[Asterisk-Users] Reject a call if no callerID

[Hermann Wecke]

I couldn't think any recipe to reject a call if no callerID is presented.
PrivacyManager and Zapateller are not an option, as the call will be 
answered before I can drop it. I just want to "silent drop" the call: no 
callerID, no answer.

Any ideas?
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[Asterisk-Users] Using T100P on E1 line

[Ashish Shinde]

Hi,
   Due to some mistake I ordered a T100P card for an E1 line. Is there
any way in which this card will work for the E1 line.
  What are the options I have?
  Will be grateful if someone can help me in this regard.

Thanks and regards,
 - Ashish
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Re: [Asterisk-Users] problem facing on Firewall, NAT and asterisk

[Brent Goran]

I'm new to Asterisk, but it seems that for clients behind a NAT (e.g. a typical home router), the way to go is to deploy a SIP proxy such as SER, and tell the device to use it as an Outbound Proxy. In so doing, all traffic is tunneled through an outbound TCP connection, with no back-connections to the client, and the NAT shouldn't be an issue.

Could someone please tell me that I have this correct before I go chasing it? (Just got Asterisk working, haven't really looked at SER yet).

Thank you,
Brent


On Wed, 2004-11-03 at 00:02, el Flynn wrote:

prasad_s wrote:


> But the problem is when I call internally between two sip client I don't get voice path between these two sip phones, i.e. I can not talk and hear from both phones,
> though I get message on the asterisk server "connected".
> Is this because of Firewall and NAT between my sip client and asterisk server?

Yes.

> But then how I get register to asterisk server?
> Is there any workaround for this problem

Read this: http://www.voip-info.org/wiki-NAT+and+VOIP

Flynn

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Re: [Asterisk-Users] Hold music while ringing

[Kenneth Lorentsen]

> What im after is a dial plan, so when a user calls into a 'specific' 
> number, instead of hte meharing hte ringing until I pickup the call on 
> my SIP phone.
> 
> Tried looking thru voip-info but the clsoest i could find was a 
> WaitUserOnHold in teh dialplan, not sure if this is what im after nor 
> how to implement it
> 
> Thanks in Advanced
> Matthew


I think what you are looking for is...
In the extension.conf file where you7 have the extension for the the phone you want to 
have music on instead of the ringing tone you add an "m" at the Dial line parameter.

Example.
exten => ,1,Dial(SIP/,20,tTm)

At the end of the line you have "m" as parameter. The "m" is for MusicOn Hold instead 
of the normal ringin tone.

I hope that is what you was looking for.


Best regards
Kenneth Lorentsen


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Re: [Asterisk-Users] Outgoing call fails on pulse dial line

[Vladyslav]

Hi
Some time ago I have tried that without success :(
There is an article on wiki about pulse dialing...
The guy who wrote that patch from Kiev and I have contact him about bugs
etc...
Anyway that patch did not work well on my installation.

Take a look at
http://www.voip-info.org/tiki-index.php?page=Asterisk%20zaptel%20pulse%20dialing

Good luck.

On Tue, 2004-11-02 at 22:57, Goran Obradovic wrote:
> I have Digium FXO/FXS card and one of my phone lines is with pulse
> dialing. At first I didn’t have dial tone at all, but after upgrading
> my Asterisk and Zaptel SW to the latest one (1.0.2) I have dial tone.
> But, when I try to dial outgoing number it fails after first key
> pressed. Does anyone know how to solve this? I am in Eastern Europe
> (Belgrade, Serbia) and our phone lines are mixture of old (pulse) and
> new ones. So I have 2 lines in my house, one with tone dialing and one
> with pulse dialing. Is this related to some signaling settings in
> Zapata.conf? 
> 
> Thanks,
> 
> Goran
> 
> 
> 
> __
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Best regards
Vlad

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Re: [Asterisk-Users] problem facing on Firewall, NAT and asterisk

[el Flynn]

prasad_s wrote:

But the problem is when I call internally between two sip client I don't get voice 
path between these two sip phones, i.e. I can not talk and hear from both phones,
though I get message on the asterisk server "connected".
Is this because of Firewall and NAT between my sip client and asterisk server?
Yes.
But then how I get register to asterisk server?
Is there any workaround for this problem
Read this: http://www.voip-info.org/wiki-NAT+and+VOIP
Flynn
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Re: [Asterisk-Users] Newbie-ISDN NT & ISDN Phone

[Kenneth Lorentsen]

> Newbie question that just joined the list:
> I am interested in using Asterisk at home.
> I have ISDN (BRI) at home with a ISDN NT (providing 2
> POTS RJ11 ports for analog phones and 2 S0 RJ45 ports
> for ISDN bus).
> I am using a cordless ISDN DECT Phone connected to S0.
> The idea is to "stick' Asterisk between:
> - Asterisk to 'talk' to ISDN line
> - My Cordless ISDN DECT to 'talk' to Asterisk.
> Could you pls tell me if this is possible? (I do not
> want to throw away my devices...)
> What hardware I need for?
> 
> Many thanks
> GV


I think that is the information that you need.
http://www.voip-info.org/wiki-Asterisk+zaphfc

Best regards
Kenneth Lorentsen

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[Asterisk-Users] problem facing on Firewall, NAT and asterisk

[prasad_s]

Hi all,
 
I am using asterisk, which is running on one 
machine having static(global) IP.
I have another machine(Internet server with global 
IP, with firewall) working as gateway for internal machines having local IP 
starting with 192.168.xxx.xxx.
My SIP client(xten-xlite) is on LAN machine and 
registers to the asterisk server through this sip phone.
All machines on the LAN, having sip phone are 
registered to asterisk server.
But the problem is when I call internally between 
two sip client I don't get voice path between these two sip phones, i.e. I can 
not talk and hear from both phones,
though I get message on the asterisk server 
"connected".
Is this because of Firewall and NAT between my sip 
client and asterisk server?
But then how I get register to asterisk 
server?
Is there any workaround for this 
problem
 
regards
Prasad Somwanshi.
 
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Re: [Asterisk-Users] Linux and Windows

[Wilson Pickett]

Ben,

> Most Windoze folks are so much into their
> Windoze routine, they won't even use a Linux or BSD box if you install

How can you and so many others speak with any authority on this
subject when you don't run both platforms daily? Sounds like Bush
talking about war, which he's never actively participated in. Windows
has advantages as does unix.

The more honest and precise answer is that if you want to use asterisk
with hardware support, you need to use linux now and maybe someday
soon FreeBSD, which is a great OS. I run all three and each system has
many specific things I like about them.

It's true that running asterisk on Windows is more than awkward and
with hardware prices today, it's best to put together a linux box with
old parts and buy the ones you don't have.
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[Asterisk-Users] Newbie-ISDN NT & ISDN Phone

[Gianni Veloce]

Newbie question that just joined the list:
I am interested in using Asterisk at home.
I have ISDN (BRI) at home with a ISDN NT (providing 2
POTS RJ11 ports for analog phones and 2 S0 RJ45 ports
for ISDN bus).
I am using a cordless ISDN DECT Phone connected to S0.
The idea is to "stick' Asterisk between:
- Asterisk to 'talk' to ISDN line
- My Cordless ISDN DECT to 'talk' to Asterisk.
Could you pls tell me if this is possible? (I do not
want to throw away my devices...)
What hardware I need for?

Many thanks
GV



__ 
Do you Yahoo!? 
Check out the new Yahoo! Front Page. 
www.yahoo.com 
 

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Re: [Asterisk-Users] Polycom IP-500 Network Problems

[Kevin P. Fleming]

Matt Hohman wrote:
at our company and have had the weirdest thing happen.  When the power
is unplugged to the supplied inline injector on the Polycom IP-500's
ALL traffic on our local network just "dies" no packets are able to
route any where. We are using 5 Netgear FSM726S (10/100 managed switch
Yes, we have seen this exact same behavior. When the phone is plugged in 
to the network without a power source, it causes the switch it's plugged 
into to become useless. I have not tried these phones plugged into a 
multi-switch network, so I can't comment on that part.
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Re: [Asterisk-Users] Hold music while ringing

[Wilson Pickett]

> What im after is a dial plan, so when a user calls into a 'specific'
> number, instead of hte meharing hte ringing until I pickup the call on
> my SIP phone.

I don't speak Gaelic, but I think you mean you want "pre-answer" with
music on hold. Answer() the incoming call and then look at Dial()
options where you will find the answer.
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RE: [Asterisk-Users] FXO devel Kit Card

[Brian]

> I have installed and configured the devel kit card with latest CVS
> drivers.  What I start Asterisk in -vvvc mode, the last line is
> 
> WARNING[2412]: chan_zap.c:1323 zt_set_hook: zt hook failed: Device or
> resource busy
> -- Starting simple switch on 'Zap/1-1'
> 
> I have rebooted the computer and am bringing up the modules manually
> 
> I am doing a
> 
> modprobe wcfxs
> Modprobe wcfxo
> 
> Is there anything else I need to do? 

ztcfg -vv

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Re: [Asterisk-Users] Wireless VOIP Phone suggestions

[TC]

Heres another thought IP to Dect wireless handsets
http://www.netvox.com.tw/english/newproducts/v208.htm

but I can't seem to find any in NorthAmerica (any one know any ?)
but there look to be some german resellers
http://tedas.de/english/ip_dect.htm



> On Tue, 2 Nov 2004, Christopher TenHarmsel wrote:
> 
> > Hi all,
> > We're using Asterisk in our office to run our phone system (right now
> > about 5 SIP phones, various Cisco 7912's and 7960's), but we are in
> > desperate need for cordless phones.  We don't need 802.11b/g phones,
> > but just something that is wireless and does SIP.  I've done some
> > searching around, and we've even tried out the one from Pulver
> > Innovations (with no luck), so I wondered if someone could make some
> > suggestions?



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RE: [Asterisk-Users] FXO devel Kit Card

[Cian O'Sullivan]

Hello  

 -- Starting simple switch on 'Zap/1-1'
means it picked up the line. 


My problem is that when I dial in, the card answers the call, but only
leaves static, and there is nothing in the asterisk CLI or logs to
indicate that it did pick up the line.

I don't understand what "device busy" referes to either.

Any pointers?

Cheers

Cian


- Original Message -
From: "Cian O'Sullivan" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
<[EMAIL PROTECTED]>
Sent: Tuesday, November 02, 2004 6:26 PM
Subject: [Asterisk-Users] FXO devel Kit Card


Hello,

I have installed and configured the devel kit card with latest CVS
drivers.  What I start Asterisk in -vvvc mode, the last line is

WARNING[2412]: chan_zap.c:1323 zt_set_hook: zt hook failed: Device or
resource busy
-- Starting simple switch on 'Zap/1-1'

I have rebooted the computer and am bringing up the modules manually

I am doing a

modprobe wcfxs
Modprobe wcfxo

Is there anything else I need to do?  When I ring in the line is picked
up with just static.  Nothing in the asterisks logs indicates it knows
it picked up the line.


Any ideas?

Cheers

Cian

-- Starting simple switch on 'Zap/1-1'
means it picked up the line. 

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[Asterisk-Users] marginal voicemail prompt sound quality

[Damon Estep]

Is there a common solution to poor sound quality when listening to the
voicemail prompts?

When I play the .gsm files in the sounds directory via windows/quicktime
the quality is far better than what I experience when listening to the
same sounds as prompts when connected to voicemail to review messages.
In cases where there is a sequence of "digit" sounds, like one-oh-one
(101) there is truncation (leading and ending) of the digit sounds and a
somewhat scratchy audio quality. Ironically, the recorded voicemail
messages are much better sounding.

The issue can be made more apparent by increasing verbosity of the CLI
(-c), the more verbose it is the more problems there are with sound
quality. This appears to be limited to playback of included .gsm sounds.
Voice channel and recorded vmail sound fine.

The relevant info is; CVS Head, RedHat 9.0, SIP, Cisco ATA 186, No X or
VESA frame buffer.

Thanks,

Damon
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Re: [Asterisk-Users] Hold music while ringing

[Nick Bachmann]

Matthew wrote:
What im after is a dial plan, so when a user calls into a 'specific' 
number, instead of hte meharing hte ringing until I pickup the call on 
my SIP phone.

Tried looking thru voip-info but the clsoest i could find was a 
WaitUserOnHold in teh dialplan, not sure if this is what im after nor 
how to implement it

Could you repost that as  coherent message?  I think what you're looking 
for is right in voip-info.org, but I really can't know until I 
understand what you're asking. 

Nick
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[Asterisk-Users] Asterisk don't start.. undefined symbol: ast_pthread_create

[Serge]

Hi List,

My asterisk don't start at all..
System : RedHat 8 , asterisk cvs 1-0-2
Error:
Asterisk Dynamic Loader Starting:
  == Parsing '/etc/asterisk/modules.conf': Found
 [chan_modem.so]Nov  3 03:29:06 WARNING[1076231168]: loader.c:242
ast_load_resource: /usr/lib/asterisk/modules/chan_modem.so: undefined
symbol: ast_pthread_create
Nov  3 03:29:06 WARNING[1076231168]: loader.c:374 load_modules: Loading
module chan_modem.so failed!

I have make update from asterisk 07/17/04 cvs...

Please Help...

Regards,
Serge


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Re: [Asterisk-Users] ISDN EDSS1 protocol support

[Marcin izo]

Francois  wrote:
> 
> Does ZapRAS allow you to serve several incoming modem calls for dial-up
> internet users, as an ISP in the good old days?
> f.

Yes but only for ISDN modems
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Re: [Asterisk-Users] Best codec for faxes?

[steve szmidt]

On Tuesday 02 November 2004 08:18 pm, Julio Arruda wrote:
> Steve Underwood wrote:

> 'replace' fax over G.711 by fax over G.711 over IP :-)
> The point being, Fax over VOIP (even using G.711), I don't believe would
> be as reliable as Fax over an ISDN b-channel :-) Better now ?

You're totally right. The advantage of ISDN is lost, from a fax viewpoint, 
largly because the unwieldly Internet don't care about lagging packets etc.

Point to point ISDN is a great dedicated circuit. But it ain't that no more 
once you connect it to the Internet. 
-- 

Steve Szmidt

"They that would give up essential liberty for temporary safety 
deserve neither liberty nor safety."
Benjamin Franklin
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Re: [Asterisk-Users] Urgent handler

[el Flynn]

Weng (eB) wrote:
> Is anyone getting urgent handler message in their * box? I am getting many of these 
> message.
> What is the meaning of these messages?
> 
> Cheers
> Weng
> 
> 

how are you running your asterisk from the command line? those messages
typically come up if you've got high verbose settings.

flynn

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Re: [Asterisk-Users] Wireless VOIP Phone suggestions

[Leo Ann Boon]

Right you are. Cisco can be expensive.
Also I did not answer the question he asked.
I have one 900Mhz and one 2.4Ghz cordless. The 2.4 can cause WIFI
interference if used to close to our network access point. I think 900Mhz
units are becoming dinosaurs. But, correct me if I am wrong, the lower
frequency travels farther than 2.4? I know this is true with the old 900mhz
versus 2.4Ghz access points. We had a site survey for both and had to
install more 2.4Ghz access points to cover that same area as covered with
fewer of the old 900MHz units. 

 

Senao (www.senao.com.tw or Engenius in some parts of the world) makes 
excellent long range cordless phones that works well with an ATA. Easily 
works within a mile in a low rise neighborhood.

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Re: [Asterisk-Users] Quintum Tenor DX

[Leo Ann Boon]

Rana
I have Asterisk working with the new Quintum ASM200, 2-FXS/FXO analog 
(the 2-port device with 4-port appearance). My comments:

a. Quintum is bloody messy to set up. No idea where they got their copy 
of the SIP specs. Their terminologies are all different from SIP norms.

b. For the E1/T1 boxes, IIRC, they're all 1/2 capacity. In order words, 
if you buy an E1 30 channel, only 16 VoIP channel (i take it as 
compression channels) is usable simulataneously. If you need to use full 
capacity, you really got to look elsewhere.

c. Voice quality is good.
d. Codec support is good, includes G.723.1 and G.729a
e. Has a nice Tenor manager - only works in Windows though.
f. CDR via TCP.
Frankly, I see Quintum's SIP support as a hack over their H.323 core. 
You might be better off getting something that's SIP from ground up. I 
understand Audiocodes has some very reasonably priced gws (relative to 
Cisco and Quintum), <

At first glance, the Quintum single port E1/T1 DX3000 at US$3,000, looks 
reasonably priced. But, if you take into account  the 1/2 capacity 
issue, you see the cost really escalates.

FYI.
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Re: [Asterisk-Users] Best codec for faxes?

[Julio Arruda]

Steve Underwood wrote:
Julio Arruda wrote:
Roger Schreiter wrote:
Matthew Boehm schrieb:
I'm not using asterisk as the "fax machine" a la rxfax and the like. 
I'm
using an ATA (linksys, grandstream, etc) plugged into a fax machine. 
I know
not to use 729 for faxing. Which 'should' I use?
...

Hi,
use G.711, and you'll have ISDN quality!
Tradidional analog adaptors for ISDN do the same,
and (analog) faxing over ISDN is working fine, isn't it?

I would expect FAX over ISDN to work better than fax over G.711 on 
some networks.
ISDN doesn't have the jitter/packet loss of IP networks (unless you 
control the whole path or has some nice SLA :-)). The price being, is 
a circuit switched technology.

Clue: G.711 is the codec used over ISDN :-)
Ok, now I understand why you mentioned that common codec, let me clarify 
my remark (I can blame on english not being my native language I guess 
:-)..,
'replace' fax over G.711 by fax over G.711 over IP :-)
The point being, Fax over VOIP (even using G.711), I don't believe would 
be as reliable as Fax over an ISDN b-channel :-) Better now ?

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RE: [Asterisk-Users] dialing from mexico mapping numbers.

[James Coberly]

Try

exten => NNN,1,Dial(${TRUNK}/01152664${EXTEN},,) ; Tijuana Local
call takes NNN, sent as 01152664NNN exten =>
NNN,2,Congestion

exten => 01NN,1,Dial(${TRUNK}/011${EXTEN:2},,) ; MX Call takes
01   NN NN NN NN NN, sent as 011 NN NN NN NN NN 
exten => 01NN,2,Congestion

exten => 001NXXNXXNXXX,1,Dial(${TRUNK}/${EXTEN:2},,)  ;  US Calls Dialed
as 001 NNN-NNN- Sent as 1-NNN-NNN- Exten =>
001NXXNXXNXXX,2,Congestion



 
XMC - Ask us for the best prices in T1's for data and voice services
nationwide.  We will not be beat!
 


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of sergio
riveros
Sent: Tuesday, November 02, 2004 5:37 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] dialing from mexico mapping numbers.


Hi,
I am running a small calling center in Mexico where people can call
anywhere 
using asterisk, most calls are either to the USA or inside Mexico.

Everything works great for me using voipjet or simplenet, however what 
confuses people making the calls is the calling sintax  to USA  is 
1-NNN-NNN- or calling to mexico the sintax is 011 52 NN NN NN NN NN,

this works fine but I think it would be better for my customer to
locally 
dial like if they were in mexico for internal purposes, even let
asteriks 
dial the number properly.

from example
when someone dials 001 NNN-NNN- let asterisk dial 1-NNN-NNN-

   USA CALL
when someone dials 01   NN NN NN NN NN let asterisk dial 01152 NN NN NN
NN 
NNMexico call
whe some dials local N NN NN NN  let asterisk dial 01152664 N NN NN NN

 tijuana call

Anyone knows how to do this ?, I have been reading and taking a look at 
extencions.conf without any success for this purpose.

Remeber the purpose for doing this is that the calling center is located
in 
Mexico, and customers get very confused having to dial the number like
being 
in the US.

Any help will be greatly apreciated.

Thanks,

Sergio




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Re: [Asterisk-Users] Best codec for faxes?

[Julio Arruda]

Steve Underwood wrote:
Julio Arruda wrote:
Roger Schreiter wrote:
Matthew Boehm schrieb:
I'm not using asterisk as the "fax machine" a la rxfax and the like. 
I'm
using an ATA (linksys, grandstream, etc) plugged into a fax machine. 
I know
not to use 729 for faxing. Which 'should' I use?
...

Hi,
use G.711, and you'll have ISDN quality!
Tradidional analog adaptors for ISDN do the same,
and (analog) faxing over ISDN is working fine, isn't it?

I would expect FAX over ISDN to work better than fax over G.711 on 
some networks.
ISDN doesn't have the jitter/packet loss of IP networks (unless you 
control the whole path or has some nice SLA :-)). The price being, is 
a circuit switched technology.

Clue: G.711 is the codec used over ISDN :-)
I understand a voice switched PCM channel/DS0 is G.711 and this also is 
the b-channel 'unrestricted 64k' ISDN circuit.
What I'm saying is, with ISDN, you have a end-to-end circuit switched 
channel, with very low delay and jitter, with g.711 over VOIP, you have 
at least the additional sampling time (20ms ?), jitter buffers, variable 
delays in the routers in the path (queueing related).

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[Asterisk-Users] reboot polycom via sip message

[Richard]

Hi,

I tried to use sip NOTIFY message to reboot polycom phones. I followed the
instruction at
http://www.voip-info.org/wiki-Polycom+reboot+hardphone+script.

However I can't get the phone reboot. Has anyone tried it successfully? If
you can please post the ngrep of the NOTIFY message, I'd really appreciate.

Thanks,

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Re: [Asterisk-Users] Wireless VOIP Phone suggestions

[Steve Maroney]

IAXy w/ Analog Cordless Phone ?!?!



Thank you,
Steve Maroney

On Tue, 2 Nov 2004, Christopher TenHarmsel wrote:

> Hi all,
> We're using Asterisk in our office to run our phone system (right now
> about 5 SIP phones, various Cisco 7912's and 7960's), but we are in
> desperate need for cordless phones.  We don't need 802.11b/g phones,
> but just something that is wireless and does SIP.  I've done some
> searching around, and we've even tried out the one from Pulver
> Innovations (with no luck), so I wondered if someone could make some
> suggestions?
>
> Thanks,
> Chris
>
> --
> Chris TenHarmsel
> Software Journeyman
> Atomic Object, LLC
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[Asterisk-Users] Dropping last digit when dialling from analogue phone.

[Demian Dixon]

Hey all,

I'm running Asterisk CVS-HEAD-09/01/04-11:36:41 (version from Voicetronix) with
a Voicetronix OpenSwitch 12 card.  I have a couple of analogue phones plugged
into the Voicetronix and a SIP phone for internal use.

The SIP phone and the internal analogue phones are in the same initial context.
I am trying to dial the number 1,021 4462821, if I do it from the SIP phone, no
problem at all.  If I do it from the analogue phone then it always cuts off the
last digit that I am dialling.. (1 for an outside line)

Any ideas why this isn't working?

here are the relevant dialplans..

[ Context 'internal' created by 'pbx_config' ]

  Include =>'parkedcalls' [pbx_config]
  Include =>'trunklocal'  [pbx_config]
  Include =>'telescum_msg'[pbx_config]
  Include =>'nzcellphone' [pbx_config]
  Include =>'tollfree'[pbx_config]
  Ignore pattern => '1'   [pbx_config]

[ Context 'nzcellphone' created by 'pbx_config' ]
  '_10N' => 1. Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})[pbx_config]
2. Congestion()   [pbx_config]

  Include =>'shortcell'   [pbx_config]

[ Context 'shortcell' created by 'pbx_config' ]
  '_10NXXX' =>  1. Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})[pbx_config]
2. Congestion()   [pbx_config]


And here is the output when I dial using an analogue phone...

   > vpb/1-5: Event [9=>[04] Station OFF Hook]
   > vpb/1-5: handle_notowned: mode=1, event[9][[04] Station OFF Hook
]=[0]
   > vpb/1-5: handle_notowned: playing dialtone
   > [04]: Playing tone
   > vpb/1-5: handle_notowned: mode=1, [9=>0]
   > vpb/1-5: Event [8=>[04] DTMF digit (up): 1]
   > vpb/1-5: handle_notowned: mode=1, event[8][[04] DTMF digit (up): 1
]=[49]
   > vpb/1-5: handle_notowned: mode=1, [8=>49]
   > vpb/1-5: Event [102=>[04] Dial End]
   > vpb/1-5: handle_notowned: mode=1, event[102][[04] Dial End
]=[0]
   > vpb/1-5: handle_notowned: mode=1, [102=>0]
   > vpb/1-5: Event [8=>[04] DTMF digit (up): 0]
   > vpb/1-5: handle_notowned: mode=1, event[8][[04] DTMF digit (up): 0
]=[48]
   > vpb/1-5: handle_notowned: mode=1, [8=>48]
   > vpb/1-5: Event [8=>[04] DTMF digit (up): 2]
   > vpb/1-5: handle_notowned: mode=1, event[8][[04] DTMF digit (up): 2
]=[50]
   > vpb/1-5: handle_notowned: mode=1, [8=>50]
   > vpb/1-5: Event [8=>[04] DTMF digit (up): 1]
   > vpb/1-5: handle_notowned: mode=1, event[8][[04] DTMF digit (up): 1
]=[49]
   > vpb/1-5: handle_notowned: mode=1, [8=>49]
   > vpb/1-5: Event [8=>[04] DTMF digit (up): 4]
   > vpb/1-5: handle_notowned: mode=1, event[8][[04] DTMF digit (up): 4
]=[52]
   > vpb/1-5: handle_notowned: mode=1, [8=>52]
   > vpb/1-5: Event [8=>[04] DTMF digit (up): 4]
   > vpb/1-5: handle_notowned: mode=1, event[8][[04] DTMF digit (up): 4
]=[52]
   > vpb/1-5: handle_notowned: mode=1, [8=>52]
   > vpb/1-5: Event [8=>[04] DTMF digit (up): 6]
   > vpb/1-5: handle_notowned: mode=1, event[8][[04] DTMF digit (up): 6
]=[54]
   > vpb/1-5: handle_notowned: mode=1, [8=>54]
   > vpb/1-5: Event [8=>[04] DTMF digit (up): 2]
   > vpb/1-5: handle_notowned: mode=1, event[8][[04] DTMF digit (up): 2
]=[50]
   > vpb/1-5: handle_notowned: mode=1, [8=>50]
   > vpb/1-5: Event [8=>[04] DTMF digit (up): 8]
   > vpb/1-5: handle_notowned: mode=1, event[8][[04] DTMF digit (up): 8
]=[56]
   > vpb/1-5: handle_notowned: mode=1, [8=>56]
   > vpb/1-5: Event [8=>[04] DTMF digit (up): 2]
   > vpb/1-5: handle_notowned: mode=1, event[8][[04] DTMF digit (up): 2
]=[50]
vpb/1-5: New call for context [internal]
-- Executing Dial("vpb/1-5", "vpb/g1/021446282") in new stack
vpb/1-12: New call for context [incoming4991109]
  ==  g1 requested, got: [vpb/1-12]
   > Restarting monitor
   > Trying to reawake monitor
   > Monitor restarted
   > vpb/1-12: starting call
  == vpb/1-12: Calling 021446282 on vpb/1-12
  == vpb/1-12: Dial parms for vpb/1-12 1/2000ms/4000ms/4000ms/12ms
  == vpb/1-12: Dial parms for vpb/1-12 tone 7->0
  == vpb/1-12: Dial parms for vpb/1-12 tone 0->1
  == vpb/1-12: Dial parms for vpb/1-12 tone 4->2
  == vpb/1-12: Dial parms for vpb/1-12 tone 7->3
  == vpb/1-12: Dial parms for vpb/1-12 tone 3->4
   > vpb/1-5: handle_notowned: mode=1, [8=>50]
   > Monitor got null event
   > vpb/1-5: Event [8=>[04] DTMF digit (up): 1]
   > vpb/1-5: handle_owned: got event: [8=>49]
   > vpb/1-5: handle_owned: putting frame type[-1]subclass[0], bridge=(nil)
-- vpb/1-12: VPB Calling 021446282 [t=12] on vpb/1-12 returned 0
vpb/1-12: chanreads: starting thread
-- Called g1/0214

Re: [Asterisk-Users] Best codec for faxes?

[Steve Underwood]

Julio Arruda wrote:
Roger Schreiter wrote:
Matthew Boehm schrieb:
I'm not using asterisk as the "fax machine" a la rxfax and the like. 
I'm
using an ATA (linksys, grandstream, etc) plugged into a fax machine. 
I know
not to use 729 for faxing. Which 'should' I use?
...
Hi,
use G.711, and you'll have ISDN quality!
Tradidional analog adaptors for ISDN do the same,
and (analog) faxing over ISDN is working fine, isn't it?

I would expect FAX over ISDN to work better than fax over G.711 on 
some networks.
ISDN doesn't have the jitter/packet loss of IP networks (unless you 
control the whole path or has some nice SLA :-)). The price being, is 
a circuit switched technology.

Clue: G.711 is the codec used over ISDN :-)
Steve
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[Asterisk-Users] dialing from mexico mapping numbers.

[sergio riveros]

Hi,
I am running a small calling center in Mexico where people can call anywhere 
using asterisk, most calls are either to the USA or inside Mexico.

Everything works great for me using voipjet or simplenet, however what 
confuses people making the calls is the calling sintax  to USA  is 
1-NNN-NNN- or calling to mexico the sintax is 011 52 NN NN NN NN NN, 
this works fine but I think it would be better for my customer to locally 
dial like if they were in mexico for internal purposes, even let asteriks 
dial the number properly.

from example
when someone dials 001 NNN-NNN- let asterisk dial 1-NNN-NNN- 
  USA CALL
when someone dials 01   NN NN NN NN NN let asterisk dial 01152 NN NN NN NN 
NNMexico call
whe some dials local N NN NN NN  let asterisk dial 01152664 N NN NN NN   
tijuana call

Anyone knows how to do this ?, I have been reading and taking a look at 
extencions.conf without any success for this purpose.

Remeber the purpose for doing this is that the calling center is located in 
Mexico, and customers get very confused having to dial the number like being 
in the US.

Any help will be greatly apreciated.
Thanks,
Sergio
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Re: [Asterisk-Users] Re: How far is IAX to be a Standard

[Steve Underwood]

Matthew Crocker wrote:
Think back,  The telco started with inline signaling.  Pulsing digits, 
cross bar switching, DTMF, CAS   'Yesterdays telco' took signaling 
out of band for a reason.  I'm not sure putting the signaling back 
into the bearer channel is a 'good thing'.
IAX signalling is out of band in a similar way to the way most modern 
telephony works. A single IAX connection contains many independant 
flows. Signalling and voice are distinct. This is equivalent to ISDN or 
SS7 signalling being its own channel but bundled in the same 
E1/OC-192/whatever as the voice. It is also not too far from how SCTP 
works for SS7 over IP.

Steve
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Fwd: [Asterisk-Users] field description /zaptel/zonedata.c

[Luís Palma]

Thanks, for the info.
How do I include a patch in asterisk for my country (Portugal) in zonedata.c?

Here are E.180 Recommendation 2 settings for my country.

Tone Frequency (ITU E.180 supplement 2) - Portugal
Type   Frequency  Cadence
DIAL TONE   400//425CONTINUOUS
SPECIAL DIAL TONE   425 1.0 â 0.2
RINGING TONE400//4251.0 â 5.0
BUSY TONE   400//4250.5 â 0.5
CONGESTION TONE 425 0.2 â 0.2
NUMBER UNOBTAINABLE TONE400//425 27)0.2 â 0.2
WAITING TONE425 0.2 â 0.2 â 0.2 â 5.0
PAYPHONE RECOGNITION TONE   1477/9410.2 â 0.2 â 0.2 â 2.0
Analogue exchanges: 400 Hz; digital exchanges: 425 Hz

As default I think we should use 425 Hz (most of exchanges are digital).

Other question, is there any policy on adding country setting as a per
needed basis on asterisk files zonedata.c?

Regards
LuÃs Palma
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[Asterisk-Users] Voicemail custom sound configuration

[asterisk]

Hi,

I am trying to customized voicemail sound prompt file.

I have changed the sound file for mailbox 1000 in 
/var/spool/astgerisk/voicemail/default/1000

I have changed the unavail.gsm file but the prompt has not changed.

I have added a new file in the directory and reference that file in 
voicemail(filename) in extensions.conf but prompt does not play.

Please advise 
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Re: [Asterisk-Users] gastman - documentation?

[Nicolás Gudiño]

Hello,

On Tue, 2 Nov 2004 17:21:45 -0600, Matthew Boehm <[EMAIL PROTECTED]> wrote:
> I've added all our SIP extensions to gastman but when someone makes a call
> to another extension, a new icon gets created. Why doesn't gastman use the
> icon that is already there? The existing icon turns green so I know that
> gastman knows its there, but...
> 
> Does anyone know how to add an icon for a queue?

Did you try [shameless plug] Flash Operator Panel? http://www.asternic.org
It can monitor queues and agents. Regards,

-- 
Nicolás Gudiño
Buenos Aires - Argentina
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Re: [Asterisk-Users] FXO devel Kit Card

[Steve Totaro]

- Original Message - 
From: "Cian O'Sullivan" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
<[EMAIL PROTECTED]>
Sent: Tuesday, November 02, 2004 6:26 PM
Subject: [Asterisk-Users] FXO devel Kit Card

Hello,
I have installed and configured the devel kit card with latest CVS
drivers.  What I start Asterisk in -vvvc mode, the last line is
WARNING[2412]: chan_zap.c:1323 zt_set_hook: zt hook failed: Device or
resource busy
   -- Starting simple switch on 'Zap/1-1'
I have rebooted the computer and am bringing up the modules manually
I am doing a
modprobe wcfxs
Modprobe wcfxo
Is there anything else I need to do?  When I ring in the line is picked
up with just static.  Nothing in the asterisks logs indicates it knows
it picked up the line.
Any ideas?
Cheers
Cian
   -- Starting simple switch on 'Zap/1-1'
means it picked up the line. 

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Re: [Asterisk-Users] gastman - documentation?

[Adam Goryachev]

On Wed, 2004-11-03 at 10:21, Matthew Boehm wrote:
> I've added all our SIP extensions to gastman but when someone makes a call
> to another extension, a new icon gets created. Why doesn't gastman use the
> icon that is already there? The existing icon turns green so I know that
> gastman knows its there, but...

Gastman displays an icon for each extension, but it will then show a
mini-version of that icon for each currently active channel (call).

ie, it shows each call, not just each extension.

> Does anyone know how to add an icon for a queue?

Nope, sorry... I found gastman good for a while, but I dropped it a long
time ago Look on the wiki, and you will probably find something that
more suits your requirements.

Regards,
Adam


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[Asterisk-Users] FXO devel Kit Card

[Cian O'Sullivan]

 Hello,

I have installed and configured the devel kit card with latest CVS
drivers.  What I start Asterisk in -vvvc mode, the last line is

WARNING[2412]: chan_zap.c:1323 zt_set_hook: zt hook failed: Device or
resource busy
-- Starting simple switch on 'Zap/1-1'

I have rebooted the computer and am bringing up the modules manually

I am doing a 

modprobe wcfxs
Modprobe wcfxo

Is there anything else I need to do?  When I ring in the line is picked
up with just static.  Nothing in the asterisks logs indicates it knows
it picked up the line.


Any ideas?

Cheers

Cian


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[Asterisk-Users] gastman - documentation?

[Matthew Boehm]

I've added all our SIP extensions to gastman but when someone makes a call
to another extension, a new icon gets created. Why doesn't gastman use the
icon that is already there? The existing icon turns green so I know that
gastman knows its there, but...

Does anyone know how to add an icon for a queue?

Thanks,
Matthew

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[Asterisk-Users] E&M timing

[Chad Wicker]

I am looking for a explanation as to the definition of these
parameters:

A variety of timing parameters can be specified as well
 Including:
prewink: Pre-wink time
preflash:Pre-flash time
wink:Wink time
flash:   Flash time
start:   Start time
rxwink:  Receiver wink time
rxflash: Receiver flashtime
debounce:Debounce timing

and also is there a way to specify DTMF on and Offtimes:  I.E. send the
dtmf for 50 milliseconds and then wait 50 milliseconds then send the
next digit.

Chad C. Wicker
Systems Engineer
Petrocom
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Re: [Asterisk-Users] Best codec for faxes?

[Julio Arruda]

Roger Schreiter wrote:
Matthew Boehm schrieb:
I'm not using asterisk as the "fax machine" a la rxfax and the like. I'm
using an ATA (linksys, grandstream, etc) plugged into a fax machine. I 
know
not to use 729 for faxing. Which 'should' I use?
...
Hi,
use G.711, and you'll have ISDN quality!
Tradidional analog adaptors for ISDN do the same,
and (analog) faxing over ISDN is working fine, isn't it?
I would expect FAX over ISDN to work better than fax over G.711 on some 
networks.
ISDN doesn't have the jitter/packet loss of IP networks (unless you 
control the whole path or has some nice SLA :-)). The price being, is a 
circuit switched technology.


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Re: [Asterisk-Users] The best SIP HW Phone and WLAN Phones forAsterisk

[Steve Totaro]

- Original Message - 
From: "Christopher TenHarmsel" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
<[EMAIL PROTECTED]>
Sent: Tuesday, November 02, 2004 4:50 PM
Subject: Re: [Asterisk-Users] The best SIP HW Phone and WLAN Phones 
forAsterisk


On Tuesday 02 November 2004 16:42, Steve Totaro wrote:
- Original Message -
From: "Christopher TenHarmsel" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[EMAIL PROTECTED]>
Sent: Tuesday, November 02, 2004 4:18 PM
Subject: Re: [Asterisk-Users] The best SIP HW Phone and WLAN Phones
forAsterisk
> We've had good luck with the Cisco 7912s in a very similar
> situation (office of 15 people), but to get SIP working you have to
> be able to download the newest firmware from Cisco.  I think
> there's some info about this on the voip-info.org wiki.
>
> We haven't had any luck with wireless SIP phones, if you find
> anything out, please let me know too.
>
> -Chris
These phones are awsome based on a very limited trial.
http://www.zyxel.com/product/P2000W.html
Those make me a little worried because we tried one of the WiSiP phones
from Pulver Innovations, which are OEM'd versions of the 2000W's, and
they were aweful, we couldn't get them configured, the documentation
was virtually non-existant, and there were no support channels.  Have
you had luck personally getting these to work?
-Chris
Yes, works great.  Asterisk server is on a public IP, nat=yes and the phone 
is set to DHCP.  It works anywhere it can get wifi.

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[Asterisk-Users] FXO module in TDM400P (UK, BT) - Hangup detection failing

[StrUK]

Hi,
I'm a UK-based * newbie with a BT line on a PIII 2GHz 1Gb Fedora Core 2  
linux box, fully patched up, experiencing a little difficulty with *  
(both CVS V1-0 and yesterday's CVS HEAD). I've been through the  
archives and even popped on to freenode/#asterisk for a while, but no  
firm resolution presented itself. Last relevant post appears to have an  
unresolved cliffhanger:  
http://lists.digium.com/pipermail/asterisk-users/2004-September/ 
061518.html

I had an X100P as my PSTN link and that was working fine for taking and  
making calls, but of course no caller ID (Plenty of "Didn't finish  
Caller-ID spill.  Cancelling" and that's about it).

Needing CID for my application, I procured an FXO module for my TDM400P  
which already had one FXS module in it (also working fine, happily  
shoving and receiving CID info around between my SIP soft phones and  
the Zyxel SIP hardphone).

The wcfxs module is correctly installed with opermode=uk (dmesg  
confirms this), and polarity CID detection works a treat. Call routing  
is fine and, naturally, none of the SIP stuff has broken :-)

HOWEVER, the FXO module isn't detecting remote party clear down events.
Two example cases:
Zap/4-1 (the FXO module on the TDM400P) detects ring because someone is  
dialling my BT line. My dialplan displays CID info to console for  
debugging and then dials my SIP softphone (SIP/stripes), SIP hardphone  
(SIP/zyxel) and Zap hardphone (Zap/1-1).

1/ If the incoming party clears down before one of the internal  
extensions answers, they continue to ring (* hasn't logged the remote  
hang up event on Zap/4-1) and dead air is heard should they  
subsequently answer.

2/ If the incoming party stays on long enough for voicemail to kick in,  
then I get a recording comprising their message, pop/click, 2secs  
silence, 5secs solid tone, pop/click, followed by dead air (and silence  
detection is the only thing stopping the message hitting the 120 second  
maximum message length ceiling)

Both of these cases are consistent with how I understand * to work  
should the channel not notice Zap/4-1 going down - bully for me, bummer  
for operational use ;-)

Speaking with the ever-so helpful folk on freenode/#asterisk, I've  
tried loopstart signalling (nice idea - failed miserably where remote  
clear down made * think that the remote party was trying to transfer  
calls - lots of MOH messages and 's' priority of the context invoked);  
I've tried tweaking indications.conf to make *'s notion of 'busy' for  
my line match what is being heard as the tone, but that hasn't worked  
and incoming calls still stay 'connected' until the voicemail's silence  
detection cuts them off.

I can live with voice messages being 9 seconds longer than they need to  
be; I can't, though, live with incoming callers that hangup before  
anything answers still resulting in the internal extensions being  
signalled.

I don't have a lighted keypad, but have seen a deflection on a  
multimeter across A/B wire to suggest that the phone company are  
signalling remote clear down with a polarity reversal.  I'm told  
(again, kind folk on irc) that BT do use supervised disconnects and so  
I am doing the right thing using fxs_ks signalling.

I guess my question is: does anyone have polarity reversal hangup  
detection working on a BT line with an fxo module in a TDM400P?

If so - your direction would be most greatly appreciated.   I've posted  
slightly censored versions of my config to http://ermy.net/senast/, in  
case that might help.

Many thanks in advance,
Mark/
--
// if it doesn't go woof when you light it, you ain't dun it right.
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Re: [Asterisk-Users] Re: loss concealment (Steve Kann)

[Steve Kann]

Public Dump wrote:

  Thank you for the URL. 
To clarify the information from the page, IF I would use iLBC I would
enjoy loss concealment ?
  


When/If this proposed jitterbuffer is implemented, then if you would be
able to enjoy PLC (Packet loss concealment) with your choice of several
codecs, including at least G711, PCM, Speex and iLBC.  The first two
with a simple concealment algorithm (I think one is described in an
RFC), and the latter via their built-in interpolation algorithms.

-SteveK


  
Message: 1
Date: Mon, 01 Nov 2004 10:30:41 -0500
From: Steve Kann <[EMAIL PROTECTED]>
Subject: Re: [Asterisk-Users] loss concealment
To: Asterisk Users Mailing List - Non-Commercial Discussion
	<[EMAIL PROTECTED]>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset="iso-8859-1"

Public Dump wrote:

  
  
Is asterisk capable of sealing (some amount) of losses that occur on 
IP based channels before it routes the Calls to a TDM channel (BRI, 
E1, etc.) to limit quality loss if IP loss occurs ?

  
  
No, not yet.

See
http://www.voip-info.org/tiki-index.php?page=Asterisk%20new%20jitterbuff
er

-SteveK
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Re: [Asterisk-Users] IAX2 audio problems but SIP OK?

[Steve Kann]

Whisker, Peter wrote:
[sorry about previous mis-post]
I have an * switch at home and one in the office. Both similar new CVS head
versions and both with chan_sip2 built in:
Asterisk CVS-HEAD-10/12/04-17:43:26
Asterisk CVS-HEAD-10/13/04-12:53:52
One is on a T1 connection and the other is on 576k/288k ADSL. The Ping time
is about 30ms between the two servers, 90% of which is the ADSL delay.
When I interconnect them with IAX2, I get rather choppy audio - with what
sounds like dropped packets and jitter. 

However when I interconnect with SIP is it clean and with no dropouts. The
network path and timings are identical for both protocols and there is
little noticeable difference when I play with the jitterbuffer setting in
iax.conf.
Does anyone have any idea why IAX protocol is causing this kind of problem?
My ADSL is PPPoE which has an MTU of 1492 I think. Could this be causing the
problems?
 

No, you shouldn't be sending any IAX2 packets which approach this size.
1) What codecs are you using in each case?
2) Do you have other traffic on the link?  It's possible that somewhere 
along your path, the RTP audio traffic from SIP is getting some kind of 
helpful QoS benefit, while the IAX2 traffic is not?

-SteveK
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RE: [Asterisk-Users] Broadvoice with multiple numbers

[Jay Milk]

I have done that, with varying success.  It works 90% of the time, but
when it doesn't, using the full number always DOES work.  As a general
rule, I don't suggest using the /ext clause anymore.

> -Original Message-
> From: Seth Remington [mailto:[EMAIL PROTECTED] 
> Sent: Tuesday, November 02, 2004 12:59 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Broadvoice with multiple numbers
> 
> 
> On Tue, 2004-11-02 at 11:29, Jay Milk wrote:
> > I have two broadvoice numbers:
> > 
> > [incoming]
> > exten => 16125551212,1,Macro(dialext_incoming,${EXT_BIZ},2000)
> > exten => 14085551212,1,Macro(dialext_incoming,${EXT_ALL},1000)
> 
> You can also add the extension you want used for incoming 
> calls in your register statement.
> 
> register => user:secret:[EMAIL PROTECTED]:port/extension
> register => 2345:[EMAIL PROTECTED]/1234
> 
> In the above example, define extension 1234 in 
> extensions.conf in the default SIP context and all incoming 
> calls will land there.
> 
> -Seth

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RE: [Asterisk-Users] Unable to find a path from GSM to SPEEX ??

[Kevin Walsh]

Atuc [EMAIL PROTECTED] wrote:
> At 17:45 02.11.2004, you wrote:
> > Have you compiled and installed SpeeX on your system?  Version 1.1.6
> > works well, although I'd guess that you don't have SpeeX installed at
> > all.
> >
> that sounds plausibly,
> but is it true that i have to install speex as system libs? also ilbc? i
> thought asterisk has all this stuff in the distribution, i run here
> another asterisk and ilbc is working out of the box without installation.
> 
Asterisk comes with iLBC source, but not SpeeX.  You will have to install
that yourself, which is not difficult to do.

-- 
   _/   _/  _/_/_/_/  _/_/  _/_/_/  _/_/
  _/_/_/   _/_/  _/_/_/_/_/  _/   K e v i n   W a l s h
 _/ _/_/  _/ _/ _/_/  _/_/[EMAIL PROTECTED]
_/   _/  _/_/_/_/  _/_/_/_/  _/_/

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RE: [Asterisk-Users] anyone got a 7910 to work with asterisk?

[Joel Berry]

Mark,
  The 7910 support in the native chan_skinny works pretty well.  You will
only get basic call features.. No Hold, speed dial, transfer, etc.  There is
another project for chan_sccp.  It seems to add many more features.
Personally, I use the chan_skinny because it is more stable.  with
chan_sccp, I could easily kill the asterisk server.  With chan_skinny, it
hasn't rebooted once.
  I had to setup a TFTP server with the following files:

XMLDefault.cnf.xml : Containing






2000

192.168.200.75




P00405000600


P00303020214
P00303020214






You will need to change the processNodeName for your Asterisk Server.  Also,
the loadInformaiton6 must match your version of your phone, or your image if
you have cisco access.

With the DHCP server telling the phone to use your TFTP server, you should
get your phone trying to hit the asterisk server.

On the asteirsk server, you need to modify the skinny.conf file in the
/etc/asterisk directory.
skinny.conf : Containing
[general]
port = 2000 ; Port to bind to, default tcp/2000
bindaddr = 192.168.200.75   ; Address to bind to
dateFormat = M-D-Y  ; M,D,Y in any order (5 chars max)
keepAlive = 120

; Typical config for a 7910
[cisco1]; Device name
device=SEPXX  ; Offical identifier  # NEed to set to your ID
;version=P002F202   ; Firmware version identifier
;host=192.168.1.144 ;
;permit=192.168.0/24; Optional, used for authentication
nat=0
callerid="Your Name" 
mailbox=1000
callwaiting=0
transfer=1
threewaycalling=0
context=longdistance
line => 301 ; Dial(Skinny/[EMAIL PROTECTED])

There are lots of resources on the voip-info.org website.  Should be able to
give you lots of pointers...

Joel Berry


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Mark
Phillips
Sent: Tuesday, November 02, 2004 2:44 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] anyone got a 7910 to work with asterisk?


I've looked all over but can't find anything about the 7910.

Mark

--

Mark Phillips, G7LTT/KC2ENI
Randolph, NJ

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[Asterisk-Users] agents can't hear callers.

[Ruben Santos]

We are having issues with calls that connect where callers can hear the 
agent speaking, but the agents can't hear the callers. I noticed the 
following on our message log.

Nov  2 12:57:13 WARNING[424067093]: ast_streamfile failed on Zap/42-1
Nov  2 12:57:13 WARNING[424067093]: Failed to write frame
Nov  2 12:57:13 WARNING[424067093]: ast_streamfile failed on Zap/42-1
Nov  2 12:57:13 WARNING[424067093]: Failed to write frame
Nov  2 12:57:13 WARNING[424067093]: Failed to write frame
Nov  2 12:57:13 WARNING[424067093]: ast_streamfile failed on Zap/42-1
We are using the following version of Asterisk;
Asterisk CVS-HEAD-10/23/04-23:35:35, Copyright (C) 1999-2004 Digium.
--

Ruben T. Santos
Director of Network Operations
Brand X Networks
(866) 487-3244 x 5203
[EMAIL PROTECTED]
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Re: [Asterisk-Users] Asterisk 1.0.2, Zaptel 1.0.2, Linux 2.6.9 on a PCEngines WRAP\Soekris net4801 in Compact Flash

[Jon Lawrence]

On Tuesday 02 November 2004 16:06, Brian Wilkins wrote:
> Sounds cool, but I heard a rumor that CF Cards don't like too many rewrites
> or they start losing data.
>
I'm not certain about this, but I think CF cards have iro 1,000,000 write 
cycles lifetime. Obviously how quickly you'd go through 100 cycles 
depends on how often you write to the card.

Jon
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Re: [Asterisk-Users] The best SIP HW Phone and WLAN Phones forAsterisk

[Christopher TenHarmsel]

On Tuesday 02 November 2004 16:42, Steve Totaro wrote:
> - Original Message -
> From: "Christopher TenHarmsel" <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <[EMAIL PROTECTED]>
> Sent: Tuesday, November 02, 2004 4:18 PM
> Subject: Re: [Asterisk-Users] The best SIP HW Phone and WLAN Phones
> forAsterisk
>
> > We've had good luck with the Cisco 7912s in a very similar
> > situation (office of 15 people), but to get SIP working you have to
> > be able to download the newest firmware from Cisco.  I think
> > there's some info about this on the voip-info.org wiki.
> >
> > We haven't had any luck with wireless SIP phones, if you find
> > anything out, please let me know too.
> >
> > -Chris
>
> These phones are awsome based on a very limited trial.
>
> http://www.zyxel.com/product/P2000W.html

Those make me a little worried because we tried one of the WiSiP phones 
from Pulver Innovations, which are OEM'd versions of the 2000W's, and 
they were aweful, we couldn't get them configured, the documentation 
was virtually non-existant, and there were no support channels.  Have 
you had luck personally getting these to work?

-Chris
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RE: [Asterisk-Users] Asterisk 1.0.2, Zaptel 1.0.2, Linux 2.6.9 on a PCEngines WRAP\Soekris net4801 in Compact Flash

[Steven Critchfield]

On Tue, 2004-11-02 at 22:39 +0100, Goran Obradovic wrote:
> I had huge production problems with CF cards just a month ago. I used to
> live in Canada and still have some business there - electronic voting
> equipment. So, we have optical voting devices with 256, 512, and 1GB CF
> cards for election definitions and voting records. Last month on Alberta
> elections we had 5 Kingston 256MB cards failing during the elections. That
> was a nightmare. It is interesting that 512 and 1GB cards were ok even with
> more writes. In any case, if you do something like this first test some
> cards for long period of time. Make some script that will constantly write
> and read the card and see when they fail. 
> Goran

Reads are non destructive, writes produce wear and eventually blocks
start to fail. the 256meg modules you mention above where probably older
and used a different technology. 

Not to mention all flash memory runs in a round robin writing fashion,
so each block should hopefully get equal usage. The smaller cards just
happen to make the round trip faster than the larger cards.

Use of a decent OS and filesystem should help detect and avoid bad
blocks, but you eventually will end with failure just like any other
media.
-- 
Steven Critchfield <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] Asterisk 1.0.2, Zaptel 1.0.2, Linux 2.6.9 on a PCEngines WRAP\Soekris net4801 in Compact Flash

[Kristian Kielhofner]

Goran Obradovic wrote:
I had huge production problems with CF cards just a month ago. I used to
live in Canada and still have some business there - electronic voting
equipment. So, we have optical voting devices with 256, 512, and 1GB CF
cards for election definitions and voting records. Last month on Alberta
elections we had 5 Kingston 256MB cards failing during the elections. That
was a nightmare. It is interesting that 512 and 1GB cards were ok even with
more writes. In any case, if you do something like this first test some
cards for long period of time. Make some script that will constantly write
and read the card and see when they fail. 
Goran

Goran,
	The filesytem on the CF is mounted read-only (ext2).  No writes other 
than the initial flashing (hopefully).  That is why I built in 
res_config_odbc.  It also makes extensive use of ramfs for temp. 
filesystems.  Plus, if you had a 2.5" hd connected, you could use that 
for writes or a backup.  Oh yeh, the Soekris net4801 also has a USB port 
for a hard drive, etc...

	But thanks, I was under the impression that CF cards are okay as long 
as you don't write to them too often.  I will be testing them more now!

--
Kristian Kielhofner
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Re: [Asterisk-Users] The best SIP HW Phone and WLAN Phones forAsterisk

[Steve Totaro]

- Original Message - 
From: "Christopher TenHarmsel" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
<[EMAIL PROTECTED]>
Sent: Tuesday, November 02, 2004 4:18 PM
Subject: Re: [Asterisk-Users] The best SIP HW Phone and WLAN Phones 
forAsterisk


We've had good luck with the Cisco 7912s in a very similar situation
(office of 15 people), but to get SIP working you have to be able to
download the newest firmware from Cisco.  I think there's some info
about this on the voip-info.org wiki.
We haven't had any luck with wireless SIP phones, if you find anything
out, please let me know too.
-Chris
These phones are awsome based on a very limited trial.
http://www.zyxel.com/product/P2000W.html

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[Asterisk-Users] anyone got a 7910 to work with asterisk?

[Mark Phillips]

I've looked all over but can't find anything about the 7910.

Mark

-- 

Mark Phillips, G7LTT/KC2ENI
Randolph, NJ
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RE: [Asterisk-Users] Asterisk 1.0.2, Zaptel 1.0.2, Linux 2.6.9 on a PCEngines WRAP\Soekris net4801 in Compact Flash

[Goran Obradovic]

I had huge production problems with CF cards just a month ago. I used to
live in Canada and still have some business there - electronic voting
equipment. So, we have optical voting devices with 256, 512, and 1GB CF
cards for election definitions and voting records. Last month on Alberta
elections we had 5 Kingston 256MB cards failing during the elections. That
was a nightmare. It is interesting that 512 and 1GB cards were ok even with
more writes. In any case, if you do something like this first test some
cards for long period of time. Make some script that will constantly write
and read the card and see when they fail. 
Goran

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian Wilkins
Sent: Tuesday, November 02, 2004 5:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk 1.0.2, Zaptel 1.0.2,Linux 2.6.9 on a
PCEngines WRAP\Soekris net4801 in Compact Flash

Sounds cool, but I heard a rumor that CF Cards don't like too many rewrites
or 
they start losing data.

On Tuesday 02 November 2004 08:50 pm, Kristian Kielhofner wrote:
> Hello all,
>
>   Here is what I have so far:
>
> grub 0.94 (serial console)
> linux 2.6.9 (compiled for the Geode SC1100, with many, many modules)
> zaptel 1.0.2 (with ztdummy for 2.6)
> unixODBC with myODBC
> mysql
> perl
> ncurses
> full terminfo database
> OpenSSH
> perl5 (with modules)
> glibc
> full locale support
> full zoneinfo
> asterisk 1.0.2 with res_config and res_config_odbc modules
>
> This (and a lot more, I know that I am forgetting some stuff) fits in
> about 244mb right now.  I want to try to slim it down some more, but as
> a good 256mb CF card is under $40 right now that seemed like an okay
> size to be at.
>
> I plan on releasing this as a HD image and all of the source and a doc
> on how I did it (it was simple).  It is basically a Gentoo stage3 x86
> install with the kernel configured, asterisk added and some stuff removed.
>
> I have gotten it to boot and * runs, but I have done no performance
> testing or anything of the sort.  I plan on doing some tonight.  If
> anything right now it could be a sollution to my "branch office SIP
> phones -> IAX2 -> main office * (no transcoding)" idea.
>
> I feel that this combined with a Soekris net4801 + 256mb flash + case +
> 2.5" hd would make for a really cool Asterisk pbx for under $400 with
> semi-standard hardware.  Very DIY.
>
> What should I do with this?  Is anybody interested in any of this?  Has
> anyone done anything like this, either with these boards or ITX, etc.?
>
> Let me know, thanks!
>
> --
> Kristian Kielhofner
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-- 
--
Heritage Communications Corporation
  Melbourne, FL USA 32935
http://www.hcc.net
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RE: [Asterisk-Users] IAX between two *

[Barton Hodges]

Pedro Mansilla wrote:
> Hi,
> 
>I'm trying to connect two * with IAX. I can't call from one * SIP
> Extension to another * SIP extensions. 
> 
>Somebody have a sample about how I can config IAX.
> 
> Thanks,
> 
> Pedro.

http://lists.digium.com/pipermail/asterisk-dev/2003-October/001927.htm
l

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Re: [Asterisk-Users] Queue Prioritization

[Peter Svensson]

On Tue, 2 Nov 2004, TC wrote:

> > I have been thinking about adding inter-queue dependencies to ICD. It
> > seems better suited to more advanced queues than the built in queue
> > system.
> >
> > Another option in icd would be to keep all callers in one queue and then
> > pop the first customer whose profile matches the agent. That change seemed
> > not to be too hard to make.

> yah but now you serial all the load on that 1 thread & what happens when
> you want each q to have it normal strategy but only allowed to work
> based on the queuepriority ...this is not the way to go unless you want it
> quick & dirty

It has some advantages though. It is ideal if you want to be "fair", 
menaing all customers wait in the same line regardless of which language 
they speak. Then each time an agent becomes available it will check and 
pop the first matching customer. This way there can be no starvation in 
one queue. For our setup this really is the ideal behaviour, that is what 
we want to have.

You are right though, it is not several different prioritized queues. It 
is one queue with properties on the callers (agents andd customers). 

I was thinking of adding it as a property list on the callers and then 
creating a version of the link_via_pop function that uses 
something similar to icd_list__find to find the first matching customer 
for a given agent. 

Peter


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[Asterisk-Users] IAX between two *

[Pedro Mansilla]

Hi,

 

   I’m trying to
connect two * with IAX. I can’t call from one * SIP Extension to another
* SIP extensions.

 

   Somebody have a
sample about how I can config IAX.

 

Thanks,

 

Pedro.

 






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Re: [Asterisk-Users] The best SIP HW Phone and WLAN Phones for Asterisk

[Christopher TenHarmsel]

We've had good luck with the Cisco 7912s in a very similar situation 
(office of 15 people), but to get SIP working you have to be able to 
download the newest firmware from Cisco.  I think there's some info 
about this on the voip-info.org wiki.

We haven't had any luck with wireless SIP phones, if you find anything 
out, please let me know too.

-Chris

On Tuesday 02 November 2004 16:00, Goran Obradovic wrote:
> Can someone recommend the best combination of HW SIP phones and
> Asterisk PBX SW? The same question is for WLAN phones. Basically, I
> have a company with around 10 people and I am thinking to use
> Asterisk for our phone system. I would like to purchase SIP and WLAN
> phones for production use so I would like to receive some
> suggestions. Thanks.
>
> G

-- 
Chris TenHarmsel
Software Journeyman
Atomic Object, LLC
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Re: [Asterisk-Users] Re: How far is IAX to be a Standard

[Marcelo Pacheco]

> Think back,  The telco started with inline signaling.  Pulsing digits,
> cross bar switching, DTMF, CAS   'Yesterdays telco' took signaling
> out of band for a reason.  I'm not sure putting the signaling back into
> the bearer channel is a 'good thing'.

Telcos did that because they're dealing with fixed size samples, without clear 
packet boundaries, where adding a whole framing sub-protocol would be crazy.

However IP already gives you that framing, so you can distinguish between IAX 
voice packets and IAX signalling packets without a lot of complication.

The only thing in favor of your argument is CDR issues with IAX transfer.

Marcelo Pacheco
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Re: [Asterisk-Users] * and Verisign SIP-7 service

[Matthew Crocker]

My primary application is modem pool termination.  I want SS7 signals 
coming in via SIP to become MGCP control messages to tell an AS5400 to 
'Answer trunk 3 and give it a v.92 modem'.  Eventually I'll get into 
VoIP.

-Matt
On Nov 2, 2004, at 3:32 PM, Stewart Nelson wrote:
In what context will Asterisk will require proxying the media stream?

I have a simple setup whereby I make my FWD account ring my Mediatrix 
2102 as an extension to my Asterisk and the delay is horrific
If FWD is speaking IAX and the Mediatrix is SIP, * must remain in the 
loop,
even in theory.

If both are SIP, reinvite is enabled, and you meet the conditions
for reinvite to be applicable, media should bypass *.
Even though H.323 and MGCP have functionality equivalent to reinvite,
I believe that * does not support it.
Of course, if both sides are IAX, the transfer occurs automatically.
--Stewart
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[Asterisk-Users] Notification of missed calls

[Nick Barnes]

Hi all,

I've got so used to saying "Yes" before the question "Does Asterisk do"
is finished, I was surprised to be asked one I wasn't sure about, so here
goes...

I would like some form of e-mail notification to be sent when a call is
dropped before it's answered, or if it fails through to voicemail, but is
dropped before a message is left. Does anybody have any idea on the best way
to implement this?

Many thanks,

Nick.



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Re: [Asterisk-Users] Asterisk 1.0.2, Zaptel 1.0.2, Linux 2.6.9 on a PCEngines WRAP\Soekris net4801 in Compact Flash

[Brian Wilkins]

Sounds cool, but I heard a rumor that CF Cards don't like too many rewrites or 
they start losing data.

On Tuesday 02 November 2004 08:50 pm, Kristian Kielhofner wrote:
> Hello all,
>
>   Here is what I have so far:
>
> grub 0.94 (serial console)
> linux 2.6.9 (compiled for the Geode SC1100, with many, many modules)
> zaptel 1.0.2 (with ztdummy for 2.6)
> unixODBC with myODBC
> mysql
> perl
> ncurses
> full terminfo database
> OpenSSH
> perl5 (with modules)
> glibc
> full locale support
> full zoneinfo
> asterisk 1.0.2 with res_config and res_config_odbc modules
>
> This (and a lot more, I know that I am forgetting some stuff) fits in
> about 244mb right now.  I want to try to slim it down some more, but as
> a good 256mb CF card is under $40 right now that seemed like an okay
> size to be at.
>
> I plan on releasing this as a HD image and all of the source and a doc
> on how I did it (it was simple).  It is basically a Gentoo stage3 x86
> install with the kernel configured, asterisk added and some stuff removed.
>
> I have gotten it to boot and * runs, but I have done no performance
> testing or anything of the sort.  I plan on doing some tonight.  If
> anything right now it could be a sollution to my "branch office SIP
> phones -> IAX2 -> main office * (no transcoding)" idea.
>
> I feel that this combined with a Soekris net4801 + 256mb flash + case +
> 2.5" hd would make for a really cool Asterisk pbx for under $400 with
> semi-standard hardware.  Very DIY.
>
> What should I do with this?  Is anybody interested in any of this?  Has
> anyone done anything like this, either with these boards or ITX, etc.?
>
> Let me know, thanks!
>
> --
> Kristian Kielhofner
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-- 
--
Heritage Communications Corporation
  Melbourne, FL USA 32935
http://www.hcc.net
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[Asterisk-Users] The best SIP HW Phone and WLAN Phones for Asterisk

[Goran Obradovic]

Can someone recommend the best combination of HW SIP phones
and Asterisk PBX SW? The same question is for WLAN phones. Basically, I have a
company with around 10 people and I am thinking to use Asterisk for our phone
system. I would like to purchase SIP and WLAN phones for production use so I
would like to receive some suggestions. Thanks. 

G






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Re: [Asterisk-Users] Queue Prioritization

[TC]

> I have been thinking about adding inter-queue dependencies to ICD. It
> seems better suited to more advanced queues than the built in queue
> system.
>
> Another option in icd would be to keep all callers in one queue and then
> pop the first customer whose profile matches the agent. That change seemed
> not to be too hard to make.
yah but now you serial all the load on that 1 thread & what happens when
you want each q to have it normal strategy but only allowed to work
based on the queuepriority ...this is not the way to go unless you want it
quick & dirty

>
> Peter
yah we have  couple of ideas here in icd

1) create a new loadable icd module & serialize the calls for any queue that
has the
queuepriority= ? flag and have this master distributor trip the pop
event on the required q
-down side 1 thread now has to do a few ms of work might be an issue
with heavy volume
  but each q is still usng it own internal strategy & the work is still
mostly distributed over
  the threads for each q

2) have an election process where the distributors look at all other q's
that have the queuepriority flag
& talk to each other using the icd event system
load balances nicely, but lot more work

3) have each agent become a icd listener on all the q's they belong to that
have the queuepriority flag set
and only pop themselves back on the list for the queue that have
customer calls with the highest priority
load balances nice, distbutors dont have to change, might be a lot of
over head with a large number
of agents all sub-scribing to customer q add events

4) ???

still mulling it over :)



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[Asterisk-Users] Outgoing call fails on pulse dial line

[Goran Obradovic]

I have Digium FXO/FXS card and one of my phone lines is with
pulse dialing. At first I didn’t have dial tone at all, but after upgrading
my Asterisk and Zaptel SW to the latest one (1.0.2) I have dial tone. But, when
I try to dial outgoing number it fails after first key pressed. Does anyone
know how to solve this? I am in Eastern Europe (Belgrade, Serbia)
and our phone lines are mixture of old (pulse) and new ones. So I have 2 lines
in my house, one with tone dialing and one with pulse dialing. Is this related
to some signaling settings in Zapata.conf? 

Thanks,

Goran






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RE: [Asterisk-Users] Linux and Windows

[Michael Giagnocavo]

>> Yes, there is network support.
>> 
>> I tried it out, but the voice quality seemed quite choppy (local machine,
>P4
>> 3GHz). Not sure if it'd actually work for any near-production scenarios.
>> 
>> -Michael
>
>Being partly responsible for AstWind, the answer to that is a categorical 
>NO WAY! There is no way that anyone, in their right mind, should consider 
>using AstWind for ANY production level services. The CoLinux kernel 
>emulates interrupt timing, and as a result is not very accurate. Hence, 
>choppy audio when disk access happens etc..
>
>CoLinux works great for IAX to IAX or SIP to IAX where no Disk Access is 
>taking place.

Thanks for clearing that up. I had been using it for IVRs. So if I created a
RAM disk for CoLinux and booted it from there... that might work?

-Michael


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RE: [Asterisk-Users] Linux and Windows

[Greg Boehnlein]

yOn Mon, 1 Nov 2004, public wrote:

> Yes Sam, you can run just about any OS (except osX) in vmware, great to
> testing distros... incidentally, you can host virtual machines in both
> windows or linux. (meaning you can setup vmware on either platform and then
> run any OS you choose on that platform)

As a side note, you can now run OsX using the PearPC powerPC emulator:

http://www.pearpc.net/

I've done it. When I get some spare time, I'm going to try to get Asterisk 
build for MaxOSX on it. :)

-- 
Vice President of N2Net, a New Age Consulting Service, Inc. Company
 http://www.n2net.net Where everything clicks into place!
 KP-216-121-ST



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[Asterisk-Users] Asterisk 1.0.2, Zaptel 1.0.2, Linux 2.6.9 on a PCEngines WRAP\Soekris net4801 in Compact Flash

[Kristian Kielhofner]

Hello all,
Here is what I have so far:
grub 0.94 (serial console)
linux 2.6.9 (compiled for the Geode SC1100, with many, many modules)
zaptel 1.0.2 (with ztdummy for 2.6)
unixODBC with myODBC
mysql
perl
ncurses
full terminfo database
OpenSSH
perl5 (with modules)
glibc
full locale support
full zoneinfo
asterisk 1.0.2 with res_config and res_config_odbc modules
This (and a lot more, I know that I am forgetting some stuff) fits in 
about 244mb right now.  I want to try to slim it down some more, but as 
a good 256mb CF card is under $40 right now that seemed like an okay 
size to be at.

I plan on releasing this as a HD image and all of the source and a doc 
on how I did it (it was simple).  It is basically a Gentoo stage3 x86 
install with the kernel configured, asterisk added and some stuff removed.

I have gotten it to boot and * runs, but I have done no performance 
testing or anything of the sort.  I plan on doing some tonight.  If 
anything right now it could be a sollution to my "branch office SIP 
phones -> IAX2 -> main office * (no transcoding)" idea.

I feel that this combined with a Soekris net4801 + 256mb flash + case + 
2.5" hd would make for a really cool Asterisk pbx for under $400 with 
semi-standard hardware.  Very DIY.

What should I do with this?  Is anybody interested in any of this?  Has 
anyone done anything like this, either with these boards or ITX, etc.?

Let me know, thanks!
--
Kristian Kielhofner
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RE: [Asterisk-Users] Linux and Windows

[Greg Boehnlein]

On Mon, 1 Nov 2004, Michael Giagnocavo wrote:

> Yes, there is network support.
> 
> I tried it out, but the voice quality seemed quite choppy (local machine, P4
> 3GHz). Not sure if it'd actually work for any near-production scenarios.
> 
> -Michael

Being partly responsible for AstWind, the answer to that is a categorical 
NO WAY! There is no way that anyone, in their right mind, should consider 
using AstWind for ANY production level services. The CoLinux kernel 
emulates interrupt timing, and as a result is not very accurate. Hence, 
choppy audio when disk access happens etc..

CoLinux works great for IAX to IAX or SIP to IAX where no Disk Access is 
taking place.

-- 
Vice President of N2Net, a New Age Consulting Service, Inc. Company
 http://www.n2net.net Where everything clicks into place!
 KP-216-121-ST



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[Asterisk-Users] ISDN Dialplan

[Paulo Adriano]

  
  

  I need some help from you. I´m using Isdn4linux with Asterisk and incoming calls are working but anytime I whant to make an outgoing call I get this message.

 

    -- Executing Answer("Modem[i4l]/ttyI0", "") in new stack


  -- Executing Dial("Modem[i4l]/ttyI0", "SIP/21|10") in new stack


  -- Called 21


  -- SIP/21-aded is ringing


  -- Nobody picked up in 1 ms


  -- Executing Dial("Modem[i4l]/ttyI0", "Modem/g1/918708798") in new stack


  Nov  2 20:00:39 WARNING[1110502320]: chan_modem.c:191 modem_call: Destination g1/918708798 requres a real destination (device:destination)


  -- Couldn't call g1/918708798


  -- Hungup 'Modem[i4l]/ttyI1'


    == Everyone is busy/congested at this time


  -- Hungup 'Modem[i4l]/ttyI0'

 

  This is the  message for the calls out to the pstn

 

  -- Executing Dial("SIP/21-d99a", "Modem/g1:213570150") in new stack


  -- Called g1:213570150


  -- Modem[i4l]/ttyI1 is busy


  -- Hungup 'Modem[i4l]/ttyI1'


    == Everyone is busy/congested at this time


  Nov  2 20:35:56 WARNING[1110502320]: pbx.c:1933 ast_pbx_run: Timeout, but no rule 't' in context 'local-access'

 
 
 

  Well if you have some time give some clues on how to solve this issue. 

 

  Regards

 

  Paulo Adriano

  Francisco Paulo AdrianoWaveLIS LDAMobile +351 91 870 87 98Office + 351 21 989 83 34Fax +351 21 989 83 35E-mail  :  [EMAIL PROTECTED]

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RE: [Asterisk-Users] Unable to get our IP address, Skinny disabled

[Joel Berry]

I too have the same issue.  I "fixed" it by changing the skinny.conf file...

 
bindaddr = 10.169.208.11   ; Actual interface to bind to.
 
Instead of having the 0.0.0.0 address, I put the actual IP.. In your case
10.169.208.11.  This got rid of the problem for me.
 
May not be the right answer, but at least it works.  Currently using two
7910 Cisco phones very pretty good stability in 1.0.2.  Just limited on
features.  
 
Joel Berry


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Gunnar Þ.
Gestsson
Sent: Tuesday, November 02, 2004 2:50 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Unable to get our IP address, Skinny disabled


Hello.
 
I have just installed Asterisk on my HP DL140 Fedora Core 1 Server.  The
server has two interface cards active.  I have been unable to fix this error
although all DNS lookup works fine.
 
my /etc/host.conf contains
 
order bind,hosts
 
/etc/hosts contains
 
# Do not remove the following line, or various programs
# that require network functionality will fail.
127.0.0.1   localhost.localdomain localhost
10.169.208.11   asterisk1 asterisk1.fjolnet.is
asterisk1.simar.fjolnet.net

Asterisk also registers the loopback address in my Gatekeeper but not the
interface used to access the Gatekeeper.
 
Has anyone a solution for me?
 
Regards,
Gunnar Gestsson
 

 

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Re: [Asterisk-Users] * and Verisign SIP-7 service

[Stewart Nelson]

In what context will Asterisk will require proxying the media stream?

I have a simple setup whereby I make my FWD account ring my Mediatrix 2102 
as an extension to my Asterisk and the delay is horrific
If FWD is speaking IAX and the Mediatrix is SIP, * must remain in the loop,
even in theory.
If both are SIP, reinvite is enabled, and you meet the conditions
for reinvite to be applicable, media should bypass *.
Even though H.323 and MGCP have functionality equivalent to reinvite,
I believe that * does not support it.
Of course, if both sides are IAX, the transfer occurs automatically.
--Stewart
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Re: [Asterisk-Users] Tone while ringing another IAX Phone

[Christopher Stephens]

Simple, add ,r to your Dial command.
>From the wiki:
'r: Generate a ringing tone for the calling party, passing no audio from
the called channel(s) until one answers. Use with care and don't insert
this by default into all your dial statements as you are killing call
progress information for the user.'

On Tue, 2 Nov 2004 19:48:06 -, "Gunnar Þ. Gestsson"
<[EMAIL PROTECTED]> said:
> Hello
>  
> I have an IAX Phone installed on two Windows machines.  When dialling from one to 
> the other the user is not supplied with a dialling tone.  I maid Asterisk read a 
> notify to > the user but it is followed by a silence for up to 20 seconds.  Is there 
> a solution for this ?
>  
> Following is my extension for the IAX Phones.
>  
> exten => _45570XX,1,Playback(vm-dialout)
> exten => _45570XX,2,Dial(IAX2/${EXTEN}, 20)
> exten => _45570XX,3,Voicemail(u${EXTEN})
> exten => _45570XX,4,Hangup
> exten => _45570XX,103,Voicemail(b${EXTEN})
> exten => _45570XX,104,Hangup
> 
> Regards, 
> Gunnar Gestsson
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Re: [Asterisk-Users] Centrex

[Tim Sailer]

On Mon, Nov 01, 2004 at 04:26:26PM -0600, Ed Devine wrote:
> It shouldn't matter if the inbound line is centrex, it's just a phone line
> ringing into asterisk at that point. For the outbound side of the equation,
> you'll have to dial 9 (or other digit as defined by the system) get dialtone
> and send the dialed digits.
> 
> You might try:
> 
> exten => _9XX,1,dial(zap/g3/${EXTEN})
> 
> replace zap/g3 with whatever you're using to dial outbound. This should put
> you on the track to figuring out how to get centrex outbound working.

My problem is to get * to see the phone *ringing*, so it can be answered.
A standard phone can get calls (phone rings) and make calls (you canb break
dialtone by hitting a key, dial 9 to get out). Asterisk simply doesn't see
the line ring. The folks at the remote end are going to take the system
home where there is a standard POTS line, and see if that's any different.
Being 550 miles away makes diagnosis hard...

Tim

-- 
><
>> Tim Sailer   ><  Coastal Internet, Inc.  <<
>> Network and Systems Operations   ><  PO Box 726  <<
>> http://www.buoy.com  ><  Moriches, NY 11955  <<
>> [EMAIL PROTECTED] ><  (631) 399-2910 IAX 17003992910  <<
><
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[Asterisk-Users] Tone while ringing another IAX Phone

[Gunnar Þ. Gestsson]

Hello
 
I have an IAX Phone 
installed on two Windows machines.  When dialling from one to the other the 
user is not supplied with a dialling tone.  I maid Asterisk read a notify 
to the user but it is followed by a silence for up to 20 seconds.  Is there 
a solution for this ?
 
Following is my 
extension for the IAX Phones.
 
exten => 
_45570XX,1,Playback(vm-dialout)exten => _45570XX,2,Dial(IAX2/${EXTEN}, 
20)exten => _45570XX,3,Voicemail(u${EXTEN})exten => 
_45570XX,4,Hangupexten => _45570XX,103,Voicemail(b${EXTEN})exten 
=> _45570XX,104,Hangup
Regards, 

Gunnar 
Gestsson
 
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[Asterisk-Users] MAX TNT SIP / Asterisk

[James Taylor]

I can't get my MAX TNT to register with Asterisk.
TAOS 11.0.
SIP phone registeration show up in Asterisk like this:
 and works.
The TNT shows up as:
.
Does anyone have this working?
Am I missing something here?
Where does the TNT get it's user name?  Or, can it work without one?
Thanks,
James Taylor
MetroTel
903-793-1956
--
Using Opera's revolutionary e-mail client: http://www.opera.com/m2/
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Re: [Asterisk-Users] adding an artificial delay to *

[Francois Menard (Mailing List Account)]

Have you thought about the NIST flakeway project? its a network simulator 
...

f.
On Tue, 2 Nov 2004, Matt Riddell wrote:
Vahan Yerkanian wrote:
Greetings,
Is there a way to add artificial delay to the rtp stream? Due to 
regulations in our country, it is required to add 400ms delay to *some* 
VoIP calls.

Is this possible with any module?
Sorry I don't know if it is possible with a module (none that I know of), but 
you could simply route your calls via New Zealand, and then through the UK 
and then to US to terminate to destination.

This all begs the question [EMAIL PROTECTED]
Why do you have regulations in your country requiring you to make VOIP crap? 
Government owned telco?

Which country?
BTW:  Depending on your volume of calls, I might be able to offer you the New 
Zealand leg of your crazy journey!

:-)
--
Cheers,
Matt Riddell
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RE: [Asterisk-Users] Wireless VOIP Phone suggestions

[Kanuri, Seshu (Company IT)]

 Or SIPURA SPA 2000s

-Original Message-
Use an ATA the plug in any cordless phone.
Works fine.

--john

>Hi all,
>We're using Asterisk in our office to run our phone system (right now 
>about 5 SIP phones, various Cisco 7912's and 7960's), but we are in 
>desperate need for cordless phones.  We don't need 802.11b/g phones, 
>but just something that is wireless and does SIP.  I've done some 
>searching around, and we've even tried out the one from Pulver 
>Innovations (with no luck), so I wondered if someone could make some 
>suggestions?
>
>Thanks,
>Chris 

 
NOTICE: If received in error, please destroy and notify sender.  Sender does not waive 
confidentiality or privilege, and use is prohibited. 
 
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RE: [Asterisk-Users] Unable to find a path from GSM to SPEEX ??

[Francois Menard (Mailing List Account)]

what's the package to fetch? debian?
f.
On Tue, 2 Nov 2004, Kevin Walsh wrote:
Atuc [EMAIL PROTECTED] wrote:
does anybody know, how to enable the the new iaxcomm client (with speex
codec!!) to work with asterisk?
i get a "Unable to find a path" error??
i have enabled speex in iax.conf,
Have you compiled and installed SpeeX on your system?  Version 1.1.6
works well, although I'd guess that you don't have SpeeX installed at
all.
--
  _/   _/  _/_/_/_/  _/_/  _/_/_/  _/_/
 _/_/_/   _/_/  _/_/_/_/_/  _/   K e v i n   W a l s h
_/ _/_/  _/ _/ _/_/  _/_/[EMAIL PROTECTED]
_/   _/  _/_/_/_/  _/_/_/_/  _/_/
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RE: [Asterisk-Users] Remote Office question, Draytek , recommende d analog phone

[Steve Hanselman]

I'd back that comment, we've about 6 2600VG's, the wireless drops
frequently, it's not too hot on the voice side and the VPN's drop regularly,
or did until RC4, now they're stable but the box locks up once or twice a
day.

We're working to get them sorted out, but it looks as though they'll be
RMA'd fairly shortly.


-Original Message-
From: Voip Business [mailto:[EMAIL PROTECTED] 
Sent: 02 November 2004 19:13
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Remote Office question, Draytek , recommended
analog phone

Shawn...

I have too many problems with a customer with a Vigor2600V ( I have
sell something about 6) and all of them are not good (they looses
registration for some reazon)

Note: 

this happens with Asterisk, SER proxy, Snom Proxy and a Voicemaster SIP

Conclussion: NOT RECOMENDED


But the vigor 2300 and others are a reall good stuff for "security"
and VPN are really great.

By the way has someone in here connect a Sipura behind a Vigor 2200 -
2300 BB router?

Regards


HA

On Tue, 2 Nov 2004 10:08:20 -0700, Shawn Dillon <[EMAIL PROTECTED]>
wrote:
> 
> 
> 
> If there a way to connect an IP phone at a remote office to a device (like
> the Sipura 3000) that would allow local POTS failover in case of the
> Asterisk or VPN going down?
> 
>  
> 
> I know the Sipura would allow a analog phone to connect to the local POTS
in
> case of a * failure, we just want to standardize on IP phones ( Polycom
600)
> all around.
> 
>  
> 
> Also has anyone had experience with the Draytek VOIP Wireless routers?
> 
>  
> 
> And finally, if we need to use a Sipura 3000 in the remote offices is
there
> any benefit with going with a Sayson analog phone versus any other?
> 
>  
> 
> Thanks
> 
>  
> 
> Shawn Dillon
> 
>  
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[Asterisk-Users] isdn to isdn data call (bristuff'ed with hfc based card)

[Tomaz]

Hi,
anyone try to setup data call from one hfc-isdn card to another (across 
asterisk) ?

I can't establish isdn data call from isdn modem -> hfc card 
->asterisk->hfc card-> telco isdn line
voice call working great , only some echo problems.

any idea? maybe something in dialplan ?
tnx,
Tomaz
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Re: [Asterisk-Users] Linux and Windows

[steve szmidt]

Look guys, this is soo not an asterisk discussion. Please, let's take it off 
list...

-- 

Steve Szmidt

"They that would give up essential liberty for temporary safety 
deserve neither liberty nor safety."
Benjamin Franklin
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Re: [Asterisk-Users] Fw: Re: How far is IAX to be a Standard

[steve szmidt]

On Tuesday 02 November 2004 02:00 pm, Steve Totaro wrote:
> >> This thread was started by Randy Bush
> >
> > false
> >
> > but i love the personal attacks.  shows the desperation.
> >
> > randy
>
> I admit my mistake, you did not start the thread and I beg to differ about
> the personal attack.  At least, I was not personally attacking you.  I dont
> even know you, but I see you have quite an impressive resume.
>
> I was just pointing out some knee jerk reactions from more than one member
> that usually does not react that way.  That coupled with blogs like this
> http://cr.yp.to/djbdns/namedroppers.html does raise an eyebrow or two.

As interested as I am in DNS (and I'd love to see more talks about it) I 
however don't think it's fair to continue this discussion on this list. 

There's a certain urge to set ones name right, but that's done with one 
statment. Or not at all, if one is happy to know that one is right and don't 
really care what others think when they can't even bother to ask the person 
directly.
-- 

Steve Szmidt

"They that would give up essential liberty for temporary safety 
deserve neither liberty nor safety."
Benjamin Franklin
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Re: [Asterisk-Users] Remote Office question, Draytek , recommended analog phone

[Voip Business]

Shawn...

I have too many problems with a customer with a Vigor2600V ( I have
sell something about 6) and all of them are not good (they looses
registration for some reazon)

Note: 

this happens with Asterisk, SER proxy, Snom Proxy and a Voicemaster SIP

Conclussion: NOT RECOMENDED


But the vigor 2300 and others are a reall good stuff for "security"
and VPN are really great.

By the way has someone in here connect a Sipura behind a Vigor 2200 -
2300 BB router?

Regards


HA

On Tue, 2 Nov 2004 10:08:20 -0700, Shawn Dillon <[EMAIL PROTECTED]> wrote:
> 
> 
> 
> If there a way to connect an IP phone at a remote office to a device (like
> the Sipura 3000) that would allow local POTS failover in case of the
> Asterisk or VPN going down?
> 
>  
> 
> I know the Sipura would allow a analog phone to connect to the local POTS in
> case of a * failure, we just want to standardize on IP phones ( Polycom 600)
> all around.
> 
>  
> 
> Also has anyone had experience with the Draytek VOIP Wireless routers?
> 
>  
> 
> And finally, if we need to use a Sipura 3000 in the remote offices is there
> any benefit with going with a Sayson analog phone versus any other?
> 
>  
> 
> Thanks
> 
>  
> 
> Shawn Dillon
> 
>  
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[Asterisk-Users] AGI Help!

[Victor Cartes]

Hi everybody!
I've got a problem. When I call the method say_digits, or stream_file or 
get_data it seems not to work. Later I realized that the problem was that I 
get a "1" digit while I stream the audio file, and the method stop.

Does anybody know why I receive that number "1" if none press it?
Thanks in advance!!
Víctor 

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[Asterisk-Users] Fw: Re: How far is IAX to be a Standard

[Steve Totaro]

This thread was started by Randy Bush
false
but i love the personal attacks.  shows the desperation.
randy

I admit my mistake, you did not start the thread and I beg to differ about 
the personal attack.  At least, I was not personally attacking you.  I dont 
even know you, but I see you have quite an impressive resume.

I was just pointing out some knee jerk reactions from more than one member 
that usually does not react that way.  That coupled with blogs like this 
http://cr.yp.to/djbdns/namedroppers.html does raise an eyebrow or two.

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[Asterisk-Users] Polycom IP-500 Network Problems

[Matt Hohman]

We currently use about 40 Polycom IP-500's and 35 Grandstream Bt101's
at our company and have had the weirdest thing happen.  When the power
is unplugged to the supplied inline injector on the Polycom IP-500's
ALL traffic on our local network just "dies" no packets are able to
route any where. We are using 5 Netgear FSM726S (10/100 managed switch
with 2 GBIC ports)  Three of them are in a stack configuration linked
over a short haul fiber link (~700 - 1000 ft) to 2 more switches which
are in a stacked configuration. Any Idea's? Anyone heard of this
problem before?


Thanks,

Matt Hohman
New Heights Church
7913 ne 58th Ave
Vancouver, WA 98665
360 750-7112
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RE: [Asterisk-Users] Broadvoice with multiple numbers

[Seth Remington]

On Tue, 2004-11-02 at 11:29, Jay Milk wrote:
> I have two broadvoice numbers:
> 
> [incoming]
> exten => 16125551212,1,Macro(dialext_incoming,${EXT_BIZ},2000)
> exten => 14085551212,1,Macro(dialext_incoming,${EXT_ALL},1000)

You can also add the extension you want used for incoming calls in your
register statement.

register => user:secret:[EMAIL PROTECTED]:port/extension
register => 2345:[EMAIL PROTECTED]/1234

In the above example, define extension 1234 in extensions.conf in the
default SIP context and all incoming calls will land there.

-Seth


> -Original Message-
> From: Richard Cook [mailto:[EMAIL PROTECTED] 
> Sent: Tuesday, November 02, 2004 9:52 AM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] Broadvoice with multiple numbers
> 
> 
> Hey,
> 
> Is anyone using Broadvoice with multiple numbers?
> 
> Was wondering if there's a way to send each number to a different
> extension.  It seems that they both come into the same context.  You
> can't specify the dial plan based on the number, doesn't work.
> 
> Any ideas?
> 
> --
> Richard Cook
> [EMAIL PROTECTED]
> Tel: 705-497-9320  ext 2010

-- 
Seth Remington
SaberLogic, LLC
661-B Weber Drive
Wadsworth, Ohio 44281
Phone: (330)335-6442
Fax: (330)336-8559

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Re: [Asterisk-Users] ISDN EDSS1 protocol support

[Francois Menard (Mailing List Account)]

Does ZapRAS allow you to serve several incoming modem calls for dial-up 
internet users, as an ISP in the good old days?
f.

On Sun, 31 Oct 2004, Martin List-Petersen wrote:
On Fri, 2004-10-29 at 13:41, Maxim Litnitsky wrote:
Hi all, I have to implement the following:
--
  |    10 voice channels   >
|---|
Prov  E1 |     256 kbit/s for VoIP >  |
Asterisk IP-PBX  |
  |     256 kbit/s for Data (http,mail) ->
|---|
--
Provider gives E1 and on this E1 I will have 10 timeslots for voice,
and others for internet.
What hardware shall I use? Provider supports EDSS1 ISDN protocol, as
I undertood Digium hardware does not support this protcol. I searched
google and lists.digium.com and found only
this:
http://www.redhat.com/archives/fedora-list/2004-October/msg03224.html
http://www.mail-archive.com/[EMAIL PROTECTED]/msg30870.html
The question: Can I implement all with Asterisk using EDSS1 protcol
and how?  Give me please a clue!!
You didn't define your question good enough.
Digium hardware does support EDSS1 (EuroISDN) without problems. However,
you didn't say, how your provider let you connect to the internet.
You have 30 channels on your E1 (30 timeslots / 64 kbit), not counting
the d-channel, which is a total of 2 mbit.
Implenting the 10 voice channels is a std. setup, but your provider
still needs to tell you, how you access the internet/data part. EDSS1 is
only a ISDN signalling protocol, you would probably have to run
something like PPP over the lasting 20 channels (or how many your
provider has assigned there) to get connectivity. Get better
specifications from your provider !!!
If your provider indeed is using PPP, then you should have a look at
ZapRAS in Asterisk (http://www.voip-info.org/wiki-Asterisk+cmd+ZapRAS)
Kind regards,
Martin List-Petersen
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RE: [Asterisk-Users] Unable to create RTP session: Too many open files(was: no subject)

[Ryan Courtnage]

On Tue, 2004-02-11 at 14:02 -0500, Gerardo Bassett wrote:
> I tried that, I raised it up too 100,000 and still the same problem.  
> 
> Another symptom is that I get dial tone from *, it allows me to dial and I
> get a ring back but the other phone does not ring at all.  Could I be
> missing some kind of configuration somewhere???

Possibly.  Turn on 'sip debug' at the * CLI (alternatively, use
Ethereal).  Have a very close look at the addresses being used in the
SIP headers between Asterisk and the client.   

Hopefully this will reveal the source of the problem.

Regards,

-- 
Ryan Courtnage
Director & CTO
Coalescent Systems Inc
403.244.8089
www.coalescentsystems.ca

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Re: [Asterisk-Users] ISDN card advise

[Paulo Adriano]

  
  

  Simon,


   


  I bought the Conceptronic ISDN PCI Card . Linux reports the card as a "Dynalink 6692 PCI"


   


  It looks like it´s suitable for the isdn4linux drivers. I´m still trying to configure it. Incoming calls are working but I´m still fighting for outgoing calls with this card.


   


  If you have AVM or EIcon card close maybe you can get better support from the guys in germany, it looks like they are developing very nice capi drivers for the AVM /EICON . http://www.junghanns.net/asterisk/


   


  This last two brands are more expensive that the one I bought .


   


   


  >>>Simon Tennant <[EMAIL PROTECTED]> 11/02 9:11 am >>>

  
   
  
  
  Sounds like a nice setup - I was planning on doing the same.  Which card


   


  did you decide on?

  
   
  
  
   
  
  
  S.


   


   


   


  On Sun, Oct 31, 2004 at 09:21:33PM +, Paulo Adriano wrote:

  
   
  
  
  >Hi,


   


  > 

  
   
  
  
  >I need an advise for a ISDN card for my HomeOffice Asterisk Setup.


   


  >Currently I started with a couple of x100p for two anolog lines coming

  
   
  
  
  >from a ISDN NT.  Works but on bridged calls  the sound quality is bad


   


  >and distortion,  if the call is being routed from the pstn back to pstn

  
   
  
  
  >on the second line.


   


  > 

  
   
  
  
  >My setup is very simple. If a call comes from the pstn our internal


   


  >extension ..rings 4 times in my SIP phone and if no answer goes to my

  
   
  
  
  >mobile phone using the pstn.


   


  > 

  
   
  
  
  >Now it s time to go shooping for a simple ISDN card an I need an advise


   


  >regarding my simple requirements.  Please advise with some options.

  
 

Re: [Asterisk-Users] Linux and Windows

[Benjamin on Asterisk Mailing Lists]

On Tue, 02 Nov 2004 10:42:10 -0500, Jason Becker
<[EMAIL PROTECTED]> wrote:
> You know, OpenVMS administrators would probably wet their pants laughing
>   at the rhetoric coming from the Linux zealots among us.
> 
> http://itmanagement.earthweb.com/erp/article.php/3380341

In a former life I used to be one of those VMS heads and it always
puzzles me how MSFT managed to so utterly and completely botch their
VMS reengineering project (the making of Windoze NT).

MSFT got their hands on the dream team from DEC including VMS' chief
architect, Dave Cutler. MSFT also had all the resources and plenty of
cash. So, how on earth is it possible to come up with a VMS clone
that's so utterly the opposite of everything VMS stands for?

One might be inclined to think MSFT would have done much better to
outsource the job to the Russians after the fall of the iron curtain.
During the cold war, they used to clone just about everything DEC did.
All those missiles we were so worried about pointing at us, they were
all engineered on Russian and Eastern German cloned VAXes and
disassembled then rebuild and localised Russian VMS system software.

Heck, if McDonalds or the Heinz Ketchup company had got their hands on
the VMS dream team and chief architect, I'm sure they would have come
up with a decent clone that would have preserved the reliability,
stability, security and consistency of VMS at least to some degree.

So, what on earth happened at MSFT that they botched this so
badly. It's puzzling.

rgds
benjk

-- 
Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya,
Tokyo, Japan.

NB: Spam filters in place. Messages unrelated to the * mailing lists
may get trashed.
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RE: [Asterisk-Users] This is VERY interesting -- A gatewaybetweenproprietary digital sets and SIP?

[Francois Menard (Mailing List Account)]

The Citel device is more expensive than replacing all the phones with the 
new business Bugetones...
f.

On Sat, 30 Oct 2004, Jim Van Meggelen wrote:
Thanks. I'll have to see about that SIP functionality.

[EMAIL PROTECTED] wrote:
I have delt with their 3com offerings and yes if you are
lucky enough to be
able to use this as a stepping stone solution then its a closed deal
(on 3com system)
- Original Message -
From: "Jim Van Meggelen" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Saturday, October 30, 2004 1:00 AM
Subject: [Asterisk-Users] This is VERY interesting -- A gateway
betweenproprietary digital sets and SIP?

Has anyone had any experience with these folks?
http://www.citel.com/index/index.asp
That could be a compelling way to displace a legacy system with an
Asterisk.
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Re: [Asterisk-Users] * and Verisign SIP-7 service

[Francois Menard (Mailing List Account)]

So, I think that Asterisk will provide the functionality that you
desire.  However, I don't know if SIP<->MGCP calls can presently
be completed without Asterisk proxying the media stream, so you
may have performance issues.  Perhaps someone else can address
that.
In what context will Asterisk will require proxying the media stream?
I have a simple setup whereby I make my FWD account ring my Mediatrix 2102 
as an extension to my Asterisk and the delay is horrific

f.
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RE: [Asterisk-Users] Unable to find a path from GSM to SPEEX ??

[Atuc]

At 19:17 02.11.2004, you wrote:
At 17:45 02.11.2004, you wrote:
Atuc [EMAIL PROTECTED] wrote:
> does anybody know, how to enable the the new iaxcomm client (with speex
> codec!!) to work with asterisk?
>
> i get a "Unable to find a path" error??
>
> i have enabled speex in iax.conf,
>
Have you compiled and installed SpeeX on your system?  Version 1.1.6
works well, although I'd guess that you don't have SpeeX installed at
all.
thanks Kevin,
that sounds plausibly,
but is it true that i have to install speex as system libs? also ilbc? i 
thought asterisk has all this stuff in the distribution, i run here 
another asterisk and ilbc is working out of the box without installation.
you are right, it helps,
thanks,
alex 

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RE: [Asterisk-Users] Unable to create RTP session: Too many open files(was: no subject)

[Gerardo Bassett]

I tried that, I raised it up too 100,000 and still the same problem.  

Another symptom is that I get dial tone from *, it allows me to dial and I
get a ring back but the other phone does not ring at all.  Could I be
missing some kind of configuration somewhere???

Jeroocko

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason Becker
Sent: Tuesday, November 02, 2004 10:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Unable to create RTP session: Too many open
files(was: no subject)

Gerardo Bassett wrote:

> I'm testing * with two XLite softphones, they are all running within a
local
> network.  I was able to install * and register the phones to it.  But when
I
> dial the other extension I get this messages:
> 
> Connected to Asterisk CVS-HEAD-10/25/04-23:24:03 currently running on
> localhost (pid = 3702) 
> Nov  2 10:32:14 WARNING[1350441904]: rtp.c:846 ast_rtp_new_with_bindaddr:
> Unable to allocate socket: Too many open files Nov  2 10:32:14
> WARNING[1350441904]: chan_sip.c:2284 sip_alloc: Unable to create RTP
> session: Too many open files 
> Nov  2 10:32:14 WARNING[1350441904]: chan_sip.c:7967 sip_request: Unable
to
> build sip pvt data for '[EMAIL PROTECTED]'
> Nov  2 10:32:14 NOTICE[1350441904]: app_dial.c:744 dial_exec: Unable to
> create channel of type 'SIP'
> 
> 
> Does anyone have any idea as to why this is happening?  

Look in /etc/asterisk/rtp.conf

I think the range is too narrow.

Please include a Subject line in your next post.

Regards,



-- 
Jason Becker
Director & CEO
Coalescent Systems Inc.
403.244.8089
www.coalescentsystems.ca
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Re: [Asterisk-Users] Zaptel Issue in Fedora Core 2 test 3

[kwijibo]

Three things you may want to look at.
1. Do those devices exist?
2. Does the user that is starting Asterisk have permission to
   access those devices?
3. I *think* FC3 comes with SELinux on by default.  This may
   or may not be getting in your way.
Steve
James Botham wrote:
Hi,
 
I have the following error once I have installed asterisk on my new 
Fedora Core 2 test 3 box with a T100P, compiled fine no problems after 
using the info on voip-info.org but when i run ztcfg i get this error 
message:
 
Notice: Configuration file is /etc/zaptel.conf
line 143: Unable to open master device '/dev/zap/ctl'
The cards appear to load fine when i modprobe them  no errors appear but 
Asterisk cannot use them when I make a call to the line in question 
asterisk doesn't acknowledge them, the config I am using is off my old 
machine (identical version same hardware and card just new OS) no errors 
on its startup euitither it seems to accept the config fine. | suspect 
that the channel has possibly changed but without ZTCFG i don't know how 
to check, does anybody have any ideas on a) what this message means and 
B is there anyway to see the channel the device is using.
 
Cheers

James Botham
Client Support Consultant
*Computer Software Group plc*
_www.computersoftware.com _
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Re: [Asterisk-Users] adding an artificial delay to *

[Peter Svensson]

On Tue, 2 Nov 2004, steve szmidt wrote:

> > It is quite true for some classes of batteries. E.g. some Li-ion batteries
> > will explode if charged (or in the case of rechargeable batteries charged
> > with the wrong voltage / polarity). They pack quite a punch as well. The
> > normal household alkaline batteries are safe as far as I know though.
> 
> And at that you need enough amperage to blow it. Normal charge is not enough, 
> your charger should blow a fuse long before. Not what you find in normal home 
> charging equipment.

For non-rechargeable Li-ion cells the required energy input can be 
relativly small, if they are not internally protected. Those designed for 
soldering on to a pcb may be unprotected as the required circuitry is 
expected to be on the pcb. 

The ones I have encountered were safe even if shorted, but not if charged. 
Even a relatively small charging current can initiate a runaway energy 
release. It all depends on what kind of battery it is. I would expect all 
household batteries to be safe or we would have heared of a lot more 
explosions. 

If it is dangerous, they will be dumb.

Peter


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