SuperMicro are very good servers, however I don't have a VAR I could recommend on those, as I assume you don't want to put them together.I'll also recommend Tyan, we use some of their 1U gear and its been working flawlessly, but again no VAR to recommend as we build them inhouse. FWIW Sun Microsystems uses the same Tyan motherboards in some of their Opteron servers.

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Re: [asterisk-users] Asterisk and Panasonic KX Model

[C F]

I did this today with a Panasonic KX-TD1232 and a Digium TDM2401E
Card. I hope to put it on the wiki soon, if you need help just tell me
with what.



On 10/30/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:

If Someone did that, How I connect extensions.conf with this type of Hybrid
system to work with asterisk inside this schema:

PSTN--->PANASONIC KX <--> Asterisk
|
|->send internal call


Thanks.

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Re: [asterisk-users] Re: [OT] wi-fi ip phone scenario

[Andrew Joakimsen]

I've done extensive testing, WDS is just as reliable as wired,however at first we had issues with some AP that would not respond and needed to be rebooted. But if its possible to wire the AP< you should since WDS will eat alot of bandwidth and also decrease the range since most the AP will have to be within range of eachother, way more than overlapping coverage
Alberto:I would suggest you try to keep all the AP on the same channel. With that large of a space I wouldnt expect too much interferance from the outside.On 10/28/06, 
Martin Joseph <[EMAIL PROTECTED]> wrote:
On 2006-10-27 11:55:14 -0700, "Andrew Joakimsen" <[EMAIL PROTECTED]> said:>>> Are you using WDS? While it won't totally fix every issue, I've found in my
> trials that turning off WDS and making sure all the AP were connected to the> same wired network was way more reliable, no more random unregistartion and> issue with registering (still seems to unregister at times, but
> re-registartion won't require a reboot).I think it's cleary true that wiring WIFI infrastructure is easier andmore reliable then WDS.On the other hand,  I have been running my little network with WDS for
over three weeks now, and it has been completely reliable.The tricks where to configure things properly and to have the basescloser together then one would think would be needed.Once this was setup. It works, and it keeps working.  We had a couple
of stress tests also, one black out and one unplugged router(carpenter).Came up cleanly and continued working fine.  No mis-registrations andno problems.Marty___
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RE: [asterisk-users] Server Recommendations

[shadowym]

I'm working with Supermicro as well. 

-Original Message-
From: Carlos Rojas [mailto:[EMAIL PROTECTED] 
Sent: Tuesday, October 31, 2006 10:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Server Recommendations

Hello,
 
I'm working with supermicro servers, for the irq problems with Dell, any
people have problems
 
Regards

 
On 10/30/06, Paul Hales <[EMAIL PROTECTED]> wrote: 


How many analog lines are you looking at? Hundreds?

PaulH

On Mon, 2006-10-30 at 17:22 -0600, Joe Dennick wrote: 
> We have a number of clients who will be needing a server to host
> Asterisk on.  Many of these clients use analog (FXO) lines that
will
> need to be connected to Asterisk via Sangoma cards.  Can anyone 
> recommend an industry-standard server (like IBM, Dell, HP, etc.)
that
> has enough open PCI slots to handle up to six of the Sangoma
cards?  We
> would like to be able to tell the customer to just go purchase
this 
> model server from this manufacturer and it will work.
Suggestions?
>
> Thank you!
>
> Joe Dennick
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Re: [asterisk-users] DTMF Tones

[Andrew Joakimsen]

Where are these DTMF tones going? From where? Be specific, post the relevant config file sections I can't read minds and I'd be surprised if 0.1% of the people who read this can either
On 10/31/06, Jason Walker <[EMAIL PROTECTED]> wrote:
I have tried beta2, beta3 and now back to 1.2.12.1 and I have correctDTMF tones 25% of the time.  I have to call several times to enter anextension.  I have a router and a packet shaper and some other stuff.
Anyone have any other ideas why this might happen.  I do not have anyZap channels but I am running CentOS4. I also do not have any cardsinstalled. Thanks___
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Re: [asterisk-users] Asterisk Call Statistics

[Andrew Joakimsen]

Check out voip-info.org, there are quite a few GUIS some even generate nice graphs!On 10/31/06, omar parihuana <
[EMAIL PROTECTED]> wrote:Hi Folks,I would like to recover all information about the calls, incoming
calls, call time, call history, etc in a Web Format,  are  there someopen source aplication for Asterisk that be easier for use. Plsanything suggestion will be very appreciate.ThanksRgds.--
Omar E.P.T-Certified Networking Professionals make better Connections!http://omarept.blogspot.com/  Usysnet CorpOpen Source Solutions
www.usysnet.com.pe___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing list
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[asterisk-users] Compatability

[Joel Hill]

Hi All,

I have a new client who has an existing  Asterisk  PABX and is looking 
for us to install a TE110P for him, However he has a Dell SC420 and I 
have never used one before.
I have had no problems with any other Dell servers which we use almost 
exclusively.


Has anyone had any good/bad experiences with the SC420 in relation with 
Digium cards?


Thanks for your help.

Joel
Asterisk IT
www.asteriskit.com.au
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Re: [asterisk-users] overlap of zap trunk groups

[Eric \"ManxPower\" Wieling]

Damon Estep wrote:

Damon Estep wrote:

Can anyone suggest any reasons why a zap (PRI) b channel should not

be a

member of multiple zap trunk group definitions?



For example;



Group 1 = Channels 1 to 23

Group 2 = channels 1 to 12

Group 3 = channels 13 to 23



The purpose is to restrict the number of channels a particular
extensions can use, but use the entire span for other extensions.

Part of a production /etc/asterisk/zaptel.conf:

group=1
channel => 1-6
group=1,2
channel => 7-12
group=0
channel => 13-16



So the correct solution is to define the channel only once, but the
group= parameter can contain many groups delimited by a comma, correct?


Correct.
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Re: [asterisk-users] FXO Cards vs. Channel bank with T1

[Dovid B]

Looking at the number's now it seems that a T1 will be more.
Anyone here sell PRI's ?

- Original Message - 
From: "Jay R. Ashworth" <[EMAIL PROTECTED]>

To: 
Sent: Tuesday, October 31, 2006 9:38 PM
Subject: Re: [asterisk-users] FXO Cards vs. Channel bank with T1



On Tue, Oct 31, 2006 at 08:20:57PM +0200, Dovid B wrote:

   Is there any advantage of getting a T1 card with a channel bank
   over 2-3 FXO cards ?


If you need enough ports to make a T-1 card cost-efficient, then you
might oughtta be looking at an Ethernet to FXO media gateway instead --
assuming you need analog interfaces.  FXO side, why not just go T-1 or
PRI?

Cheers,
-- jra
--
Jay R. Ashworth 
[EMAIL PROTECTED]
Designer  Baylink RFC 
2100
Ashworth & AssociatesThe Things I Think'87 
e24
St Petersburg FL USA  http://baylink.pitas.com +1 727 647 
1274


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  they stop having sex with you."  -- Jennifer Crusie; _Fast_Women_
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Re: [asterisk-users] Grandstream ATA 286 tdm400 and Asterisk 1.2-13

[Erick Perez]

I forgot to mention that the Carrier that owns the ATA box was not
willing to let me connect directly over IP, I was only allowed to use
the FXS port. He already ack that he has a problem with
disconnections.


On 10/31/06, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:

On Mon, Oct 30, 2006 at 11:17:52AM -0500, Erick Perez wrote:
> Hi people,
>
> I would like to read your suggestions as to where the issue might be.
> ATA286=Grandstream Budgetone ATA 286 in SIP mode. One Lan Port and one FXS
> port.
> TDM04B= 4 FXO signal fxls
> There is a 8FXO-to-SIP unit in this scenario that works perfectly so i
> will not make mention of it.
>
> PSTNVOIPprovider---Internet---ATA286--tdm04b---Asterisk1.2.-13

What exactly is the point is such settings? Why not connect directly to
the provider over SIP? Or to the ATA over SIP?

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Erick Perez
Panama Sistemas
Integradores de Telefonia IP y Soluciones Para Centros de Datos
Panama, Republica de Panama
Cel Panama. +(507) 6694-4780

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Re: [asterisk-users] Re: IAX2 show peers - description

[Marian Rychtecky]

Hi friends,
 thank you for comments...

Marian

Tomislav Parčina napsal(a):

In article <[EMAIL PROTECTED]>, [EMAIL PROTECTED] says...

Hi,

I think the (T) is for Trunk.

Regards
Fred


Hi Fred!

I believe that T is for trunk. Thank you.


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Re: [asterisk-users] Asterisk both behind a NAT and outside at the same time

[C F]

On 10/31/06, Brad Templeton <[EMAIL PROTECTED]> wrote:

On Tue, Oct 31, 2006 at 07:40:35PM +0800, Leo Ann Boon wrote:
> >
> Have you tried setting the externalip and localnet parameters?
>
Localnet makes some sense, and is set (should be the default anyway, no?)

externalip, as I understand it, is for an Asterisk which is behind
a NAT.  This asterisk is not behind a NAT to anybody.  The
phones are behind a NAT to the outside world but not to the
Asterisk box, which has two ethernets on it, one for the internal
natwork and one for the real internet.

It uses bindaddr=0.0.0.0 and listens to both addresses.


> Sorry for my previous post I misunderstood the problem.
> You should set canreinvite=no to all sip peers that connect from outside.


That's precisely what I don't want to do.  This would block native
bridging in the one case where it's most important.


The correct behaviour, as I see it is:

a) Native bridge when connecting two external channels -- everybody is on 
the real internet


It might not work if one of them is NATed. Therefore the correct way
to do this is to use canreinvite=no


b) Native bridge when connecting two internal channels -- everybody is on 
the 192.168.* network


canreinvite=yes will take care of this.


c) Route RTP through Asterisk when connecting internal and external


Again by adding canreinvite=no to externals you have this.


d) When a channel is to a device behind a remote NAT, the usual rules apply
   (either use STUN or other smart NAT, or route RTP through Asterisk)


How will asterisk know? The correct *setting* (not behavior) is
canreinvite=no for the external devices.



The "super" correct behaviour, which I don't expect but would be nice is

e) Clever native bridge between internal and external by being aware that 
the device
   talks to the outside world using a different address than it talks to 
you.
   (Possibly if the phones use STUN they will tell Asterisk their external 
IP, which
   is not the same as Asterisk's though it's on the same subnet)



I have used localnet=192.168.* and nat=yes on a local device and it still
attempts an incorrect native bridge between internal and external, with
one-way audio.

If I do canreinvite=no on the local devices then it works of course, but
now means the local phones will never native bridge amongst themselves.
In a larger network, that would be a problem, and it's a poor result in any
network.



Why are you so against having the RTP go thru asterisk?
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[asterisk-users] compilation problem with asterisk-addons

[Erick Perez]

Hi,

Trying to compile asterisk-addons 1.2.5 on Centos 4.4 produces this:

Note: MySQL libraries are installed and the structure is as follows:
/usr/src/astsources/asterisk-1.2.13
/usr/src/astsources/asterisk-addons-1.2.5

in /usr/src/astsources/asterisk-addons-1.2.5 I do:
make clean
make

and the output is:

./mkdep -fPIC -I../asterisk -D_GNU_SOURCE  -I/usr/include/mysql   `ls *.c`
app_addon_sql_mysql.c:25:27: asterisk/file.h: No such file or directory
app_addon_sql_mysql.c:26:29: asterisk/logger.h: No such file or directory
app_addon_sql_mysql.c:27:30: asterisk/channel.h: No such file or directory
app_addon_sql_mysql.c:28:26: asterisk/pbx.h: No such file or directory
app_addon_sql_mysql.c:29:29: asterisk/module.h: No such file or directory
app_addon_sql_mysql.c:30:34: asterisk/linkedlists.h: No such file or directory
app_addon_sql_mysql.c:31:31: asterisk/chanvars.h: No such file or directory
app_addon_sql_mysql.c:32:27: asterisk/lock.h: No such file or directory
app_saycountpl.c:11:27: asterisk/file.h: No such file or directory
app_saycountpl.c:12:29: asterisk/logger.h: No such file or directory
app_saycountpl.c:13:30: asterisk/channel.h: No such file or directory
app_saycountpl.c:14:26: asterisk/pbx.h: No such file or directory
app_saycountpl.c:15:29: asterisk/module.h: No such file or directory
app_saycountpl.c:16:27: asterisk/lock.h: No such file or directory
cdr_addon_mysql.c:23:29: asterisk/config.h: No such file or directory
cdr_addon_mysql.c:24:30: asterisk/options.h: No such file or directory
cdr_addon_mysql.c:25:30: asterisk/channel.h: No such file or directory
cdr_addon_mysql.c:26:26: asterisk/cdr.h: No such file or directory
cdr_addon_mysql.c:27:29: asterisk/module.h: No such file or directory
cdr_addon_mysql.c:28:29: asterisk/logger.h: No such file or directory
cdr_addon_mysql.c:29:26: asterisk/cli.h: No such file or directory
res_config_mysql.c:41:30: asterisk/channel.h: No such file or directory
res_config_mysql.c:42:29: asterisk/logger.h: No such file or directory
res_config_mysql.c:43:29: asterisk/config.h: No such file or directory
res_config_mysql.c:44:29: asterisk/module.h: No such file or directory
res_config_mysql.c:45:27: asterisk/lock.h: No such file or directory
res_config_mysql.c:46:30: asterisk/options.h: No such file or directory
res_config_mysql.c:47:26: asterisk/cli.h: No such file or directory
res_config_mysql.c:48:28: asterisk/utils.h: No such file or directory
make -C format_mp3 all
make[1]: Entering directory
`/usr/src/astsources/asterisk-addons-1.2.5/format_mp3'
gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations   -D_REENTRANT -D_GNU_SOURCE  -O6-c -o
common.o common.c
common.c:1:29: asterisk/logger.h: No such file or directory
common.c: In function `decode_header':
common.c:93: warning: implicit declaration of function `ast_log'
common.c:93: error: `LOG_WARNING' undeclared (first use in this function)
common.c:93: error: (Each undeclared identifier is reported only once
common.c:93: error: for each function it appears in.)
make[1]: *** [common.o] Error 1
make[1]: Leaving directory
`/usr/src/astsources/asterisk-addons-1.2.5/format_mp3'
make: *** [format_mp3/format_mp3.so] Error 2


Thanks for your help.


--

Erick Perez
Panama Sistemas
Integradores de Telefonia IP y Soluciones Para Centros de Datos
Panama, Republica de Panama
Cel Panama. +(507) 6694-4780

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Re: [asterisk-users] Snom or Cisco Phones?

[Conrad Wood]

On Tue, 2006-10-31 at 13:29 -0600, Joe Dennick wrote:
> Comparing Snom to Cisco phones is sort of like comparing Mercedes to Kia 
> cars

Not really. Both are very good phones. 

* My Clients prefer cisco because it looks more business-like. - The new
snom phones do look better though and the side car rules.
* The Cisco phone 'feels' very good in your hand, and the voicequality
is superb. (I'd say slightly better than that of the snom 360)

* Technically, I find the snom phone more advanced and I can do more
cool stuff with it - Cisco doesn't seem to like giving features away in
SIP.
* Snom phones, for example, have freely programmable buttons that can
park/retrieve/transfer calls, show line status etc. I can't get that to
work with Cisco phones at all.
* Putting custom ringtones (and choosing which ones to use) is a
no-brainer with snoms and real trouble with ciscos.
* On ciscos, I find the "upgrade" path from sccp to sip a totally
unnecessary annoyance.



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Re: AW: [asterisk-users] Snom or Cisco Phones?

[Aaron Daniel]

That and Cisco won't give you the time of day if you don't use their
stuff ;)

We have about 1600 of the Cisco's on campus, and unless you run them on
the call manager, you're not gonna have nearly as many features as any
other phone that's designed with SIP in mind.  That said, if you need a
phone with dialtone, a pretty screen, and limited xml services, then I
will say that the cisco's are extremely easy to provision once you
figure out the upgrade paths.

(Oh, and we're running 7940's and 7960's... if you're looking at the
7912's, etc, good luck, they're a _complete_ pain to work with)

Aaron

On Tue, 2006-10-31 at 20:41 +0100, Christian Stredicke wrote:
> I think one of the differences is: We do pay attention to Asterisk and this 
> mailing list ;-)
> 
> CS 
> 
> -Ursprüngliche Nachricht-
> Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Joao Pereira
> Gesendet: Dienstag, 31. Oktober 2006 13:47
> An: asterisk-users@lists.digium.com
> Betreff: [asterisk-users] Snom or Cisco Phones?
> 
> Hello
> I need to buy IP Phones to work with Asterisk, and I'm in doubt between Snom 
> and Cisco Phones.
> Can you gurus, please, give me your impression of these 2 brands? I need to 
> focus more in SIP and Asterisk compatibility and less in pricing (yes, I know 
> the Cisco are more expensive).
> Are there any features that Snom has, that Cisco doesnt? And are these 
> features important?
> Thanks
> 
> Joao Pereira
> 
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-- 
Aaron Daniel
Senior Voice Analyst
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198

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Re: [asterisk-users] Server Recommendations

[Conrad Wood]

On Tue, 2006-10-31 at 13:52 -0500, Carlos Rojas wrote:
> Hello,
>  
> I'm working with supermicro servers, for the irq problems with Dell,
> any people have problems
>  

I second the supermicro servers - particularly the opteron range based on 
Serverworks HS1000  chipset.
Excellent stuff. Well designed, no irq problems and no timing problems.

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Re: [asterisk-users] Snom or Cisco Phones?

[Mike Dent]

On 10/31/06, Joao Pereira <[EMAIL PROTECTED]> wrote:

Hello
I need to buy IP Phones to work with Asterisk, and I'm in doubt between
Snom and Cisco Phones.
Can you gurus, please, give me your impression of these 2 brands? I need
to focus more in SIP and Asterisk compatibility and less in pricing
(yes, I know the Cisco are more expensive).
Are there any features that Snom has, that Cisco doesnt? And are these
features important?
Thanks

Joao Pereira



I have a Cisco 7960 here in the home office, I recenty purchased a
Snom 300 for the lounge. I wrote a very quick mini-review on my blog:-

http://www.g6phf.co.uk/site/2006/10/05/snom-300-voip-phone-mini-review/

Christian @ Snom, whilst I have your 'ear' here :) Please can you add
a backlight to future revisions of the Snom 300, it would be most
welcome!!

thanks
Mike
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Re: [asterisk-users] Strange Characters in CLI on TTY9

[Tzafrir Cohen]

On Tue, Oct 31, 2006 at 02:03:33PM -0500, Forrest Beck wrote:
> When I look at TTY9 (using init.d and safe_asterisk to start the
> asterisk process), I am getting some strange characters.  When a
> application is run the and the CLI shows the application executing the
> languange almost looks russian...??
> 
> Anyone seen this before?
> http://picasaweb.google.com/jonforrest.beck/AsteriskCLI

Bogus terminal settings show color as cyrillic. vim with syntax
hilighting will probably give you a similar result.

Consult your distro's gurus. Some relevant keyfors: consolechars ,
setfont 

-- 
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[asterisk-users] Re: DTMF Tones

[Nick Adams]

Jason Walker wrote:
I have tried beta2, beta3 and now back to 1.2.12.1 and I have correct 
DTMF tones 25% of the time.  I have to call several times to enter an 
extension.  I have a router and a packet shaper and some other stuff. 
Anyone have any other ideas why this might happen.  I do not have any 
Zap channels but I am running CentOS4. I also do not have any cards 
installed. Thanks


What phones and codec are you using?

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RE: [asterisk-users] overlap of zap trunk groups

[Damon Estep]

> 
> Damon Estep wrote:
> > Can anyone suggest any reasons why a zap (PRI) b channel should not
be a
> > member of multiple zap trunk group definitions?
> >
> >
> >
> > For example;
> >
> >
> >
> > Group 1 = Channels 1 to 23
> >
> > Group 2 = channels 1 to 12
> >
> > Group 3 = channels 13 to 23
> >
> >
> >
> > The purpose is to restrict the number of channels a particular
> > extensions can use, but use the entire span for other extensions.
> 
> Part of a production /etc/asterisk/zaptel.conf:
> 
> group=1
> channel => 1-6
> group=1,2
> channel => 7-12
> group=0
> channel => 13-16
> 

So the correct solution is to define the channel only once, but the
group= parameter can contain many groups delimited by a comma, correct?
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Re: [asterisk-users] Asterisk architecure

[G(P)L]

jez . a écrit :

Dear all,

I've recently installed Asterisk and am trying to understand where 
exactly Asterisk 'fits' in my VOIP architecture. Can/does Asterisk work 
as a proxy. I am specifically interested in SIP. Could anyone perhaps 
point me out to a diagram with SIP users and Asterisk to better 
understand how I should set up my network?


Thank you



Hi,

You can find some interesting diagram here :
http://www.tech-invite.com/Ti-sip-dialog.html

Other diagrams more "architecture ortiented" :
http://lehmann.free.fr/divers/SIP%20tutorial.pdf
slides 32 and after.
The document is not mine :)

If you want something more specific to Asterisk's architecture, I 
recommand you this book : 
http://www.eyrolles.com/Informatique/Livre/9780596009625/livre-asterisk.php


Bye
Guillaume Lehmann

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[asterisk-users] FXO Card's vs. T1

[Dovid B]

Is there any advantage of getting a T1 card with a 
channel bank over 2-3 FXO cards ?
Thanks.
 
Dovid
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Re: [asterisk-users] IPv6

[Tim Panton]

On 26 Oct 2006, at 15:33, David Bandel wrote:


Folks,

Anyone know if Asterisk supports IPv6?  If not, is support planned?


There was a talk at astricon on this. (I think the slides will be  
available

from astricon.net).

The short answer is no, not yet, but folks are working on it.

Tim.

Tim Panton

www.mexuar.com



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Re: [asterisk-users] "Cheapest" way to determine channels in a group from outside asterisk?

[Tim Panton]

On 26 Oct 2006, at 12:12, Nick Adams wrote:

I need to determine the number of active calls in a group from  
outside of Asterisk. Currently I poll the manager API and parse the  
channel status list but this is becoming too expensive on CPU.


What are my options? What is considered "standard practice" ?  
Update a DB field? Poll the manager api? Use an asterisk -rv 'some  
command' call?


That depends on your configuration. If you already use SNMP in your  
organisation, you

might want to use that.

If you are/have a java coder, there is some support for the asterisk  
MIB in the free-ware

from snmp.westhawk.co.uk

(Disclaimer - I wrote large chunks of it so I'm biased :-) )


Tim Panton

www.mexuar.com



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Re: [asterisk-users] simultaneous ring - call groups or queues or something else?

[Stephen Bosch]

Brian Rogan wrote:
> You can just seperate multiple phones with "&" in the Dial command,
> as the voip-info wiki page shows:
> 
> http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial

Thanks! It's not always clear where to look first for these things.

I'm repeatedly blown away by the ease of configuration and flexibility
of Asterisk.

-Stephen-

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Re: [asterisk-users] Live creation of trunk groups

[Tzafrir Cohen]

On Mon, Oct 30, 2006 at 03:08:39PM -0500, Andre Courchesne - Consultant wrote:
> Hi,
> 
>  Is there a way to create trunk groups while asterisk is running.
> 
>  For exemple let's say that zapata.conf defines g0 as channels 1-23
> 
>  I would like (while asterisk is running) define g1 as 1-10 and g1 as 10-23
> 
>  Any hints appreciated.

Edit zapata.conf and from the asterisk cli run 'reload' or 'reload
chan_zap.so' . This will apply most changes from apata.conf. Basically
anything that doesn't change the "very nature" of the channel. Tat is:
you will not be able to create and destory channels that way, or even
change their signalling. But you'll be able to change probably all other
parameters.

-- 
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Re: [asterisk-users] Grandstream ATA 286 tdm400 and Asterisk 1.2-13

[Tzafrir Cohen]

On Mon, Oct 30, 2006 at 11:17:52AM -0500, Erick Perez wrote:
> Hi people,
> 
> I would like to read your suggestions as to where the issue might be.
> ATA286=Grandstream Budgetone ATA 286 in SIP mode. One Lan Port and one FXS 
> port.
> TDM04B= 4 FXO signal fxls
> There is a 8FXO-to-SIP unit in this scenario that works perfectly so i
> will not make mention of it.
> 
> PSTNVOIPprovider---Internet---ATA286--tdm04b---Asterisk1.2.-13

What exactly is the point is such settings? Why not connect directly to
the provider over SIP? Or to the ATA over SIP?

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
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Re: [asterisk-users] channel.c: Avoided initial deadlock

[Tim Panton]

On 26 Oct 2006, at 13:25, asterisk wrote:


Hi all,

Can tell me somebody what meen : channel.c: Avoided initial deadlock

Our customer makes calls with our softphone (with IAX2).
Sometimes the softphon freezes. The call is ACTIVE but the user  
cant hang it up.

At this time in the log file (asterisk/messages) appear the next line:
channel.c: Avoided initial deadlock.

we use: SVN-branch-1.2-r46176M
with VoIP channel (ADSL)

Can you help me? What is the problem?


If you can send us either the output of
iax2 debug or an ethereal trace of the packets in
a conversation that fails I'll take a look.

At a guess your softphone has a bug, and asterisk is just issuing a  
warning,

but I don't have enough evidence yet.


Tim Panton

www.mexuar.com



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AW: [asterisk-users] Snom or Cisco Phones?

[Christian Stredicke]

I think one of the differences is: We do pay attention to Asterisk and this 
mailing list ;-)

CS 

-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Joao Pereira
Gesendet: Dienstag, 31. Oktober 2006 13:47
An: asterisk-users@lists.digium.com
Betreff: [asterisk-users] Snom or Cisco Phones?

Hello
I need to buy IP Phones to work with Asterisk, and I'm in doubt between Snom 
and Cisco Phones.
Can you gurus, please, give me your impression of these 2 brands? I need to 
focus more in SIP and Asterisk compatibility and less in pricing (yes, I know 
the Cisco are more expensive).
Are there any features that Snom has, that Cisco doesnt? And are these features 
important?
Thanks

Joao Pereira

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Re: [asterisk-users] FXO Cards vs. Channel bank with T1

[Jay R. Ashworth]

On Tue, Oct 31, 2006 at 08:20:57PM +0200, Dovid B wrote:
>Is there any advantage of getting a T1 card with a channel bank
>over 2-3 FXO cards ?

If you need enough ports to make a T-1 card cost-efficient, then you
might oughtta be looking at an Ethernet to FXO media gateway instead --
assuming you need analog interfaces.  FXO side, why not just go T-1 or
PRI?

Cheers,
-- jra
-- 
Jay R. Ashworth[EMAIL PROTECTED]
Designer  Baylink RFC 2100
Ashworth & AssociatesThe Things I Think'87 e24
St Petersburg FL USA  http://baylink.pitas.com +1 727 647 1274

"That's women for you; you divorce them, and 10 years later,
  they stop having sex with you."  -- Jennifer Crusie; _Fast_Women_
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Re: [asterisk-users] overlap of zap trunk groups

[Eric \"ManxPower\" Wieling]

Damon Estep wrote:

Can anyone suggest any reasons why a zap (PRI) b channel should not be a
member of multiple zap trunk group definitions?

 


For example;

 


Group 1 = Channels 1 to 23

Group 2 = channels 1 to 12

Group 3 = channels 13 to 23

 


The purpose is to restrict the number of channels a particular
extensions can use, but use the entire span for other extensions.


Part of a production /etc/asterisk/zaptel.conf:

group=1
channel => 1-6
group=1,2
channel => 7-12
group=0
channel => 13-16

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Re: [asterisk-users] FXO Cards vs. Channel bank with T1

[Eric \"ManxPower\" Wieling]

Dovid B wrote:

Is there any advantage of getting a T1 card with a channel bank over 2-3 FXO 
cards ?
Thanks.


In my experience a T-1 port w/channel bank just works better.  The more 
cards you use, the more interrupts are generated.


My standard configuration for analog FXS ports is a T-1 card (Digium or 
Sangoma) and an Adtran TA750 Channel Bank.  The Adtrans can be found 
very cheap on eBay.  FXO ports tend to be much expensive, but you can 
find them on eBay as well.


Why not just get a PRI or channelized voice T-1?
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Re: [asterisk-users] FXO Cards vs. Channel bank with T1

[John Novack]

YES!

Many machines do NOT work well with multiple analog cards. Especially 
the Digium ones.
Channel banks with FXO circuits are harder to come by on the used 
market, though

Many all FXS channel banks can be had used, though.

If you want multiple FXO's and do not want to go the T1 route, look 
towards the Sangoma A200


John Novack


Dovid B wrote:
Is there any advantage of getting a T1 card with a channel bank over 
2-3 FXO cards ?

Thanks.
 
Dovid



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Re: [asterisk-users] Snom or Cisco Phones?

[Joe Dennick]

Comparing Snom to Cisco phones is sort of like comparing Mercedes to Kia 
cars


Joao Pereira wrote:


Hello
I need to buy IP Phones to work with Asterisk, and I'm in doubt 
between Snom and Cisco Phones.
Can you gurus, please, give me your impression of these 2 brands? I 
need to focus more in SIP and Asterisk compatibility and less in 
pricing (yes, I know the Cisco are more expensive).
Are there any features that Snom has, that Cisco doesnt? And are these 
features important?

Thanks

Joao Pereira

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Re: [asterisk-users] Registration problem

[sergio . dippolito]

firewall? i dont think so because sometimes the phone can register ok  
and sudendly the appears unregistered


Leonardo Silva <[EMAIL PROTECTED]> ha escrito:


2006/10/31, Jon Farmer <[EMAIL PROTECTED]>:




Sergio R. D'Ippolito wrote:

Hi all, i have an * version: Asterisk SVN-branch-1.2-r45691, I need to
register a linksys 922 phone thru internet and when I make sip debug
command i see this debug information:



*/SIP/2.0 401 Unauthorized/*

/Via: SIP/2.0/UDP x.x.x.x:1025;branch=z9hG4bK-43bf8123;received=x.x.x.x/

/From: "SPA922" ;tag=685bbad1fae3325do0/

/To: "SPA922" ;tag=as4da6f6ce/

/Call-ID: [EMAIL PROTECTED]/

/CSeq: 5503 REGISTER/

/User-Agent: incore-PBX/

/Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY/

/WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",

nonce="372b2479"/

Asterisk is asking the phone to resend the registration with
WWW-Authenticate using MD5 hash. Make sure the phone supports this and
retry. Or you could turn this option off in the sip.conf.

Regards

Jon

--
Jon Farmer
Telford, Shropshire, UK
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Maybe a Firewall ?

--
Leonardo Silva
fone: 16 8143-1146




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[asterisk-users] DTMF Tones

[Jason Walker]

I have tried beta2, beta3 and now back to 1.2.12.1 and I have correct 
DTMF tones 25% of the time.  I have to call several times to enter an 
extension.  I have a router and a packet shaper and some other stuff. 
Anyone have any other ideas why this might happen.  I do not have any 
Zap channels but I am running CentOS4. I also do not have any cards 
installed. Thanks


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[asterisk-users] Strange Characters in CLI on TTY9

[Forrest Beck]

When I look at TTY9 (using init.d and safe_asterisk to start the
asterisk process), I am getting some strange characters.  When a
application is run the and the CLI shows the application executing the
languange almost looks russian...??

Anyone seen this before?
http://picasaweb.google.com/jonforrest.beck/AsteriskCLI
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Re: [asterisk-users] simultaneous ring - call groups or queues or something else?

[Brian Rogan]

You can just seperate multiple phones with "&" in the Dial command,
as the voip-info wiki page shows:

http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial

On Tue, Oct 31, 2006 at 10:28:32AM -0700, Stephen Bosch wrote:
> Hi, folks:
> 
> I need to be able to have a single DID ring multiple remote (IP and
> PSTN) extensions, and then pass the call to whichever picks up first.
> I'm sure this is old hat -- lots of providers offer it.
> 
> I see that Trixbox will do it, but it's not clear how it's doing it.
> They use different terminology -- a "ring group" and "hunt strategy"
> 
> How can it be done with a straight Asterisk server?
> 
> Thanks for the help!
> 
> -Stephen-
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Re: [asterisk-users] Server Recommendations

[Carlos Rojas]

Hello,
 
I'm working with supermicro servers, for the irq problems with Dell, any people have problems
 
Regards 
On 10/30/06, Paul Hales <[EMAIL PROTECTED]> wrote:
How many analog lines are you looking at? Hundreds?PaulHOn Mon, 2006-10-30 at 17:22 -0600, Joe Dennick wrote:
> We have a number of clients who will be needing a server to host> Asterisk on.  Many of these clients use analog (FXO) lines that will> need to be connected to Asterisk via Sangoma cards.  Can anyone
> recommend an industry-standard server (like IBM, Dell, HP, etc.) that> has enough open PCI slots to handle up to six of the Sangoma cards?  We> would like to be able to tell the customer to just go purchase this
> model server from this manufacturer and it will work.  Suggestions?>> Thank you!>> Joe Dennick> ___> --Bandwidth and Colocation provided by 
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RE: [asterisk-users] Snom or Cisco Phones?

[Ejay Hire]

Cisco Cisco or Linksys Cisco?

Cisco Cisco, I'd prefer the Snom.  Linksys Cisco, it's a tossup.

I've worked with dozens of the Cisco 7960 phones, 25 of the Linksys, and 3
Snom.

My specific issues with the Cisco included poor echo cancellation, problems
with nat traversal, and no web interface.  I didn't like any of the default
ringers on the Snom phones, but the users really liked the LED call
appearance lights compared to the 7960 LCD.  I have no complaints about the
Linksys phones.

Ejay Hire

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joao Pereira
Sent: Tuesday, October 31, 2006 11:21 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Snom or Cisco Phones?

Hello
I need to buy IP Phones to work with Asterisk, and I'm in doubt between Snom
and Cisco Phones.
Can you gurus, please, give me your impression of these 2 brands? I need to
focus more in SIP and Asterisk compatibility and less in pricing (yes, I
know the Cisco are more expensive).
Are there any features that Snom has, that Cisco doesnt? And are these
features important?
Thanks

Joao Pereira

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Re: [asterisk-users] Asterisk both behind a NAT and outside at the same time

[Brad Templeton]

On Tue, Oct 31, 2006 at 07:40:35PM +0800, Leo Ann Boon wrote:
> >  
> Have you tried setting the externalip and localnet parameters?
> 
Localnet makes some sense, and is set (should be the default anyway, no?)

externalip, as I understand it, is for an Asterisk which is behind
a NAT.  This asterisk is not behind a NAT to anybody.  The
phones are behind a NAT to the outside world but not to the
Asterisk box, which has two ethernets on it, one for the internal
natwork and one for the real internet.

It uses bindaddr=0.0.0.0 and listens to both addresses.  


> Sorry for my previous post I misunderstood the problem.
> You should set canreinvite=no to all sip peers that connect from outside.


That's precisely what I don't want to do.  This would block native
bridging in the one case where it's most important.


The correct behaviour, as I see it is:

a) Native bridge when connecting two external channels -- everybody is on 
the real internet
b) Native bridge when connecting two internal channels -- everybody is on 
the 192.168.* network
c) Route RTP through Asterisk when connecting internal and external
d) When a channel is to a device behind a remote NAT, the usual rules apply
   (either use STUN or other smart NAT, or route RTP through Asterisk)

The "super" correct behaviour, which I don't expect but would be nice is

e) Clever native bridge between internal and external by being aware that 
the device
   talks to the outside world using a different address than it talks to 
you.
   (Possibly if the phones use STUN they will tell Asterisk their external 
IP, which
   is not the same as Asterisk's though it's on the same subnet)



I have used localnet=192.168.* and nat=yes on a local device and it still
attempts an incorrect native bridge between internal and external, with
one-way audio.

If I do canreinvite=no on the local devices then it works of course, but
now means the local phones will never native bridge amongst themselves.
In a larger network, that would be a problem, and it's a poor result in any
network.

This is the latest svn of 1.2, by the way.
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[asterisk-users] FXO Cards vs. Channel bank with T1

[Dovid B]

Is there any advantage of getting a T1 card with a 
channel bank over 2-3 FXO cards ?
Thanks.
 
Dovid
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[asterisk-users] overlap of zap trunk groups

[Damon Estep]

Can anyone suggest any reasons why a zap (PRI) b channel
should not be a member of multiple zap trunk group definitions?

 

For example;

 

Group 1 = Channels 1 to 23

Group 2 = channels 1 to 12

Group 3 = channels 13 to 23

 

The purpose is to restrict the number of channels a
particular extensions can use, but use the entire span for other extensions.






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Re: [asterisk-users] Astricon followup

[Dal]

Hello All,
 
This is a great list post, I have blogged about it 
here: http://www.asteriskvoipnews.com/asterisk_news/astricon_2006_followup.html
 
It would be great if people could post there 
response on this post along with the list.  I love reading answers to 
questions like this.  Thanks,
 
-Dal

  - Original Message - 
  From: 
  Dean Collins 
  
  To: Asterisk Users Mailing List - 
  Non-Commercial Discussion 
  Sent: Tuesday, October 31, 2006 8:44 
  AM
  Subject: [asterisk-users] Astricon 
  followup
  
  
  For the benefit of those outside 
  of the USA or those unable to make it to 
  Astricon; I wanted to send out this email.
   
   
  For those of 
  you who attended Astricon in Dallas last week what was the one thing that 
  you saw that made the trip worthwhile?
  (if we post 
  enough information or comments it will be of benefit for those that didn’t 
  attend)
   
   
   
  For me personally it was the 
  volume of neat add-on applications that the Asterisk community are developing; 
  Over time I’m hoping that this leads to something like AppExchange from 
  Salesforce.com were people can choose from over 300+ applications or addons 
  for SF.
   
  I really want to see more speech 
  recognition applications but I think it’s great what Lumen-vox are 
  doing.
   
  I’d also like to see someone post 
  some more modified “ftp to text to speech” http://www.voip-info.org/wiki/view/asterisk+at+home+festival+weather+configuration
  It doesn’t need to be weather, how 
  about Oil futures or wheat prices or score for the weekends games. Any text 
  file accessible by FTP can be implemented into this script. I’d like to see 
  more.
   
  I’m hoping that over time we can 
  see even more to the point that people buy Asterisk just for the applications 
  and we can quote the same price if not more than cisco because of these addon 
  applications.
   
   
   
   
  Cheers,
  Dean
  www.Mexuar.com  
   
  
  

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RE: [asterisk-users] Asterisk on virtual machine

[Ryan Amos]

Asterisk does not work very well in a VM
due to the timeslicing. Dropped calls, jittery audio and echo can all creep in.

 

Good news is that an AD controller runs
just fine in VMware. Just make sure the box has enough RAM to keep it happy,
and use a physical second disk for the Windows install. So I’d suggest
running Asterisk in Linux as the native OS, and running VMware with Windows Server
as a guest OS. This setup should work just fine for you.

 









From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam Robins
Sent: Tuesday, October 31, 2006
9:53 AM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: [asterisk-users] Asterisk
on virtual machine



 

We have a centralized infrastructure where
we deploy Asterisk servers in remote call centers for authentication and
transcoding. SIP g729a calls are then sent over an MPLS VPN to a central
Asterisk farm, from which calls are sent/received via PRI.

 

To avoid placing two servers in each call
center, one for Asterisk and another for Windows AD services, we have been
playing with VMWare. Can anyone provide their experiences in using Asterisk in
a VMWare configuration?  Good/bad/ugly?

 

Thanks,

Adam






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[asterisk-users] simultaneous ring - call groups or queues or something else?

[Stephen Bosch]

Hi, folks:

I need to be able to have a single DID ring multiple remote (IP and
PSTN) extensions, and then pass the call to whichever picks up first.
I'm sure this is old hat -- lots of providers offer it.

I see that Trixbox will do it, but it's not clear how it's doing it.
They use different terminology -- a "ring group" and "hunt strategy"

How can it be done with a straight Asterisk server?

Thanks for the help!

-Stephen-
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[asterisk-users] Example Polycom function key config

[Jamie Heckford]

Hi,

Has anyone here reprogrammed their Polycom features keys using
sip/ipmid.cfg?

If so I would be really grateful if someone could send me an example as
I have tried various entries for hours now and don't seem to be getting
anywhere.

Any help appreciated. 


Kind regards

Jamie Heckford
Technical Consultant
  

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[asterisk-users] Snom or Cisco Phones?

[Joao Pereira]

Hello
I need to buy IP Phones to work with Asterisk, and I'm in doubt between 
Snom and Cisco Phones.
Can you gurus, please, give me your impression of these 2 brands? I need 
to focus more in SIP and Asterisk compatibility and less in pricing 
(yes, I know the Cisco are more expensive).
Are there any features that Snom has, that Cisco doesnt? And are these 
features important?

Thanks

Joao Pereira

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Re: [asterisk-users] Asterisk and ARI (Aterisk Recording Interface) integration problem

[Zeeshan Zakaria]

Everytime a voicemail is recorded, a .txt file is generated. It was working fine before and permissions were automatically set. On my home server it is working perfectly fine. This is another server, with the same settings, and all of a sudden today it has started to give this error. Voicemails etc recorded yesterday are all fine, no problem with permissions. I don't remember changing anything on the server today which could have started giving this error.
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Re: [asterisk-users] Asterisk on virtual machine

[Andrew Latham]

What about the AD in the VM. or running Open LDAP on the Asterisk server.

On 10/31/06, Adam Robins <[EMAIL PROTECTED]> wrote:





We have a centralized infrastructure where we deploy Asterisk servers in
remote call centers for authentication and transcoding.  SIP g729a calls are
then sent over an MPLS VPN to a central Asterisk farm, from which calls are
sent/received via PRI.



To avoid placing two servers in each call center, one for Asterisk and
another for Windows AD services, we have been playing with VMWare.  Can
anyone provide their experiences in using Asterisk in a VMWare
configuration?  Good/bad/ugly?



Thanks,

Adam
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--
---
Andrew Latham - AKA: LATHAMA (lay-th-ham-eh)
[EMAIL PROTECTED] - [EMAIL PROTECTED]
If any of the above are down we have bigger problems than my email!
Hind sight is most always 20/20 or better.
---
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RE: [asterisk-users] Asterisk web interface is not parsing the PHPpages

[Jordan Kirby]

Possibly a silly question, but do you have php installed 
and configured in apache?


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Alok 
MohapatraSent: 31 October 2006 15:45To: 
asterisk-users@lists.digium.comSubject: [asterisk-users] Asterisk web 
interface is not parsing the PHPpages 


Hi All,
  
I have installed Asterisk 1.2.10 on Fedora 5. I have installed Asterisk 
Management Portal (AMP) for web interface. 
After installing properly when 
opening in the webpage it is not parsing the index.php for the AMP. My Database 
is MySQL.and web server is Apache 2.2.
 
Please let me know is this 
configuration problem or this is the problem with Apache (Apache 2.2) 
.
 

Thanks and 
Regards
Alok 
Mohapatra
 
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Re: [asterisk-users] Asterisk web interface is not parsing the PHP pages

[Alex Robar]

Alok,
 
Two things: 
 
1) You said you installed AMP. AMP has ceased development a while ago, but is survived by the FreePBX project. If you actually installed AMP and not FreePBX, I would suggest you get FreePBX running first. A lot of effort went into improving FreePBX from AMP.

 
2) You typically won't find much help for the GUIs from this list because the GUIs have their own mailing lists and forums. Try posting your question to FreePBX.org. You're more likely to get a response there.

 
Alex 
On 10/31/06, Alok Mohapatra <[EMAIL PROTECTED]> wrote:



Hi All,
  I have installed Asterisk 1.2.10 on Fedora 5. I have installed Asterisk Management Portal (AMP) for web interface. 

After installing properly when opening in the webpage it is not parsing the index.php for the AMP. My Database is MySQL.and web server is Apache 
2.2.
 
Please let me know is this configuration problem or this is the problem with Apache (Apache 2.2) .
 

Thanks and Regards
Alok Mohapatra
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http://lists.digium.com/mailman/listinfo/asterisk-users-- Alex Robar[EMAIL PROTECTED] 
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re: [asterisk-users] Live creation of trunk groups

[Andre Courchesne - Consultant]

Well it works. 


If I have group=0 that includes all my channels, I can create group=1 which is 
a subset and a simple reload makes this g1 available to dial on that subset.


Message: 12
Date: Mon, 30 Oct 2006 15:25:06 -0700
From: "Alyed Tzompa" <[EMAIL PROTECTED]>
Subject: re: [asterisk-users] Live creation of trunk groups
To: 
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset="iso-8859-1"


My advice is to first make some tests to see if a reload is 
enough for Asterisk to read any group definitions change in zapata.conf, 
otherwise no on-the-fly change will work

Alyed  




Return-Path: <[EMAIL PROTECTED]> Mon Oct 30 13:23:36 2006
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Mon, 30 Oct 2006 13:23:36 -0700

Hi,

Is there a way to create trunk groups while asterisk is running.

For exemple let's say that zapata.conf defines g0 as channels 1-23

I would like (while asterisk is running) define g1 as 1-10 and g1 as 10-23

Any hints appreciated.

Andre Courchesne
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[asterisk-users] channel.c: Unable to request channel ZAP

[Asterisk]

Hi All,

 

I have one rather
annoying problem...my PBX can work great for weeks, when suddenly I start
receiving these messages when I try to make a zaptel call:

 

Oct 31 13:52:47
NOTICE[15636] app_dial.c: Unable to create channel of type 'ZAP' (cause 34 -
Circuit/channel congestion)

Oct 31 13:52:49
NOTICE[15648] channel.c: Unable to request channel ZAP/g1/247

 

I'm using Sangoma A104
card (with four E1 spans), and these problems are only occurring on the first
two spans (which are connected to a legacy PBX) – the second two spans, which
are connected to the Telco, work perfectly. Even more: when these messages
start to occur, I can hardly initiate any call via problematic two spans (1st and
2nd), where I can with no problem initiate a new call thru the unproblematic
two spans (3rd and 4th).

 

Restart of the Asterisk
is the only cure so far…

 

Does anyone know what
could possibly be the cause, or how could I troubleshot this problem?

 

Regards.

Alex






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Re: [asterisk-users] IVR

[Tzafrir Cohen]

On Mon, Oct 30, 2006 at 06:54:40PM -0500, Vitalie Apostu wrote:
> Greetings,
> 
> If somebody knows how to concatenate several .gsm files in one  or create a
> macro and use with background() please reply.

As simple as:

  cat file1.gsm file2.gsm > both.gsm

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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Re: [asterisk-users] Asterisk Call Statistics

[yusuf]

omar parihuana wrote:

Hi Folks,

I would like to recover all information about the calls, incoming
calls, call time, call history, etc in a Web Format,  are  there some
open source aplication for Asterisk that be easier for use. Pls
anything suggestion will be very appreciate.

Thanks

Rgds.


Hi,

If you have asterisk-addons, you can get all CDR's, which include all the above statistics, written 
to a MySQL or PGSQL database.  It would then be very easy to get this on to a web page.


--
thanks,
yusuf

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Re: [asterisk-users] Asterisk web interface is not parsing the PHP pages

[Time Bandit]

After installing properly when opening in the webpage it is not parsing the
index.php for the AMP. My Database is MySQL.and web server is Apache 2.2.


Please let me know is this configuration problem or this is the problem with
Apache (Apache 2.2) .

The problem is probably that you didn't install PHP

yum install php
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[asterisk-users] Astricon followup

[Dean Collins]

For the benefit of those outside of the USA or those
unable to make it to Astricon; I wanted to send out this email.

 

 

For those of you who attended Astricon in Dallas last week what was
the one thing that you saw that made the trip worthwhile?

(if we post enough information or comments
it will be of benefit for those that didn’t attend)

 

 

 

For me personally it was the volume of neat add-on
applications that the Asterisk community are developing; Over time I’m
hoping that this leads to something like AppExchange from Salesforce.com were
people can choose from over 300+ applications or addons for SF.

 

I really want to see more speech recognition applications
but I think it’s great what Lumen-vox are doing.

 

I’d also like to see someone post some more modified “ftp
to text to speech” http://www.voip-info.org/wiki/view/asterisk+at+home+festival+weather+configuration

It doesn’t need to be weather, how about Oil futures
or wheat prices or score for the weekends games. Any text file accessible by
FTP can be implemented into this script. I’d like to see more.

 

I’m hoping that over time we can see even more to the
point that people buy Asterisk just for the applications and we can quote the
same price if not more than cisco because of these addon applications.

 

 

 

 

Cheers,

Dean

www.Mexuar.com  

 






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