date:20060522

Hi Paul,

Asterisk often uses a proxy for its calls. What kind of proxy do you
have?

Also, If you have the server setup for nat=yes in the [general] area
then ALL calls will get nat'd regardless of their locality. The best
place to put this stement is in the relevant part of the sip.conf file
that deals with the devices you want to have nat'd.

If you only want to nat devices from your office LAN but not devices or
service providers out on the Internet then you need to do a bit more
configuration.

I've pasted my config file below for your perusal. I have phone handsets
on the LAN but my phone provider is on the Internet. I don't nat
internally but do externally.

; SIP Configuration for Asterisk
;
[general]
port = 5060   ; Port to bind to
bindaddr = 192.168.201.15 ; Address to bind to
localnet = 192.168.201.0/24   ; Internal NETWORK address
;externhost = g7ltt.dyndns.org ; Address for NAT'd SIP messages
;externrefresh = 10
externip = 68.196.143.250
nat = no
srvlookup = yes   ; Enable DNS lookups
context = from-sip-external
dtmfmode = inband
disallow = all
allow = ulaw
allow = g726
allow = gsm
allow = ilbc
tos = lowdelay
canreinvite = no
pedantic = no
videosupport = yes
callerid = 9738281625
qualify = yes
realm=g7ltt.dyndns.org

; put external SIP provider registration here
register = user:@sip.broadvoice.com:password:[EMAIL PROTECTED]

[2201] ; Mark's MDA
type=friend
host=dynamic
context=from-sip-internal
username=2201
secret=blah
dtmfmode=rfc2833
mailbox=2201
disallow=all
allow=gsm

[2202] ; WiFi cordless
type=friend
host=dynamic
context=from-sip-internal
username=2202
secret=blah
dtmfmode=rfc2833
mailbox=2202
callgroup=1
pickupgroup=1

[2203] ;
type=friend
host=dynamic
context=from-sip-internal
username=2203
secret=blah
dtmfmode=rfc2833
mailbox=2203
callgroup=1
pickupgroup=1

[sip.broadvoice.com] ; main outgoing provider
user=phone
username=9738281625
type=peer
secret=password
nat=yes
insecure=very
host=sip.broadvoice.com
port=5060
fromuser=9738281625
fromdomain=sip.broadvoice.com
dtmfmode=inband
context=enicommunications
canreinvite=no
authname=9738281625
qualify=1000
disallow=all
allow=ulaw
allow=g726
allow=ilbc

You'll notice that nat=no is set in my [general] area. That means that
unless I say otherwise all devices are considered local and so no nat
required. In  the [sip.broadvoice.com] area I turn on the nat. I think
that in your case you do the reverse of this.

I'm on the end of a cable modem and so I *should* use the externhost
settings as my number could change dynamicly but as I've found that it
never does save myself the DNS lookup.

Hope this helps.

Mark

On Mon, 2006-05-22 at 02:55 -0700, Paul David wrote:
 Good Day All
 I recently implemnetd asterisk  outside our LAN (external network).It
 works well in a NAT settings.
 But on external network with PROXY setting ASTERISK DID NOT WORK.
 
 My question are 
 1 Can ASTERISK work in a PROXY setting .
 2 If it can work how can i implement it .
 
 Expecting your reply 
 Thanks 
 
 Paul 
 
 
 
 
 __
 Yahoo! Messenger with Voice. PC-to-Phone calls for ridiculously low
 rates.
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Re: [Asterisk-Users] VoiceMail Groups

2006-05-22 Thread picciuX

it's simple:exten = x,n,VoiceMail([EMAIL PROTECTED][EMAIL PROTECTED][EMAIL PROTECTED])the messages played in this case, if the corresponding options are specified ('u' or 'b'), are the ones for the first mailbox specified.
If the first mailbox has delete=yes option in voicemail.conf, the message, after broadcast to the other mailboxes, will be deleted from it, otherwise it will stay also in the first mailbox.It's all, more or less...
HTH2006/5/18, Forrest Beck [EMAIL PROTECTED]:













Has anyone seen good scripts or documentation on Voicemail
groups? We are looking to have a system where you can send a voicemail to
multiple mailboxes.











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Re: FW: [Asterisk-Users] WiFi / GSM VoIP Handsets..

As I understand it, the device uses either GSM or VoIP to access the
carrier? Which cell phone carrier supports GSM and VoIP in the EU?

They've been punting this thing around the shows in the US for a couple
of years now but none of the carriers support it. With GSM having such
blanket coverage I don't see many carriers going this way. I can
understand this working in Asia where coverage is only in the majorly
populated areas and even then only outside.




On Mon, 2006-05-22 at 16:53 +0800, Sam Tam wrote:
  
 
 Well I think we all need to look at something like this first.
 We will be one of the first people in Europe who will be selling this.
 If anyone is interested do drop me an email.
 
  
 
 Picture of the phone can be found here.
 
 http://cyber-telecom.net/wifi-gsm.jpg
 
  
 
 GSM / VoIP Over WiFi Dual-Mode Phone
 
 CYBER-TELECOM released the world first commercial GSM/VoIP Over WiFi
 dual-mode smart phone, in March 2006. With a tri-band GSM/GPRS (Class
 10) radio and an IEEE 802.11b WLAN chipset, It enables end customers
 to enjoy broadband multimedia services at WLAN covered homes, offices,
 hot spots/zones as well as reliable GSM/GPRS service anytime anywhere.
 It shows an outstanding performance in power management, mobility
 management, security, mobile VoIP, and voice quality, no matter what
 kind of access points it connects, as the result of CYBER-TELECOM
 Wireless's advanced technologies solving the critical problems of VoIP
 Over WiFi. the phone has passed most of regulation certification
 programs and has done interoperability testing with over 40 VoIP
 service providers, system integrators, and infrastructure equipment
 vendors worldwide. the phone is an ideal device for fixed mobile
 convergence. 
 
 Hardware Specification
 
 Intel PXA271 processor with embedded Linux
 2.4 inch TFT touch screen, QVGA, 260k Colors
 Built-in speaker/microphone, 2.4mm stereo and headset
 1.3M pixel CMOS camera
 USB slave 
 Mini SD
 1100 mAh Li-ion battery 
 
 GSM Specification
 
 Frequency bands: 900/1800/1900 MHz
 GPRS Class 10
 SMS, MMS, WAP applications
 FTA/CTA certification
 FCC/CE certification
 
 WLAN Specification
 
 IEEE 802.11b 
 RF channels: US: 11, ETSI: 13, Japan: 14
 High-gain internal antenna
 WEP 64/128 bits, WPA, 802.1x
 EAP PSK/LEAP/PEAP/TTLS/SIM
 Power saving modes
 Fast roaming between access points
 
 VoIP Specification
 
 SIP: IETF RFC 3261
 Codec: G.711, G.729a/b, G.723 
 Acoustic echo cancellation
 Dynamic jitter buffer
 Voice activity detection
 Stun-based NAT traversal
 
 Input Methods
 
 Handwriting Recognition
  English
  Chinese
  Numeric characters
 Soft Keypads
  Qwerty
  Standard phone dialpad
  Symbol
 
 Power Management Features
 
 Standby time
 100 Hours (GSM on, WLAN on)
  200 Hours (GSM on, WLAN off)
 Talk time
  VoIP Over WiFi: 3.3 Hours
  GSM: 7.8 Hours
 MP3 play time
  5.8 Hours (GSM on, WLAN on)
  6.2 Hours (GSM on, WLAN off)
 
 Fixed Mobile Convergence Features
 
 Simultaneously activated GSM and WLAN air interfaces
 Handling simultaneously GSM and VoIP Over WiFi incoming calls
 SIP-based seamless handover between GSM/VoIP Over WiFi
 Automatic/manual switch for out-going calls between GSM and VoIP Over
 WiFi
 Automatic/manual switch for data applications using GPRS or WLAN
 Unified phone book for both GSM and VoIP Over WiFi.
 Unified GUI for applications (phone, E-mail, browser, QQ)
 
 Call Features
 
 Call hold
 Call waiting
 Call mute
 Call forward
 Call transfer
 3-way conference
 Voice mail
 SMS over SIP
 Phone book - (1000 entries with photos)
 Incoming call prompt with picture
 View phonebook during call
 Enter sketch pad during call
 Adjust volume during call
 Auto-answer/flip answer
 Quick silence
 Turbo dial
 Manual/Auto/Earphone redial
 Call history (20 entries)
 
 Data Application Features
 
 POP3 E-mail client (SSL support)
  100 full E-mails with attachments up to 200KB
  Document viewer for MS-Office and PDF files
 Web browser: HTML4.01, JAVAScript1.5, SSL3.0, HTTP1.1, CSS1.0
 Instant messaging: QQ
 
 Multimedia Features
 
 Video format: MP4, 3GPP
 Audio format: MP3, WAV, MIDI, AMR
 Picture format: WBMP, BMP, JPEG, GIF
 Camcorder: QVGA, QCIF
 Media Player
  Audio: MP3 player
  Video: up to 30 frames/second QVGA MP4/3GPP
 
 PIM Features
 
 Calendar
 Schedule management
 Alarm clock
 Voice recorder
 World time
 Currency converter
 Anniversary
 
 Other Features
 
 English - Chinese dictionary
 Calculator
 World time
 Notepad
 Sketch pad
 File transfer
 Counter
 Timer
 
  
 
 
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Re: [Asterisk-Users] SIP to IAX - forcing codec pass thru

Hi Peter,

I don't see any codec allow=blah statements. If your end user has
something like

[gradwell]
disallow=all
allow=gsm

Then you'll be forced to send them a GSM coded call. 

Why not force the codec at your end by only supporting one? If the
customer then transcodes the call when it gets forwarded to his handset
there's not much you can do about that but at least you'll have handed
the call off in the best format you can source.

Mark

On Mon, 2006-05-22 at 09:57 +0100, Peter Gradwell wrote:
 hi
 
 We take calls inbound via SIP from our Cisco PSTN gateways, and pass it 
 to customers using IAX (they run their own asterisk servers).
 
 We've noticed that asterisk is transcoding the call into a different 
 codec, if the customer prefers a codec different to that which our cisco 
 gw prefers. As such, the quality of the call can degrade.
 
 We'd rather asterisk just passed through the RTP stream and maintained 
 the same codec, so that all asterisk did was signalling conversion.
 
 
 
 sip.conf...
 
 ---
 
 [sip-router-1.gradwell.net]
 context=sip-inbound
 type=peer
 host=sip-router-1.gradwell.net
 
 [sip-router-2.gradwell.net]
 context=sip-inbound
 type=peer
 host=sip-router-2.gradwell.net
 
 ---
 
 iax.conf...
 
 [general]
 bandwidth=high
 disallow=lpc10
 jitterbuffer=yes
 dropcount=2
 maxjitterbuffer=500
 maxexcessbuffer=80
 minexcessbuffer=10
 jittershrinkrate=1
 tos=lowdelay
 
 
 ---
 
 when a call comes in, we dial something like this, in our dial plan:
 
  -- Executing Goto(SIP/213.166.5.134-118f5310, 
 sip-users|7770002|1) in new stack
  -- Goto (sip-users,7770002,1)
  -- Executing Dial(SIP/213.166.5.134-118f5310, 
 IAX2/user:[EMAIL PROTECTED]/441376350002) in new stack
  -- Called user:[EMAIL PROTECTED]/441376350002
  -- Call accepted by customerip (format alaw)
  -- Format for call is alaw
  -- IAX2/customerip:4569-23 answered SIP/213.166.5.134-118f5310
 
 thanks
 peter
 

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[Asterisk-Users] Recommended SIP phones?

2006-05-22 Thread Pieter Claassen

I am dying here with linphone (not sure if it is crap software or just me 
being an idiot) but out of the box debian installations of two linphones fail 
with a Got SIP response 415 Unsupported Media Type back from 192.168.1.3

Can anybody recommend a particular SIP soft phone that broadly satisfies the 
following criteria?
1. Run on linux.
2. Simple to use and setup.
3. Is preferably packaged in APT.
4. OSS compliant license.

Any recommendations (or let me know if I am doing something wrong with 
linphone) are welcome.

Thanks
Pieter
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Re: [Asterisk-Users] WiFi VoIP Handsets..

Don't ya just love living in this technological backwater they call the
USA? DECT technology was released almost 20 years ago. In most of the
world it's been and gone.

Anyone in the UK or Hong Kong remember Rabit and having to find a Green
Dot Hot Spot by the train station or post office? When you got home the
thing would mysteriously become part of your home phone system. 


On Sun, 2006-05-21 at 18:42 -0500, Eric ManxPower Wieling wrote:
 Are any of these FCC licensed for use in the USA.  DECT in the USA is 
 VERY new.
 

I believe that DECT is approved for use here. Either that or Staples et
al are selling loads of illegal multi handset DECT phones. Some with
VoIP some without.

Mark

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Re: [Asterisk-Users] VoiceMail Groups

2006-05-22 Thread Filip Drągowski





Look into Your mail soft configration.
I'm using postfix , and made in aliases file :
    allusers: user1,user2,ser3,user4

then send voicemail from asterisk to alluser account.

Filip.

  
  
  

  
  Has anyone seen good
scripts or documentation on Voicemail
groups?  We are looking to have a system where you can send a voicemail
to
multiple mailboxes.
   
   
  
  

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[Asterisk-Users] how to customize voicemail

Is there any way to customize VoiceMail ?
I would like to customize the message played to callers sent to the
voicemail becouse the extension is busy or otherwise unavailable.
Is it a way to record a welcome message and use it ?

thanks in advance,

Andrea


Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla.

Visitate il sito http://www.frameweb.it

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Re: [Asterisk-Users] SIP to IAX - forcing codec pass thru

2006-05-22 Thread bails

Hi Peter, as one of your customers I would ask you not to dissallow g729 
on IAX2  as we currently use it extensively.


Bails

Mark Phillips wrote:

Hi Peter,

I don't see any codec allow=blah statements. If your end user has
something like

[gradwell]
disallow=all
allow=gsm

Then you'll be forced to send them a GSM coded call. 


Why not force the codec at your end by only supporting one? If the
customer then transcodes the call when it gets forwarded to his handset
there's not much you can do about that but at least you'll have handed
the call off in the best format you can source.

Mark

On Mon, 2006-05-22 at 09:57 +0100, Peter Gradwell wrote:


hi

We take calls inbound via SIP from our Cisco PSTN gateways, and pass it 
to customers using IAX (they run their own asterisk servers).


We've noticed that asterisk is transcoding the call into a different 
codec, if the customer prefers a codec different to that which our cisco 
gw prefers. As such, the quality of the call can degrade.


We'd rather asterisk just passed through the RTP stream and maintained 
the same codec, so that all asterisk did was signalling conversion.




sip.conf...

---

[sip-router-1.gradwell.net]
context=sip-inbound
type=peer
host=sip-router-1.gradwell.net

[sip-router-2.gradwell.net]
context=sip-inbound
type=peer
host=sip-router-2.gradwell.net

---

iax.conf...

[general]
bandwidth=high
disallow=lpc10
jitterbuffer=yes
dropcount=2
maxjitterbuffer=500
maxexcessbuffer=80
minexcessbuffer=10
jittershrinkrate=1
tos=lowdelay


---

when a call comes in, we dial something like this, in our dial plan:

-- Executing Goto(SIP/213.166.5.134-118f5310, 
sip-users|7770002|1) in new stack

-- Goto (sip-users,7770002,1)
-- Executing Dial(SIP/213.166.5.134-118f5310, 
IAX2/user:[EMAIL PROTECTED]/441376350002) in new stack

-- Called user:[EMAIL PROTECTED]/441376350002
-- Call accepted by customerip (format alaw)
-- Format for call is alaw
-- IAX2/customerip:4569-23 answered SIP/213.166.5.134-118f5310

thanks
peter




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[Asterisk-Users] Got reject for frame 0, but we only have others!

2006-05-22 Thread Sebastian Kayser

Hi all,

what could be the cause for the following messages? 

May 22 12:44:53 WARNING[3791]: chan_zap.c:8498 zt_pri_error: 2 !! Got reject 
for frame 0, but we only have others!
May 22 12:44:54 WARNING[3791]: chan_zap.c:8498 zt_pri_error: 2 !! Got reject 
for frame 0, but we only have others!
May 22 12:44:55 WARNING[3791]: chan_zap.c:8498 zt_pri_error: 2 !! Got reject 
for frame 0, but we only have others!
May 22 12:44:56 WARNING[3791]: chan_zap.c:8498 zt_pri_error: 2 !! Got reject 
for frame 0, but we only have others!
May 22 12:44:57 WARNING[3791]: chan_zap.c:8498 zt_pri_error: 2 !! Got reject 
for frame 0, but we only have others!

They flood my asterisk log since some days and i can't relate the date
they started to a specific configuration change. I haven't yet noticed
any bad side-effects, but i tend to feel better without any WARNINGs
in my * log. 

I have already enabled bri intense debug span 2, but i have to admit
that i am not able to recognize any valuable pattern not to mention any
sense at all :( (maybe that's because i am a systems administrator and
not a telco guy). 

At least there is some kind of BRI traffic each time such a message is
logged. The BRI debug can be found at 

http://skayser.de/mls/au/reject-bri-intense-debug.txt

Maybe some of you are more capable to interpret those cryptic BRI
messages.

asterisk*CLI show version
Asterisk 1.2.7.1-BRIstuffed-0.3.0-PRE-1p

Cheers

- Sebastian
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RE: FW: [Asterisk-Users] WiFi / GSM VoIP Handsets..

2006-05-22 Thread Dean Collins

Mark, how ignorant are you?

Dean


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark
Phillips
Sent: Monday, 22 May 2006 6:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: FW: [Asterisk-Users] WiFi / GSM VoIP Handsets..

As I understand it, the device uses either GSM or VoIP to access the
carrier? Which cell phone carrier supports GSM and VoIP in the EU?

They've been punting this thing around the shows in the US for a couple
of years now but none of the carriers support it. With GSM having such
blanket coverage I don't see many carriers going this way. I can
understand this working in Asia where coverage is only in the majorly
populated areas and even then only outside.




On Mon, 2006-05-22 at 16:53 +0800, Sam Tam wrote:
  
 
 Well I think we all need to look at something like this first.
 We will be one of the first people in Europe who will be selling this.
 If anyone is interested do drop me an email.
 
  
 
 Picture of the phone can be found here.
 
 http://cyber-telecom.net/wifi-gsm.jpg
 
  
 
 GSM / VoIP Over WiFi Dual-Mode Phone
 
 CYBER-TELECOM released the world first commercial GSM/VoIP Over WiFi
 dual-mode smart phone, in March 2006. With a tri-band GSM/GPRS (Class
 10) radio and an IEEE 802.11b WLAN chipset, It enables end customers
 to enjoy broadband multimedia services at WLAN covered homes, offices,
 hot spots/zones as well as reliable GSM/GPRS service anytime anywhere.
 It shows an outstanding performance in power management, mobility
 management, security, mobile VoIP, and voice quality, no matter what
 kind of access points it connects, as the result of CYBER-TELECOM
 Wireless's advanced technologies solving the critical problems of VoIP
 Over WiFi. the phone has passed most of regulation certification
 programs and has done interoperability testing with over 40 VoIP
 service providers, system integrators, and infrastructure equipment
 vendors worldwide. the phone is an ideal device for fixed mobile
 convergence. 
 
 Hardware Specification
 
 Intel PXA271 processor with embedded Linux
 2.4 inch TFT touch screen, QVGA, 260k Colors
 Built-in speaker/microphone, 2.4mm stereo and headset
 1.3M pixel CMOS camera
 USB slave 
 Mini SD
 1100 mAh Li-ion battery 
 
 GSM Specification
 
 Frequency bands: 900/1800/1900 MHz
 GPRS Class 10
 SMS, MMS, WAP applications
 FTA/CTA certification
 FCC/CE certification
 
 WLAN Specification
 
 IEEE 802.11b 
 RF channels: US: 11, ETSI: 13, Japan: 14
 High-gain internal antenna
 WEP 64/128 bits, WPA, 802.1x
 EAP PSK/LEAP/PEAP/TTLS/SIM
 Power saving modes
 Fast roaming between access points
 
 VoIP Specification
 
 SIP: IETF RFC 3261
 Codec: G.711, G.729a/b, G.723 
 Acoustic echo cancellation
 Dynamic jitter buffer
 Voice activity detection
 Stun-based NAT traversal
 
 Input Methods
 
 Handwriting Recognition
  English
  Chinese
  Numeric characters
 Soft Keypads
  Qwerty
  Standard phone dialpad
  Symbol
 
 Power Management Features
 
 Standby time
 100 Hours (GSM on, WLAN on)
  200 Hours (GSM on, WLAN off)
 Talk time
  VoIP Over WiFi: 3.3 Hours
  GSM: 7.8 Hours
 MP3 play time
  5.8 Hours (GSM on, WLAN on)
  6.2 Hours (GSM on, WLAN off)
 
 Fixed Mobile Convergence Features
 
 Simultaneously activated GSM and WLAN air interfaces
 Handling simultaneously GSM and VoIP Over WiFi incoming calls
 SIP-based seamless handover between GSM/VoIP Over WiFi
 Automatic/manual switch for out-going calls between GSM and VoIP Over
 WiFi
 Automatic/manual switch for data applications using GPRS or WLAN
 Unified phone book for both GSM and VoIP Over WiFi.
 Unified GUI for applications (phone, E-mail, browser, QQ)
 
 Call Features
 
 Call hold
 Call waiting
 Call mute
 Call forward
 Call transfer
 3-way conference
 Voice mail
 SMS over SIP
 Phone book - (1000 entries with photos)
 Incoming call prompt with picture
 View phonebook during call
 Enter sketch pad during call
 Adjust volume during call
 Auto-answer/flip answer
 Quick silence
 Turbo dial
 Manual/Auto/Earphone redial
 Call history (20 entries)
 
 Data Application Features
 
 POP3 E-mail client (SSL support)
  100 full E-mails with attachments up to 200KB
  Document viewer for MS-Office and PDF files
 Web browser: HTML4.01, JAVAScript1.5, SSL3.0, HTTP1.1, CSS1.0
 Instant messaging: QQ
 
 Multimedia Features
 
 Video format: MP4, 3GPP
 Audio format: MP3, WAV, MIDI, AMR
 Picture format: WBMP, BMP, JPEG, GIF
 Camcorder: QVGA, QCIF
 Media Player
  Audio: MP3 player
  Video: up to 30 frames/second QVGA MP4/3GPP
 
 PIM Features
 
 Calendar
 Schedule management
 Alarm clock
 Voice recorder
 World time
 Currency converter
 Anniversary
 
 Other Features
 
 English - Chinese dictionary
 Calculator
 World time
 Notepad
 Sketch pad
 File transfer
 Counter
 Timer
 
  
 
 
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Re: [Asterisk-Users] how to customize voicemail

2006-05-22 Thread Avi Miller



On 22/05/2006, at 9:13 PM, [EMAIL PROTECTED] wrote:


Is it a way to record a welcome message and use it ?


Dial into VoiceMailMain() and hit 0 for Mailbox options. You can  
record both an Unavailable and a Busy message. :)


cYa,
Avi

--
National Manager - Special Projects

 Sydney / Melbourne / Canberra / Hobart / London /
  2/340 Gore StreetT: +61 (0) 3 9235 5400
  Fitzroy, VIC F: +61 (0) 3 9235 5444
  3065 W: http://www.squiz.net

.   Open Source - Own It - Squiz.net .. /




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Re: [Asterisk-Users] SIP to IAX - forcing codec pass thru

2006-05-22 Thread Peter Gradwell


Mark Phillips wrote:

Hi Peter,

I don't see any codec allow=blah statements. If your end user has
something like

[gradwell]
disallow=all
allow=gsm

Then you'll be forced to send them a GSM coded call. 


Why not force the codec at your end by only supporting one? If the
customer then transcodes the call when it gets forwarded to his handset
there's not much you can do about that but at least you'll have handed
the call off in the best format you can source.


mmm, but as you've seen, some customers like using multiple codecs. The 
cisco kit is able to support a raft of options - and it does transcoding 
very nicely - so the optimum solution is to have the cisco + customer's 
asterisk agree on the same codec, and then have our asterisk server (in 
the middle) do as little as possible.


cheers
peter

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Re: [Asterisk-Users] WiFi VoIP Handsets..

2006-05-22 Thread Steve Underwood


Mark Phillips wrote:


Don't ya just love living in this technological backwater they call the
USA? DECT technology was released almost 20 years ago. In most of the
world it's been and gone.

Anyone in the UK or Hong Kong remember Rabit and having to find a Green
Dot Hot Spot by the train station or post office? When you got home the
thing would mysteriously become part of your home phone system. 
 

Yeah. I remember that. Apparently you don't, or you might recall it 
didn't use DECT.




On Sun, 2006-05-21 at 18:42 -0500, Eric ManxPower Wieling wrote:
 

Are any of these FCC licensed for use in the USA.  DECT in the USA is 
VERY new.


   



I believe that DECT is approved for use here. Either that or Staples et
al are selling loads of illegal multi handset DECT phones. Some with
VoIP some without.
 


Steve

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Re: [Asterisk-Users] how to customize voicemail

Thank you very much.
I tried it and it works fine.
IIt i exactly what I needed.

Andrea



   
 Avi Miller
 [EMAIL PROTECTED] 
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 Sent by:  Asterisk Users Mailing List -   
 asterisk-users-bo Non-Commercial Discussion   
 [EMAIL PROTECTED] asterisk-users@lists.digium.com   
 m.com  cc 
   
   Subject 
 22/05/2006 13.25  Re: [Asterisk-Users] how to 
   customize voicemail 
   
 Please respond to 
  Asterisk Users   
  Mailing List -   
  Non-Commercial   
Discussion 
 [EMAIL PROTECTED] 
 ists.digium.com  
   
   





On 22/05/2006, at 9:13 PM, [EMAIL PROTECTED] wrote:

 Is it a way to record a welcome message and use it ?

Dial into VoiceMailMain() and hit 0 for Mailbox options. You can
record both an Unavailable and a Busy message. :)

cYa,
Avi

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[Asterisk-Users] I get MOH when the caller hangs up

2006-05-22 Thread Michael Knill


I get MOH when the caller hangs up.
Is there any way I can just get Busy tone.

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Re: [Asterisk-Users] Snom firmwares suck

2006-05-22 Thread Remco Barende


On Fri, 19 May 2006, Steve Davies wrote:


On 5/19/06, Remco Barende [EMAIL PROTECTED] wrote:

Most people seem quite positive about Snom phones, I cannot share this
opinion.

The displays are dying quite often, and firmware is buggy. I have tried
every firmware from 4.5 up to 5.x and 6.04 but keep having problems with
phones locking up or rebooting during an ongoing conversation.

REALLY annoying for a phone that is advertised / targeted as a business
class phone


Hmmm... A random statement out of the blue... I assume that you meant
to add Does any kind soul have a suggestion to help out? :-)

I find that the snom phones can be over-sensetive to network glitches,
which with the default configuration can cause a reboot (usually
caused by cheap switches). Try changing the reboot on ethernet unplug
setting to ignore.


Tried setting the phone to ignore, it didn't help. Still the phone 
reboots occasionally during a conversation.


But thanks for the suggestion!
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RE: [Asterisk-Users] I get MOH when the caller hangs up

2006-05-22 Thread Steven Totaro









Exten = h,1,hangup ?























From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael Knill
Sent: Monday, May 22, 2006 8:36 AM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] I get
MOH when the caller hangs up






I get MOH when the caller hangs up. Is there any way I
can just get Busy tone. 

Regards

Michael
Knill








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Re: [Asterisk-Users] asterisk and ODBC

2006-05-22 Thread Dumpolid Exeplish

I used the following command to connect to the MS SQL (Server 2000)isql -v ODBC-DSN sa ***and it worked. But if u can recomend another tool that i can use toi test this, pls do.thanks
On 5/19/06, Sean Cook [EMAIL PROTECTED] wrote:
have you tested to make sure that you can connect to the odbc resourceoutside of asterisk via perl/php/(insert random language here)?Makesure odbc is setup correctly and working before proceeding with theasterisk part.
SeanDumpolid Exeplish wrote: Hi, I have duetifully followed the instructions in the cdr.txt but asterisk still cannot connect to the MS SQL server. I can connect using the ODBC native connector bu asterisk still cannot
 On 5/16/06, *Bruce Reeves* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
 wrote: Yes you can use MSSQL with the ODBC driver, I have it working for CDR logs, I had to install unixODBC and configure it then use the cdr_odbc.conf file to specify. Check in your source files or svn
 checkout for a docs folder and the cdt.txt file. On 5/16/06, *Dumpolid Exeplish*  [EMAIL PROTECTED] mailto:
[EMAIL PROTECTED] wrote: Hi All, How can i use Microsoft SQL server with asterisk, Can the unix ODBC diriver interface MSQL?? and what module would i be using to access ODBC from asterisk??
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Re: [Asterisk-Users] WiFi VoIP Handsets..


Mark Phillips wrote:

Don't ya just love living in this technological backwater they call the
USA? DECT technology was released almost 20 years ago. In most of the
world it's been and gone.



I believe that DECT is approved for use here. Either that or Staples et
al are selling loads of illegal multi handset DECT phones. Some with
VoIP some without.


Silly me, I was looking online for DECT AND USA  Are the multihandset 
phones at Staples, etc really DECT or are they vendor specific?   I want 
the DECT roaming features and the DECT won't screw up WiFi features I 
keep hearing about.


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Re: FW: [Asterisk-Users] WiFi / GSM VoIP Handsets..

If you want to roam between GSM and WiFi while on a call, the GSM 
carrier is going to have to support it.


Dean Collins wrote:

Mark, how ignorant are you?

Dean


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark
Phillips
Sent: Monday, 22 May 2006 6:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: FW: [Asterisk-Users] WiFi / GSM VoIP Handsets..

As I understand it, the device uses either GSM or VoIP to access the
carrier? Which cell phone carrier supports GSM and VoIP in the EU?

They've been punting this thing around the shows in the US for a couple
of years now but none of the carriers support it. With GSM having such
blanket coverage I don't see many carriers going this way. I can
understand this working in Asia where coverage is only in the majorly
populated areas and even then only outside.




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Re: [Asterisk-Users] I get MOH when the caller hangs up

You would normally get Congestion tone.  Show us the dialplan for 
outgoing calls.


Michael Knill wrote:
I get MOH when the caller hangs up. Is there any way I can just get Busy 
tone.




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Re: [Asterisk-Users] Snom firmwares suck

2006-05-22 Thread Steve Davies


On 5/22/06, Remco Barende [EMAIL PROTECTED] wrote:

On Fri, 19 May 2006, Steve Davies wrote:
 I find that the snom phones can be over-sensetive to network glitches,
 which with the default configuration can cause a reboot (usually
 caused by cheap switches). Try changing the reboot on ethernet unplug
 setting to ignore.

Tried setting the phone to ignore, it didn't help. Still the phone
reboots occasionally during a conversation.

But thanks for the suggestion!


Oh well.. Perhaps this is a genuinely faulty unit then?

Thanks for the datapoint.
Steve
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Re: [Asterisk-Users] Snom 320 Shared line + speed dial

2006-05-22 Thread Dr. Michael J. Chudobiak


Just after some info on the Snom 320 before I got out an buy some...
 
I'm looking to use the shared line feature and hints with * so that i 
can monitor the activity of other users, but I'm not sure If this also 
turns the programmable buttons into a speed dial for quick transfers etc 
(or if it can be done). Ideally, I just want the users to be able to see 
the state of other users and be able to transfer to that user by using 
the programmable buttons...
 
Is this possible with this phone ??


Set the buttons to Destination mode, and set up corresponding hints 
in extensions.conf. Then the LED shows the user's status and hitting 
transfer+button will transfer a call to that user. Just hitting the 
button will dial that user's extension.


See SNOM SUBSCRIBE/NOTIFY support for monitoring extension states at 
http://www.voip-info.org/wiki/view/Asterisk+phone+snom.


- Mike

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Re: FW: [Asterisk-Users] WiFi / GSM VoIP Handsets..

2006-05-22 Thread Steve Kennedy

On Mon, May 22, 2006 at 08:14:19AM -0500, Eric ManxPower Wieling wrote:

 If you want to roam between GSM and WiFi while on a call, the GSM 
 carrier is going to have to support it.

There is a protocol for this (UMA), however few operators as yet support
it.

T-Mobile offer a webnwalk tarrif (unlimited data access for a fixed
monthly fee), but they are are going to (if not already) block VoIP
calls - they've realised that users are using VoIP (probably Skype) and
not making GSM voice calls - and the voice revenue is declining.


Steve

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Re: [Asterisk-Users] Experience with IBM X346 machines and Sangoma

2006-05-22 Thread Rich Adamson


I wanted to see where those periodical spikes are coming from so I
started shutting things down. The first thing to go was
Asterisk.

[...]

Is there something funny happening with my zaptel?
Wolfgang Zweimueller, can you give this a try too? Does your
spiking stop when you stop zaptel?


The spikes go away after unloading wanrouter modules but *before*
removing the zaptel module. Seems I have to contact Sangoma.


My understanding is that sangoma's drivers hook into zaptel, and its 
likely zaptel (and/or associated card drivers) is at the bottom of the 
spikes. Doubtful its a sangoma issue.



It seems the majority of folks that were involved with this in the
past 'assumed' the results were what was impacting fax through the
TDM400. But, don't think anyone proved that.


Dont't know anything about TDM400 but we had some issues with Modems
which were using the Asterisk-path.


Since the issue is common with both the TDM400 and sangoma cards, I 
don't believe the spikes are associated with the sangoma drivers.


R.


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[Asterisk-Users] Re: I get MOH when the caller hangs up

2006-05-22 Thread Tony Mountifield

In article [EMAIL PROTECTED],
Steven Totaro [EMAIL PROTECTED] wrote:
 
 Exten = h,1,hangup ?

No, there's never any need to call Hangup in the h extension, because
by the time h is called, the call is already hung up, by definition.

Cheers
Tony
-- 
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Work: [EMAIL PROTECTED] - http://www.softins.co.uk
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[Asterisk-Users] exten = *0. not possible

2006-05-22 Thread Patrick

Hi all,

It seems that using exten = _*0. is not possible in extensions.conf. I
changed disconnect = *0 in features.conf to something else. From what I
can tell with the little C knowledge I have is that it's caused by a
hardcoded *0 value chan_zap.c.

Line 5730 of chan_zap.c (svn rev 1077) shows:

} else if (!strcmp(exten, *0)) {
struct ast_channel *nbridge = 
p-subs[SUB_THREEWAY].owner;
struct zt_pvt *pbridge = NULL;

Can I just change that value to something else like *9 or even
totally remove the code so I can use exten = _*0. in my dialplan?

I'd also appreciate some guidance how to go about removing all the
hardcoded US service codes from chan_zap.c so those values become
available for dialplan sourcery.

Thanks and regards,
Patrick
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[Asterisk-Users] Option to reach someone in voicemail?


Hi,
Is there anyway to add an option to dial someone from voicemail?

I know I can make 0 go to operator... however, I want to do something
like our Nortel did which was Press 7 to reach XYZ and 7 could be
programmed to point to a specific person/extension/number.

Can I do this?
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Re: [Asterisk-Users] Option to reach someone in voicemail?

2006-05-22 Thread Sean Cook

Why not use exten = a,... and do a for more options press *, then
have it drop into an IVR...

Sean


Matt wrote:
 Hi,
 Is there anyway to add an option to dial someone from voicemail?

 I know I can make 0 go to operator... however, I want to do something
 like our Nortel did which was Press 7 to reach XYZ and 7 could be
 programmed to point to a specific person/extension/number.

 Can I do this?
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Re: [Asterisk-Users] Option to reach someone in voicemail?


That would be fine... and I know you can do alot of stuff with
Asterisk, you just have to think outside the box(tm) sometimes.
but my question with doing that is, then how do I make it go to this
message only when the person is not available, and not everytime
someone gets transfered to this extension?

On 5/22/06, Sean Cook [EMAIL PROTECTED] wrote:

Why not use exten = a,... and do a for more options press *, then
have it drop into an IVR...

Sean


Matt wrote:
 Hi,
 Is there anyway to add an option to dial someone from voicemail?

 I know I can make 0 go to operator... however, I want to do something
 like our Nortel did which was Press 7 to reach XYZ and 7 could be
 programmed to point to a specific person/extension/number.

 Can I do this?
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[Asterisk-Users] Re: exten = *0. not possible

2006-05-22 Thread Tony Mountifield

In article [EMAIL PROTECTED],
Patrick [EMAIL PROTECTED] wrote:
 Hi all,
 
 It seems that using exten = _*0. is not possible in extensions.conf. I
 changed disconnect = *0 in features.conf to something else. From what I
 can tell with the little C knowledge I have is that it's caused by a
 hardcoded *0 value chan_zap.c.
 
 Line 5730 of chan_zap.c (svn rev 1077) shows:
 
 } else if (!strcmp(exten, *0)) {
   struct ast_channel *nbridge = 
   p-subs[SUB_THREEWAY].owner;
   struct zt_pvt *pbridge = NULL;
 
 Can I just change that value to something else like *9 or even
 totally remove the code so I can use exten = _*0. in my dialplan?
 
 I'd also appreciate some guidance how to go about removing all the
 hardcoded US service codes from chan_zap.c so those values become
 available for dialplan sourcery.

Hi Patrick,

Are you using analogue phones connected to your asterisk box (directly
or through a channel bank)? Because it looks from the code as if that
is the only situation where those codes are captured by the chan_zap
driver and processed specially.

If you want to disable them, just insert an 'x' into the string, or some
other non-DTMF character. Then the strcmp will never match:

} else if (!strcmp(exten, x*0)) {

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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Re: [Asterisk-Users] Option to reach someone in voicemail?

2006-05-22 Thread Sean Cook

Matt wrote:
 That would be fine... and I know you can do alot of stuff with
 Asterisk, you just have to think outside the box(tm) sometimes.
 but my question with doing that is, then how do I make it go to this
 message only when the person is not available, and not everytime
 someone gets transfered to this extension?

exten = _4XXX,1,Dial(SIP/${EXTEN},20,tT)
exten = _4XXX,2,Goto(s-${DIALSTATUS},1)

exten = s-UNAVAILABLE,1,Background(some-ivr-recorded-message)
...

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RE: [Asterisk-Users] Re: I get MOH when the caller hangs up

2006-05-22 Thread Douglas Garstang

Do you have a 'g' option in your dial command? That will cause the dial plan to 
continue executing after they hangup I think.

 -Original Message-
 From: Tony Mountifield [mailto:[EMAIL PROTECTED]
 Sent: Monday, May 22, 2006 8:15 AM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] Re: I get MOH when the caller hangs up
 
 
 In article 
 [EMAIL PROTECTED],
 Steven Totaro [EMAIL PROTECTED] wrote:
  
  Exten = h,1,hangup ?
 
 No, there's never any need to call Hangup in the h extension, because
 by the time h is called, the call is already hung up, by definition.
 
 Cheers
 Tony
 -- 
 Tony Mountifield
 Work: [EMAIL PROTECTED] - http://www.softins.co.uk
 Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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Re: [Asterisk-Users] Option to reach someone in voicemail?


Ok,
That works... just couldn't think how to do it.


exten = _4XXX,1,Dial(SIP/${EXTEN},20,tT)
exten = _4XXX,2,Goto(s-${DIALSTATUS},1)

exten = s-UNAVAILABLE,1,Background(some-ivr-recorded-message)
...

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Re: [Asterisk-Users] Re: exten = *0. not possible

2006-05-22 Thread Patrick

On Mon, 2006-05-22 at 14:39 +, Tony Mountifield wrote:
[snip]
 Are you using analogue phones connected to your asterisk box (directly
 or through a channel bank)? Because it looks from the code as if that
 is the only situation where those codes are captured by the chan_zap
 driver and processed specially.

Yup, analog phones hooked up through a TDM31B.

 If you want to disable them, just insert an 'x' into the string, or some
 other non-DTMF character. Then the strcmp will never match:
 
 } else if (!strcmp(exten, x*0)) {

Thanks for the tip. I will give it a try and report back.

Regards,
Patrick
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Re: FW: [Asterisk-Users] WiFi / GSM VoIP Handsets..

2006-05-22 Thread Peter Bowyer


On 22/05/06, Steve Kennedy [EMAIL PROTECTED] wrote:

On Mon, May 22, 2006 at 08:14:19AM -0500, Eric ManxPower Wieling wrote:

 If you want to roam between GSM and WiFi while on a call, the GSM
 carrier is going to have to support it.

There is a protocol for this (UMA), however few operators as yet support
it.

T-Mobile offer a webnwalk tarrif (unlimited data access for a fixed
monthly fee), but they are are going to (if not already) block VoIP
calls - they've realised that users are using VoIP (probably Skype) and
not making GSM voice calls - and the voice revenue is declining.


They block VoIP and IM, supposedly to protect their users from a poor
quality experience. Of course, it's really to protect their voice and
SMS revenues.

Peter

--
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Email: [EMAIL PROTECTED]
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Re: [Asterisk-Users] Asterisk settings Net2Phone

2006-05-22 Thread Guillermo Salas M.

On Tue, 2006-05-09 at 11:37 -0300, Vinícius Bossle Fagundes wrote:
 Hi,
  
 I´m looking for settings to configure net2phone carrier in my
 asterisk. I found this configurations, but it´s not work. I don´t
 known if this configuration is for voice line or voice access account.
 Anybody can help me, with other configuration? 
  

I've some net2phone accounts working with Asterisk.

 Thanks.
  
 
  
 sip.conf 
 [general] 
 useragent = X-Lite release 1103m 
 register = PHONENUMBER:[EMAIL PROTECTED] 
 

---
sip.conf
---
[general]
useragent = Cisco ATA 186  v3.1.0 atasip
register=NET2PHONEACCOUNT:[EMAIL PROTECTED]

[net2phone]
username=NET2PHONEACCOUNT
useragent=Cisco ATA 186  v3.1.0 atasip (040211A)
type=peer
secret=PINNUMBER
qualify=no
nat=yes
insecure=very
host=sip.net2phone.com
fromuser=NET2PHONEACCOUNT
fromdomain=net2phone.com
canreinvite=no
allow=g723



 [net2phone] 
 type = peer 
 host = sip.net2phone.com 
 username = PHONENUMBER 
 secret = PASSWORD 
 fromuser = PHONENUMBER 
 fromdomain = net2phone.com 
 context = incoming 
 insecure = very 
 canreinvite = no 
 
 extensions.conf 
 [outgoing] 
 exten = _9NXXNXX,1,Dial(SIP/net2phone/${EXTEN:1}) 
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Re: FW: [Asterisk-Users] WiFi / GSM VoIP Handsets..

2006-05-22 Thread Steve Underwood


Steve Kennedy wrote:


On Mon, May 22, 2006 at 08:14:19AM -0500, Eric ManxPower Wieling wrote:

 

If you want to roam between GSM and WiFi while on a call, the GSM 
carrier is going to have to support it.
   



There is a protocol for this (UMA), however few operators as yet support
it.
 

Probably few ever will support it. It isn't in their interests to 
support it. The people who defined UMA failed to work out a sane 
business model for it before starting.



T-Mobile offer a webnwalk tarrif (unlimited data access for a fixed
monthly fee), but they are are going to (if not already) block VoIP
calls - they've realised that users are using VoIP (probably Skype) and
not making GSM voice calls - and the voice revenue is declining.
 


Regards,
Steve

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Re: FW: [Asterisk-Users] WiFi / GSM VoIP Handsets..

2006-05-22 Thread Roshan Sembacuttiaratchy

Or you need a very sophisticated phone onto which the corresponding
software can be loaded, for example the Nokia N series with the software
from Avaya. This combination allows transparent handover of a call from
WiFi to GSM, and back again to WiFi. It's not a cheap solution, though,
and only an option really for a company who are really serious about
using this technology. I don't think Asterisk comes into play at all,
as they have their own propriotory server for this. Here's a link which
kind of describes what I've mentioned here:

http://www.pctoday.com/Editorial/article.asp?article=articles/2006/t0407/13t07/13t07.aspguid=

And something partly related from the Avaya site:

http://www.avaya.com/gcm/master-usa/en-us/products/offers/one-x_mobile_edition.htm

HTH,

Roshan

On Mon, May 22, 2006 at 08:14:19AM -0500, Eric ManxPower Wieling scribbled:
If you want to roam between GSM and WiFi while on a call, the GSM
carrier is going to have to support it.

Dean Collins wrote:
Mark, how ignorant are you?

Dean

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark
Phillips
Sent: Monday, 22 May 2006 6:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: FW: [Asterisk-Users] WiFi / GSM VoIP Handsets..

As I understand it, the device uses either GSM or VoIP to access the
carrier? Which cell phone carrier supports GSM and VoIP in the EU?

They've been punting this thing around the shows in the US for a couple
of years now but none of the carriers support it. With GSM having such
blanket coverage I don't see many carriers going this way. I can
understand this working in Asia where coverage is only in the majorly
populated areas and even then only outside.

--
Now accepting new clients in Birmingham, Atlanta, Huntsville,
Chattanooga, and Montgomery.
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[Asterisk-Users] Office to Office via IAX2 problems

I'm going to try and lay out all the relevant information I have here 
in this one post.  I can provide more info if necessary.


ISSUE 1:
Office A routinely looses connection to Office B. When typing IAX2 
Show Peers, it will show as Unreachable. I issue IAX2 Reload and it 
will work again for 1-3 days (haven't narrowed the time down yet). My 
theory is that the DSL at Office2 is changing IP addresses regularly 
and this is the cause of the problem??? This has been going on since 
I set up Office B (2-3 weeks). I never had to touch Office B box. 
Office B seemed to maintain connection, until now (see Issue 2).


ISSUE 2:
Office B will not connect to Office A via IAX2 any more. The command 
IAX2 Show Peers shows Office A as Unreachable. IAX2 Reload won't fix 
it. I even rebooted the box (MS tricks never die). Up until 
yesterday, Office B always remained connected to Office A (or at 
least since I set up Office B - 2-3 weeks ago). Each office has port 
4569 forwarded to its * box. I even moved Office A box into DMZ, no 
help.   Note, Office A extensions can call extensions at Office B.


Here is the log file after I issue IAX2 Reload from Office B box:
May 18 16:29:38 DEBUG[3168] manager.c: Manager received command 'Command'
May 18 16:29:38 DEBUG[3168] manager.c: Manager received command 'Command'
May 18 16:29:48 VERBOSE[3170] logger.c: == Parsing 
'/etc/asterisk/iax.conf': May 18 16:29:48 VERBOSE[3170] logger.c: == 
Parsing '/etc/asterisk/iax.conf': Found
May 18 16:29:48 VERBOSE[3170] logger.c: == Parsing 
'/etc/asterisk/iax_additional.conf': May 18 16:29:48 VERBOSE[3170] 
logger.c: == Parsing '/etc/asterisk/iax_additional.conf': Found
May 18 16:29:48 NOTICE[3170] chan_iax2.c: Ignoring bindport on 
reload   '--- DON'T KNOW WHAT THIS IS
May 18 16:29:48 NOTICE[3170] chan_iax2.c: Ignoring bindaddr on 
reload   '--- OR THIS
May 18 16:29:48 DEBUG[3170] db.c: Unable to find key '450' in family 
'IAX/Registry' ' 450 UNUSED EXTENSION
May 18 16:29:48 VERBOSE[3170] logger.c: -- doing lookup for 
'officea.kicks-ass.net'

May 18 16:29:49 NOTICE[3170] chan_iax2.c: Still have a callno...
May 18 16:29:49 VERBOSE[3170] logger.c: == Loaded firmware 'iaxy.bin'
May 18 16:29:49 VERBOSE[3170] logger.c: == Parsing 
'/etc/asterisk/iaxprov.conf': May 18 16:29:49 VERBOSE[3170] logger.c: 
== Parsing '/etc/asterisk/iaxprov.conf': Found
May 18 16:29:49 VERBOSE[3170] logger.c: -- Loaded provisioning 
template 'default'
May 18 16:29:49 VERBOSE[3170] logger.c: -- Loaded provisioning 
template 'default'


I'm an admitted newbie, but this seems (to me) that this may be 
networking/Linux issue and not Asterisk???



SETUP:

OFFICE A:
-Asterisk 1.2.5
-Linksys WRT54GS router w/ SVEASOFT Alchemy-pre7a v3.37.6.8sv
-TW Roadrunner cablemodem (business class)
- * box sits behind router with port 4569 forwarded to * box. As 
noted, I moved * box to DMZ at one time, no help. Internal static IP 
address of 192.168.1.24


iax.conf:
[general]
bindport = 4569 ; Port to bind to (IAX is 4569)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
disallow=all
allow=ulaw
allow=alaw
allow=gsm
mailboxdetail=yes

#include iax_additional.conf
#include iax_custom.conf

iax_additional.conf:
[lpeaus]
username=lpeaus-user
type=peer
secret=secret
qualify=yes
host=officeb.kicks-ass.net
context=from-internal

[lpenb-user]
type=user
secret=secret
host=officeb.kicks-ass.net
context=from-internal



OFFICE B:
-Asterisk 1.2.7.1
-Linksys WRT54GL router
-SBC, er ATT, DSL
-Set up same as Office A (network wise). Internal static network IP 
address of 192.168.1.20


iax.conf:
[general]
bindport = 4569 ; Port to bind to (IAX is 4569)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
disallow=all
allow=ulaw
allow=alaw
allow=gsm
mailboxdetail=yes

#include iax_custom.conf
#include iax_additional.conf

iax_additional.conf :
[lpeaus-user]
type=user
secret=secret
host=officea.kicks-ass.net
context=from-internal

[lpenb]
username=lpenb-user
type=peer
secret=secret
qualify=yes
host=officea.kicks-ass.net
context=from-internal


TIA,
Doug


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Re: [Asterisk-Users] Net2phone on asterisk

2006-05-22 Thread Guillermo Salas M.

On Sun, 2006-05-21 at 19:45 -0500, Daniel wrote:
 Has anyone setup a n2p account into asterisk?
 

Yes, check
http://lists.digium.com/pipermail/asterisk-users/2006-May/152317.html

Regards,


Guillermo.

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Re: FW: [Asterisk-Users] WiFi / GSM VoIP Handsets..

2006-05-22 Thread Julio Arruda


Peter Bowyer wrote:

On 22/05/06, Steve Kennedy [EMAIL PROTECTED] wrote:


On Mon, May 22, 2006 at 08:14:19AM -0500, Eric ManxPower Wieling wrote:

 If you want to roam between GSM and WiFi while on a call, the GSM
 carrier is going to have to support it.

There is a protocol for this (UMA), however few operators as yet support
it.

T-Mobile offer a webnwalk tarrif (unlimited data access for a fixed
monthly fee), but they are are going to (if not already) block VoIP
calls - they've realised that users are using VoIP (probably Skype) and
not making GSM voice calls - and the voice revenue is declining.



They block VoIP and IM, supposedly to protect their users from a poor
quality experience. Of course, it's really to protect their voice and
SMS revenues.


From what I understand, T-Mobile UK just announced they would block 
VOIP earlier this month, but that is quite recent, and I don't recall 
seeing this 'globally' announced.



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Re: [Asterisk-Users] behaviour depending on count of used lines

2006-05-22 Thread picciuX

switch from global variables to group funcions. It's much more functional.Basically, you can set a group for an incoming channel, and count group instances across all channels. No necessity to decrement a variable, because group setting is channel dependent, so disappear when given channel is hangup.
Search for GROUP() and GROUP_COUNT() functions on http://www.voip-info.orgIt's a thing like this:exten = 33006712,1,GotoIf($[${GROUP_COUNT(LINES)} = 10]?101:201)
exten = 33006712,101,Dial(mISDN/g:E1/34507725)exten = 33006712,201,Set(GROUP()=LINES)exten = 33006712,202,Dial(SIP/192.168.0.65:5080SIP/192.168.0.65:5090)
Basically, in priority 201 you set the group named LINES for that channel before dialing your sip phone.In priority 1, instead, you check your used lines counter getting the number of active asterisk channels having group set to LINES.
that's all2006/5/22, Christophorus Laube [EMAIL PROTECTED]:
Hi there,I want to set up an extension set that acts different depending on the countof used lines. I have a EuroISDN E1 board with mISDN and I only want to offer10 lines. Therefore I set up a global variables LINES in the general section
of extensions.conf and instantiate it with 0. I a call is incoming I checkthe LINES variable wether is 10 or more. If so I make a call transfer. If notI increment the variable and direct the call to an internal SIP address.
After finishing the call I want to decrement the variable again, of course.My extension set looks like this way:[general]static=yeswriteprotect=noLINES = 0[E1]exten = 33006712,1,GotoIf($[${LINES} = 10]?101:201)
exten = 33006712,101,Dial(mISDN/g:E1/34507725)exten = 33006712,201,SetGlobalVar(LINES=$[ ${LINES} +1 ])exten = 33006712,202,Dial(SIP/192.168.0.65:5080SIP/192.168.0.65:5090)exten = 33006712-ANSWER,203,Answer()
exten = 33006712-HANGUP,204,SetGlobalVar(LINES=$[ ${LINES} -1])exten = 33006712,205,SetGlobalVar(LINES=$[ ${LINES} -1 ])exten = 33006712,206,Hangup()My problem is that the increment works perfectly, but the decrement is not
working. I added the last two extensions only because the hangup-extensiondid not work.Can anyone of you help me, please.TIA, Christophorus Laube___--Bandwidth and Colocation provided by 
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RE: FW: [Asterisk-Users] WiFi / GSM VoIP Handsets..

2006-05-22 Thread Dean Collins

I don't know why everyone is getting hung up on call handover etc, just
being able to make cheap outgoing calls is all I'm looking for.

I think this is going to be a great thing.


Dean


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Roshan
Sembacuttiaratchy
Sent: Monday, 22 May 2006 11:14 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: FW: [Asterisk-Users] WiFi / GSM VoIP Handsets..

Or you need a very sophisticated phone onto which the corresponding 
software can be loaded, for example the Nokia N series with the software

from Avaya.  This combination allows transparent handover of a call from

WiFi to GSM, and back again to WiFi.  It's not a cheap solution, though,

and only an option really for a company who are really serious about 
using this technology.  I don't think Asterisk comes into play at all, 
as they have their own propriotory server for this.  Here's a link which

kind of describes what I've mentioned here:

http://www.pctoday.com/Editorial/article.asp?article=articles/2006/t0407
/13t07/13t07.aspguid=

And something partly related from the Avaya site:

http://www.avaya.com/gcm/master-usa/en-us/products/offers/one-x_mobile_e
dition.htm

HTH,

Roshan

On Mon, May 22, 2006 at 08:14:19AM -0500, Eric ManxPower Wieling
scribbled:
 If you want to roam between GSM and WiFi while on a call, the GSM 
 carrier is going to have to support it.
 
 Dean Collins wrote:
 Mark, how ignorant are you?
 
 Dean
 
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Mark
 Phillips
 Sent: Monday, 22 May 2006 6:54 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: FW: [Asterisk-Users] WiFi / GSM VoIP Handsets..
 
 As I understand it, the device uses either GSM or VoIP to access the
 carrier? Which cell phone carrier supports GSM and VoIP in the EU?
 
 They've been punting this thing around the shows in the US for a
couple
 of years now but none of the carriers support it. With GSM having
such
 blanket coverage I don't see many carriers going this way. I can
 understand this working in Asia where coverage is only in the majorly
 populated areas and even then only outside.
 
 
 
 -- 
 Now accepting new clients in Birmingham, Atlanta, Huntsville, 
 Chattanooga, and Montgomery.
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[Asterisk-Users] TLS from a Sponsored Google Summer of Coding?

2006-05-22 Thread Dave Wise

I happened on to a website from Google that says there was a 
Digium/Google sponsored project to add certificates and TLS and TCP (as 
opposed to just UDP) to Asterisk.  Does anyone know anything about this 
as it indicates that it works in the current asterisk (since like August 
of 2005).


The web site is: http://savannah.nongnu.org/projects/asterisk-tcp
Go to the News Page and it indicates TLS works out of the box and that 
TCP works and That they tried it with Asterisk 1.2 in Aug 2005



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[Asterisk-Users] Script AGI on C

2006-05-22 Thread cleviton.araujo

Hi Folks:

I used that one example for AGI script on C web, only to fill the working with 
the Asterisk. I compiled and it worked great. I executed accidentally the ls -l 
command in directory where was the source and executable, I noted and was 
surprised that because the executable size was to further 20 times more than 
source.

I executed the gcc -Os source.c -o executable.agi command several times, with 
otimization flags different. Maximum i can affort to reduce the executable size 
was 17 times.

The source size full comment is 448 Bytes;
The size executable was about 7615 Bytes. (the maximum i got to reduce)

I was hope the executable size was in the order of magnitude of the proper 
source size, since the comments are long.

Do one get to explain because of this?
Is this overhead consequence of linking with the operational system?
The script use only four functions of stdio.h library. It was seem that the 
compiler include all stdio.h functions and compile all them.

I would like somebody of list to clear my doubt.

Regards,
Cleviton.


Here below small script used I grasp on site: 
http://home.cogeco.ca/~camstuff/agi.html

/* C works just fine with Asterisk but you should use 'setlinebuf' on stdout 
and stderr. This causes buffering one line at a time 
(rather than using a larger buffer). If you *don't* do this on stdout then your 
script will hang up while Asterisk waits for a 
command but the (long) buffer isn't full yet. A minimal AGI script in C looks 
like this: */
//
   #include stdio.h
   main() {
   charline[80];
   /* use line buffering */
   setlinebuf(stdout);
   setlinebuf(stderr);
   /* read and ignore AGI environment */
   while (1) {
   fgets(line,80,stdin);
   if (strlen(line) = 1) break;
   }
   /* Send asterisk a command */
   printf(SAY NUMBER 123 \\\n);
   /* Read response from Asterisk and show on console */
   fgets(line,80,stdin);
   fputs(line,stderr);
   }

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Re: FW: [Asterisk-Users] WiFi / GSM VoIP Handsets..

I wonder what they think VoIP is? Are they just port blocking? Could
they be doing packet inspection? Do they think all UDP trafic is VoIP?

On Mon, 2006-05-22 at 11:14 -0400, Julio Arruda wrote:

 
  From what I understand, T-Mobile UK just announced they would block 
 VOIP earlier this month, but that is quite recent, and I don't recall 
 seeing this 'globally' announced.
 


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Re: [Asterisk-Users] Script AGI on C

2006-05-22 Thread Diego Aguirre


Oi...

eu respondi sua mensagem na lista asteriskbrasil, mas com a moderação 
dela, só deve chegar amanhã, hehehe


tenta um strip no arquivo.

# strip executable.agi

isso deve reduzir mais um pouco o tamanho do seu arquivo...

Diego Aguirre
Infodag - Informática
FWD#: 459696
Nikotel#: 99 21 8138-2710
EnumLookup#: +55 21 8138-2710
DUNDi-br#: 21 8138-2710


[EMAIL PROTECTED] escreveu:

Hi Folks:

I used that one example for AGI script on C web, only to fill the working with 
the Asterisk. I compiled and it worked great. I executed accidentally the ls -l 
command in directory where was the source and executable, I noted and was 
surprised that because the executable size was to further 20 times more than 
source.

I executed the gcc -Os source.c -o executable.agi command several times, with 
otimization flags different. Maximum i can affort to reduce the executable size 
was 17 times.

The source size full comment is 448 Bytes;
The size executable was about 7615 Bytes. (the maximum i got to reduce)

I was hope the executable size was in the order of magnitude of the proper 
source size, since the comments are long.

Do one get to explain because of this?
Is this overhead consequence of linking with the operational system?
The script use only four functions of stdio.h library. It was seem that the 
compiler include all stdio.h functions and compile all them.

I would like somebody of list to clear my doubt.

Regards,
Cleviton.


Here below small script used I grasp on site: 
http://home.cogeco.ca/~camstuff/agi.html

/* C works just fine with Asterisk but you should use 'setlinebuf' on stdout and stderr. This causes buffering one line at a time 
(rather than using a larger buffer). If you *don't* do this on stdout then your script will hang up while Asterisk waits for a 
command but the (long) buffer isn't full yet. A minimal AGI script in C looks like this: */

//
   #include stdio.h
   main() {
   charline[80];
   /* use line buffering */
   setlinebuf(stdout);
   setlinebuf(stderr);
   /* read and ignore AGI environment */
   while (1) {
   fgets(line,80,stdin);
   if (strlen(line) = 1) break;
   }
   /* Send asterisk a command */
   printf(SAY NUMBER 123 \\\n);
   /* Read response from Asterisk and show on console */
   fgets(line,80,stdin);
   fputs(line,stderr);
   }

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Re: [Asterisk-Users] TLS from a Sponsored Google Summer of Coding?

2006-05-22 Thread Kevin P. Fleming

Dave Wise wrote:
 I happened on to a website from Google that says there was a
 Digium/Google sponsored project to add certificates and TLS and TCP (as
 opposed to just UDP) to Asterisk.  Does anyone know anything about this
 as it indicates that it works in the current asterisk (since like August
 of 2005).

To 'Asterisk'? That is not a correct description.

The project was to add TCP and TLS support for SIP (to chan_sip). The
code was submitted and has been sitting in the bug tracker waiting for
someone to be able to thoroughly review it, but the general consensus is
that the code that was provided was a good proof of concept, but not a
good long-term implementation.
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[Asterisk-Users] Asterisk Nortel Legacy Integration

2006-05-22 Thread Tron




Hi 
Srs.

 we have to integrate a Nortel MATRA M6501-L with Asterisk with a TE410P. 
All call from outside get into asterisk and asterisk send to Nortel in a correct 
way. My problem is when a call is made from Nortel to Asterisk. If we digit a 
national Number in Spain([98]ZXXX or 6) all work find. But if we 
digit an international number call doesn't progress. I Have seen in asterisk 
console that Nortel send to asterisk '00' and after it send rest of the number 
for international number.

I have 
made

exten=00,1,Goto(international,s,1)

[international]
exten=s,1,WaitExten(3)

exten=_X.,1,Dial(Zap/g1/00${EXTEN})

But I think that the 
problem is in zapata.conf, which is the next:

[trunkgroups]

[channels]language=escontext=defaultswitchtype=euroisdnpridialplan=unknownprilocaldialplan=unknown;resetinterval 
= 3600;overlapdial=yes; priindication = outofband; pritimer = 
t200,1000; pritimer = 
t313,4000usecallerid=yeshidecallerid=nocallwaiting=yes;restrictcid=nousecallingpres=yescallwaitingcallerid=yesthreewaycalling=yestransfer=yescanpark=yescancallforward=yescallreturn=yesechocancel=yesechocancelwhenbridged=yes;echotraining=yes;relaxdtmf=yesrxgain=0.0txgain=0.0

group=1signalling=pri_cpecallgroup=1pickupgroup=1immediate=nochannel=1-15,17-31,32-46,48-62

group=2context=from-nortelsignalling=pri_netoverlapdial=yescallgroup=1pickupgroup=2immediate=nochannel=63-77,79-93,94-108,110-124

Anyone could help 
me?

Regards,
tron
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Re: [Asterisk-Users] Office to Office via IAX2 problems

Don't specify the remote side by name, specify it by IP address.  If 
asterisk experiences even 1 dns failure it will not try again until a 
reload/restart/whatever.


[EMAIL PROTECTED] wrote:
I'm going to try and lay out all the relevant information I have here in 
this one post.  I can provide more info if necessary.


ISSUE 1:
Office A routinely looses connection to Office B. When typing IAX2 Show 
Peers, it will show as Unreachable. I issue IAX2 Reload and it will work 
again for 1-3 days (haven't narrowed the time down yet). My theory is 
that the DSL at Office2 is changing IP addresses regularly and this is 
the cause of the problem??? This has been going on since I set up Office 
B (2-3 weeks). I never had to touch Office B box. Office B seemed to 
maintain connection, until now (see Issue 2).


ISSUE 2:
Office B will not connect to Office A via IAX2 any more. The command 
IAX2 Show Peers shows Office A as Unreachable. IAX2 Reload won't fix it. 
I even rebooted the box (MS tricks never die). Up until yesterday, 
Office B always remained connected to Office A (or at least since I set 
up Office B - 2-3 weeks ago). Each office has port 4569 forwarded to its 
* box. I even moved Office A box into DMZ, no help.   Note, Office A 
extensions can call extensions at Office B.


Here is the log file after I issue IAX2 Reload from Office B box:
May 18 16:29:38 DEBUG[3168] manager.c: Manager received command 'Command'
May 18 16:29:38 DEBUG[3168] manager.c: Manager received command 'Command'
May 18 16:29:48 VERBOSE[3170] logger.c: == Parsing 
'/etc/asterisk/iax.conf': May 18 16:29:48 VERBOSE[3170] logger.c: == 
Parsing '/etc/asterisk/iax.conf': Found
May 18 16:29:48 VERBOSE[3170] logger.c: == Parsing 
'/etc/asterisk/iax_additional.conf': May 18 16:29:48 VERBOSE[3170] 
logger.c: == Parsing '/etc/asterisk/iax_additional.conf': Found
May 18 16:29:48 NOTICE[3170] chan_iax2.c: Ignoring bindport on reload   
'--- DON'T KNOW WHAT THIS IS
May 18 16:29:48 NOTICE[3170] chan_iax2.c: Ignoring bindaddr on reload   
'--- OR THIS
May 18 16:29:48 DEBUG[3170] db.c: Unable to find key '450' in family 
'IAX/Registry' ' 450 UNUSED EXTENSION
May 18 16:29:48 VERBOSE[3170] logger.c: -- doing lookup for 
'officea.kicks-ass.net'

May 18 16:29:49 NOTICE[3170] chan_iax2.c: Still have a callno...
May 18 16:29:49 VERBOSE[3170] logger.c: == Loaded firmware 'iaxy.bin'
May 18 16:29:49 VERBOSE[3170] logger.c: == Parsing 
'/etc/asterisk/iaxprov.conf': May 18 16:29:49 VERBOSE[3170] logger.c: == 
Parsing '/etc/asterisk/iaxprov.conf': Found
May 18 16:29:49 VERBOSE[3170] logger.c: -- Loaded provisioning template 
'default'
May 18 16:29:49 VERBOSE[3170] logger.c: -- Loaded provisioning template 
'default'


I'm an admitted newbie, but this seems (to me) that this may be 
networking/Linux issue and not Asterisk???



SETUP:

OFFICE A:
-Asterisk 1.2.5
-Linksys WRT54GS router w/ SVEASOFT Alchemy-pre7a v3.37.6.8sv
-TW Roadrunner cablemodem (business class)
- * box sits behind router with port 4569 forwarded to * box. As noted, 
I moved * box to DMZ at one time, no help. Internal static IP address of 
192.168.1.24


iax.conf:
[general]
bindport = 4569 ; Port to bind to (IAX is 4569)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
disallow=all
allow=ulaw
allow=alaw
allow=gsm
mailboxdetail=yes

#include iax_additional.conf
#include iax_custom.conf

iax_additional.conf:
[lpeaus]
username=lpeaus-user
type=peer
secret=secret
qualify=yes
host=officeb.kicks-ass.net
context=from-internal

[lpenb-user]
type=user
secret=secret
host=officeb.kicks-ass.net
context=from-internal



OFFICE B:
-Asterisk 1.2.7.1
-Linksys WRT54GL router
-SBC, er ATT, DSL
-Set up same as Office A (network wise). Internal static network IP 
address of 192.168.1.20


iax.conf:
[general]
bindport = 4569 ; Port to bind to (IAX is 4569)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
disallow=all
allow=ulaw
allow=alaw
allow=gsm
mailboxdetail=yes

#include iax_custom.conf
#include iax_additional.conf

iax_additional.conf :
[lpeaus-user]
type=user
secret=secret
host=officea.kicks-ass.net
context=from-internal

[lpenb]
username=lpenb-user
type=peer
secret=secret
qualify=yes
host=officea.kicks-ass.net
context=from-internal


TIA,
Doug


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[Asterisk-Users] Re: Configuring a TDM400P with one FXS port

2006-05-22 Thread M.Hockings


Hi Mohamed,

I am beginning to wonder if either the card is not compatible with the 
box it is in.  That is even though the box is a PII400 which should meet 
the minimum requirements (300mhz) but I am finding that it won't always 
boot or hangs after boot with the card installed.  So I'm looking around 
for another unused box to try it in.


Running the command string :

 modprobe zaptel;modprobe wctdm

Just gives the now-familiar:

ZT_CHANCONFIG failed on  channel 1: No such device or address (6)
FATAL: Error running install command for wctdm

It seems that maybe my best plan right now is to just put this on hold 
until our new IBM/Lenovo server arrives, hopefully this week. It should 
be able to handle this new hardware with no problems.


Many thanks for your assistance Mohammed.

Mike

mohamed kerbachi wrote:

Hi,

Strange, i get the correct output from the ddmesg:
...
...
Freshmaker passed register test
Module 0: Installed -- AUTO FXS/DPO
Module 1: Installed -- AUTO FXS/DPO
Module 2: Installed -- AUTO FXS/DPO
Module 3: Installed -- AUTO FXO (FCC mode)
...
...
..

As you see i have TDM400P with 3xFXS + 1xFXO

Try to issue:
modprobe zaptel;modprobe wctdm

And give me the output of the command :lspci



--- M.Hockings [EMAIL PROTECTED] a écrit :


mohamed kerbachi wrote:

Hi,
I have a TDM400P and it works,
Send us the output of your dmesg command.

Regards.
--- M.Hockings [EMAIL PROTECTED] a écrit

:

In my attempt to setup a single FXS line I have

been
following the 
instructions for Telephony Card Drivers on the

asteriskdocs.org site.
I have managed to checkout, make and install the
zaptel code and can 
load the zaptel module but when I attempt to load
the wcfxs module it 
tells me:


ZT_CHANCONFIG failed on  channel 1: No such

device

or address (6)
FATAL: Error running install command for wctdm

On the card the single FXS module is in the

position
at the back of the 
TDM400P (i.e., closest to the connectors)


In the /etc/zaptel.conf file I have put the
following at the bottom of 
the file:


fxoks=1
loadzone=us
defaultzone=us

I have also tried fxoks=4 with the same results
other than that the 
channel number changes in the message.


Any idea what I am configuring incorrectly ?

Mike

Hi Mohamed, if you can help guide me in the right
direction I would be 
most appreciative.


The output of dmesg is below.

In the machine is a TDM400P with one FXS, a X101P
FXO, an ethernet card 
and an old ISA modem.


Mike


Linux version 2.6.9-34.EL ([EMAIL PROTECTED])
(gcc version 3.4.5 
20051201 (Red Hat 3.4.5-2)) #1 Wed Mar 8 00:07:35

CST 2006
BIOS-provided physical RAM map:
  BIOS-e820:  - 0009fc00
(usable)
  BIOS-e820: 0009fc00 - 000a
(reserved)
  BIOS-e820: 000f - 0010
(reserved)
  BIOS-e820: 0010 - 17ffa000
(usable)
  BIOS-e820: 17ffa000 - 17ffe000
(ACPI data)
  BIOS-e820: 17ffe000 - 1800
(ACPI NVS)
0MB HIGHMEM available.
383MB LOWMEM available.
Using x86 segment limits to approximate NX
protection
zapping low mappings.
On node 0 totalpages: 98298
   DMA zone: 4096 pages, LIFO batch:1
   Normal zone: 94202 pages, LIFO batch:16
   HighMem zone: 0 pages, LIFO batch:1
DMI 2.0 present.
ACPI: RSDP (v000 IBM
  ) @ 0x000fe030

ACPI: RSDT (v001 IBMV66XA0x0001 Acer
0x) @ 0x17ffa000
ACPI: FADT (v001 IBMV66XA0x0001 Acer
0x) @ 0x17ffa028
ACPI: DSDT (v001   IBMV66XA  0x1000 MSFT
0x010a) @ 0x
ACPI: BIOS age (1998) fails cutoff (2001),
acpi=force is required to 
enable ACPI

ACPI: Disabling ACPI support
Built 1 zonelists
Kernel command line: ro
root=/dev/VolGroup00/LogVol00 quiet
Initializing CPU#0
CPU 0 irqstacks, hard=c03e7000 soft=c03e6000
PID hash table entries: 2048 (order: 11, 32768
bytes)
Detected 400.948 MHz processor.
Using tsc for high-res timesource
Console: colour VGA+ 80x25
Dentry cache hash table entries: 65536 (order: 6,
262144 bytes)
Inode-cache hash table entries: 32768 (order: 5,
131072 bytes)
Memory: 384696k/393192k available (2117k kernel
code, 7884k reserved, 
669k data, 144k init, 0k highmem)

Calibrating delay using timer specific routine..
803.04 BogoMIPS 
(lpj=401522)

Security Scaffold v1.0.0 initialized
SELinux:  Initializing.
SELinux:  Starting in permissive mode
There is already a security framework initialized,
register_security failed.
selinux_register_security:  Registering secondary
module capability
Capability LSM initialized as secondary
Mount-cache hash table entries: 512 (order: 0, 4096
bytes)
CPU: After generic identify, caps: 0183f9ff 
 
CPU: After vendor identify, caps:  0183f9ff 
 
CPU: L1 I cache: 16K, L1 D cache: 16K
CPU: L2 cache: 512K
CPU: After all inits, caps:0183f1ff 
 0040
Intel machine check architecture supported.
Intel machine check reporting enabled on CPU#0.
CPU: Intel

[Asterisk-Users] Re: Recommended SIP phones?

2006-05-22 Thread M.Hockings


Pieter Claassen wrote:
I am dying here with linphone (not sure if it is crap software or just me 
being an idiot) but out of the box debian installations of two linphones fail 
with a Got SIP response 415 Unsupported Media Type back from 192.168.1.3


Can anybody recommend a particular SIP soft phone that broadly satisfies the 
following criteria?

1. Run on linux.
2. Simple to use and setup.
3. Is preferably packaged in APT.
4. OSS compliant license.

Any recommendations (or let me know if I am doing something wrong with 
linphone) are welcome.


Thanks
Pieter


I've been using both the windows and linux versions of xten lite and 
they work fine when configured correctly...


Mike

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Re: [Asterisk-Users] Office to Office via IAX2 problems

Thanks.  I had already tried going into Office B box and change host= 
(office A's IP address), performed iax2 reload, and this did not work either.


At 10:51 AM 5/22/2006, you wrote:
Don't specify the remote side by name, specify it by IP address.  If 
asterisk experiences even 1 dns failure it will not try again until 
a reload/restart/whatever.


[EMAIL PROTECTED] wrote:


SETUP:
OFFICE A:
iax_additional.conf:
[lpeaus]
username=lpeaus-user
type=peer
secret=secret
qualify=yes
host=officeb.kicks-ass.net
context=from-internal
[lpenb-user]
type=user
secret=secret
host=officeb.kicks-ass.net
context=from-internal

OFFICE B:
iax_additional.conf :
[lpeaus-user]
type=user
secret=secret
host=officea.kicks-ass.net
context=from-internal
[lpenb]
username=lpenb-user
type=peer
secret=secret
qualify=yes
host=officea.kicks-ass.net
context=from-internal



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[Asterisk-Users] Centos 4.3 Issues

2006-05-22 Thread Greg Boehnlein

Hello,
I was wondering if anyone out there is successfully running 
Asterisk 1.2 svn w/ Centos 4.3. I had an experience over the last two 
weeks that has me scratching my head and muttering strange things in the 
wee hours of the morning. I am going to try and be as descriptive as my 
brain will allow right now, but if there is something that I do not cover, 
please do not hesitate to ask and I'll be happy to answer.

For the last 2 years, I have been running a mixture of Tao Linux 
and Centos (both RHEL derivatives) on our production boxes. Asterisk has 
run flawlessly on all installations. Last week, I updated one of our 
gateway boxes from Centos 4.2 (under which it ran for 6 months without 
issue) to the new 4.3 code. Almost immediately, we began to experience 
problems. Asterisk would core w/ the following:

#0  0x004878ab in test_err () from 
/usr/lib/asterisk/modules/codec_g729a.so

The segfaults would happen under very light loads, in some cases 
with just a single call. Kevin was able to log in to the box, and put a 
debugging version of codec_g729 on the box. He determined that the problem 
was that the values that were being returned in that routine were 
incorrect. I.E. something in the system was returning a non-zero value 
when multiplying a number by 0. Barring any other explanations, we 
assumed that there was a hardware issue somewhere, either in the memory, 
or the FPU on the CPU.
So, we replaced the box w/ a brand new Dual-Core system running a 
Dual-Core Pentium D 920. We loaded the 32 bit version of Centos 4.3 onto 
the box and proceeded to start testing. BAM.. same problem.. the backtrace 
showed the failure in the same routine.
We scratched our heads, and after many hours of trying various 
things (backing off the kernel to 2.6.9-22) and even moving to the new 
development kernel 2.6.9-34.19 (from the testing tree) we could do nothing 
to solve the issue.
Mind you, this is the exact same behavior on two different 
hardware platforms running the exact same distribution. We even loaded up 
a third box and could reproduce the behavior on it as well. Three 
different boxes, one common distribution.

As a test, we installed Fedora Core 5 x86_64 on the new Dual Core 
box and ran extensive tests overnight, simulating 96 channels doing G729 
to Ulaw transcoding. The box ran completely stable. No hiccups.

So, this morning, we put it back into the cluster, and it's now 
taking about 200 concurrent calls, doing an insane amount of transcoding 
and it is working just fine. Before, it would have cored in the first 
couple of minutes.

I'm scratching my head here, because I generally have had excellent 
experiences with Centos. However, I have NO idea what might be the issue 
here. Could it be the kernel? (We tried three different ones!). Could it 
be the libc? Maybe it is the compiler?

In any case, if anyone is having success with Centos 4.3 (32 bit), please 
speak up. I'd like to get to the bottom of it. I generally do not like to 
run Fedora on production equipment as it is generally bleeding edge. In 
this case, FC5 is running 2.6.16 something..

-- 
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] [EMAIL PROTECTED] doing SIP URI calls

2006-05-22 Thread Joao Pereira


Hello to all
Im trying to make SIP URI calls with my [EMAIL PROTECTED], and I followed this:
http://slacker.com/~nugget/projects/asterisk/page7

So I putted in extensions.conf:

MYDOMAIN = xxx.xxx.xxx.xxx
MYFQDN = xxx.xxx.xxx.xxx

[macro-uridial]
exten = s,1,NoOp(Outbound SIP URI call ${ARG1})
exten = s,2,SetCIDNum(5125380508)
exten = s,3,Dial(SIP/${ARG1})
exten = s,4,Congestion()


and in extensions_custom.conf :

[from-internal-custom]
exten = _.,1,NoOp(Incoming Call from house extension ${CALLERID} for 
[EMAIL PROTECTED])

exten = _.,2,GotoIf($[${LEN(${SIPDOMAIN})} = 0]?10)
exten = _.,3,GotoIf($[${SIPDOMAIN} = ${MYDOMAIN}]?10)
exten = _.,4,GotoIf($[${SIPDOMAIN} = ${MYFQDN}]?10)
exten = _.,5,GotoIf($[${SIPDOMAIN} = ${MYFQDN}:5060]?10)
exten = _.,6,NoOp(@${SIPDOMAIN} is remote, forwarding...)
exten = _.,7,Macro(uridial,[EMAIL PROTECTED])
exten = _.,8,HangUp()
exten = _.,10,Goto(noturi,${EXTEN},1)
exten = h,1,HangUp()

[noturi]
include = local
include = trunkld
include = trunkint
include = emergency



Then, I try to call [EMAIL PROTECTED] and the call fails:

asterisk debug:
Looking for 613 in from-internal (domain fwd.pulver.com)
Reliably Transmitting (NAT) to xxx.xxx.xxx.xxx:5060:
SIP/2.0 404 Not Found



If I have _. in [from-internal-custom] why do the call fails?
Thanks
Joao Pereira


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[Asterisk-Users] Polycom Echo

2006-05-22 Thread Kevin Ragsdale

Hello,

We just experienced a problem that we though might be useful to anyone
using Polycom phones.  We are installing a new system at one of our
remote offices and were experienced a ton of echo on our side while the
remote side was on speakerphone.  It turns out that the desk surface was
causing the echo - when the phone was lifted off the desk, the echo
disappeared.  We also were able to put a mousepad under the phone to
eliminate the echo as well.  Since the microphone is on the bottom of
the phone, there must be some kind of sound reflection going on.

Just a heads up - this is the first time we've seen this happen.

Kevin
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[Asterisk-Users] Persistennt Data of Queue with Dynamic Agents

2006-05-22 Thread Pimjai Wesnarat


Hi all,

I would like to ask for some help about the queue here. I want to 
implement a call Queue that when there's no agent logged in, they should 
execute the next extension. eg. if I do it like this


exten = 700,1,Answer
exten = 700,2,Queue(TestQueue)
exten = 700,3,Playback(noagent)
exten = 700,4,Hangup

When there's no agent present in TestQueue, it should tell the user that 
there's no agent available now and hang up the call. We are able to do 
this thing successfully only by using the dynamic agent -- 
AddQueueMember().

However, using this approach gave us a few more problems that are:

1. The realtime data for dynamic agents is not really persistent. The 
setting persistentmembers=yes in queues.conf is only saving the queue 
information, if you login and logout an agent all queue member related 
data is lost. Is there any way to make these data persistent?


2. If Asterisk is rebooted, all the information is reset - the queue and 
agents information are all reset. Is there any way to avoid this?


3.  It seems that it is possible to login one agent into the same 
extension. Is there anyway to avoid this??



Any hint would be appreciated.

Pim



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Re: [Asterisk-Users] Centos 4.3 Issues

2006-05-22 Thread Greg Oliver

On Mon, 2006-05-22 at 12:16 -0400, Greg Boehnlein wrote:
 Hello,
   I was wondering if anyone out there is successfully running 
 Asterisk 1.2 svn w/ Centos 4.3. I had an experience over the last two 
 weeks that has me scratching my head and muttering strange things in the 
 wee hours of the morning. I am going to try and be as descriptive as my 
 brain will allow right now, but if there is something that I do not cover, 
 please do not hesitate to ask and I'll be happy to answer.
 
   For the last 2 years, I have been running a mixture of Tao Linux 
 and Centos (both RHEL derivatives) on our production boxes. Asterisk has 
 run flawlessly on all installations. Last week, I updated one of our 
 gateway boxes from Centos 4.2 (under which it ran for 6 months without 
 issue) to the new 4.3 code. Almost immediately, we began to experience 
 problems. Asterisk would core w/ the following:
 
 #0  0x004878ab in test_err () from 
 /usr/lib/asterisk/modules/codec_g729a.so
 
   The segfaults would happen under very light loads, in some cases 
 with just a single call. Kevin was able to log in to the box, and put a 
 debugging version of codec_g729 on the box. He determined that the problem 
 was that the values that were being returned in that routine were 
 incorrect. I.E. something in the system was returning a non-zero value 
 when multiplying a number by 0. Barring any other explanations, we 
 assumed that there was a hardware issue somewhere, either in the memory, 
 or the FPU on the CPU.
   So, we replaced the box w/ a brand new Dual-Core system running a 
 Dual-Core Pentium D 920. We loaded the 32 bit version of Centos 4.3 onto 
 the box and proceeded to start testing. BAM.. same problem.. the backtrace 
 showed the failure in the same routine.
   We scratched our heads, and after many hours of trying various 
 things (backing off the kernel to 2.6.9-22) and even moving to the new 
 development kernel 2.6.9-34.19 (from the testing tree) we could do nothing 
 to solve the issue.
   Mind you, this is the exact same behavior on two different 
 hardware platforms running the exact same distribution. We even loaded up 
 a third box and could reproduce the behavior on it as well. Three 
 different boxes, one common distribution.
 
   As a test, we installed Fedora Core 5 x86_64 on the new Dual Core 
 box and ran extensive tests overnight, simulating 96 channels doing G729 
 to Ulaw transcoding. The box ran completely stable. No hiccups.
 
   So, this morning, we put it back into the cluster, and it's now 
 taking about 200 concurrent calls, doing an insane amount of transcoding 
 and it is working just fine. Before, it would have cored in the first 
 couple of minutes.
 
 I'm scratching my head here, because I generally have had excellent 
 experiences with Centos. However, I have NO idea what might be the issue 
 here. Could it be the kernel? (We tried three different ones!). Could it 
 be the libc? Maybe it is the compiler?
 
 In any case, if anyone is having success with Centos 4.3 (32 bit), please 
 speak up. I'd like to get to the bottom of it. I generally do not like to 
 run Fedora on production equipment as it is generally bleeding edge. In 
 this case, FC5 is running 2.6.16 something..
 

Have you tried compiling statically on CentOS 4.2 and running on 4.3?

I am assuming you have made sure the dist is up to date with patches.
We do not use 729, so I cannot try it out for you, but we do use CentOS.
Is it only w/ SVN, or all releases of *?

-Greg

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[Asterisk-Users] Problems with Park and MOH

2006-05-22 Thread Michael Knill


I know I have read something about this
but when I put a call on Park, I get loud and distorted MOH. I think it
had something to do with RTP in one direction only and there was some sync
problem.

Can someone tell me how to fix this.

I am using Asterisk version 1.2.7.1
with AstLinux.

Regards
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[Asterisk-Users] playback windows recorded sound

2006-05-22 Thread Akpome Akpoguma


Hi guys,

I recorded a wav file on my windows xp laptop and tried to playback on 
asterisk but got the following error..unexpected header error 
18.when  I recorded sound using a sip phone on asterisk and compared 
with what I recorded on windows the sound property looked the same.does 
anyone have an idea how I can resolve this?


response would be highly appreciated.

rgds,

_
Don't just search. Find. Check out the new MSN Search! 
http://search.msn.com/


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Re: [Asterisk-Users] Centos 4.3 Issues

2006-05-22 Thread Kevin P. Fleming

Greg Oliver wrote:

 I am assuming you have made sure the dist is up to date with patches.
 We do not use 729, so I cannot try it out for you, but we do use CentOS.
 Is it only w/ SVN, or all releases of *?

The problem does not appear to be happening in Asterisk itself, but in
the G.729 codec module. The symptom is a failure of a floating-point
expression to produce the proper result, leading to a segfault when the
code tries to access a non-existent array member.

We tried building the codec without any optimization at all, and still
experienced the same issue. This points to some sort of 'core' problem
on the system, but as Greg said, hardware has been ruled out. That
leaves basically the kernel and the C library's floating point stuff, I
believe... but switching kernels did not help (although they were all
very similar kernels).
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[Asterisk-Users] UUI field

2006-05-22 Thread Ray Iallip


How is the UserUserInfo field in the Q931 messagetranslated to SIP message.

Does Asterisk have RFC 3398 support?


Thanks 

--Ray
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[Asterisk-Users] FXS Caller ID revisted

2006-05-22 Thread Dan Elder

Hi All, posted last week but didn't get any responses. I'm trying to figure
out why some of our analog phones aren't showing CID when hooked up to
asterisk. To recap, I have an Aastra Powertouch 350, which shows caller ID
fine when connected to the PSTN, but when hooked up to asterisk, CID does
not show. I've hooked up another phone to the same * port that the Aastra
phone is on,  it DOES show CID, so I'm assuming my settings  such are at
least partially correct, can anyone point me to some options or areas I can
look to troubleshoot this issue? Been pulling my hair out on this for days 
just can't seem to get it sorted.

I'm using asterisk 1.2.0 with a Carrier Access ABII channel bank. When
another CID capable phone is hooke up to the same port, CID works fine, the
Aastra phone is however unable to read the incoming CID from * apparently.

Any pointers would be greatly appreciated, I've searched the Wiki  the CID
faq's to no avail.

Thanks in advance

Dan

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Re: [Asterisk-Users] Centos 4.3 Issues

2006-05-22 Thread alist


Greg Boehnlein wrote:


Hello,
	I was wondering if anyone out there is successfully running 
Asterisk 1.2 svn w/ Centos 4.3. I had an experience over the last two 
weeks that has me scratching my head and muttering strange things in the 
wee hours of the morning. I am going to try and be as descriptive as my 
brain will allow right now, but if there is something that I do not cover, 
please do not hesitate to ask and I'll be happy to answer.




Greg,

When I upgraded to 4.3 I experienced problems with some non-asterisk 
RPM's that were compiled on earlier versions of CentOS 4. Once they were 
recompiled on a fully updated 4.3 system they worked fine. Have you 
tried recompiling everything?


Andrew
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Re: [Asterisk-Users] playback windows recorded sound

2006-05-22 Thread Doug Lytle


Akpome Akpoguma wrote:

Hi guys,

I recorded a wav file on my windows xp laptop and tried to playback on 
asterisk but got the following error..unexpected header error 
18.when  I recorded sound using a sip phone on asterisk and 
compared with what I recorded on windows the sound property looked the 
same.does anyone have an idea how I can resolve this?



This should help:

http://www.voip-info.org/wiki/view/Asterisk+sound+files

Doug

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[Asterisk-Users] A few queue questions


#1 - Is there anyway to 'cancel a wrap'?  That is.. when an agent is
waiting their 45 seconds or so for the next call, is there anyway to
cancel that and continue taking calls?

#2 - I have agents who log into multiple queues.  I have them staticly
assigned to all queues, but they log in with their agent ids (defined
in agents.conf).  It seems that they will be wrapping on queue 1,
but a call from queue 2 will come in while they should still be
wrapping.  Is this normal?  How can I get around it?
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RE: [Asterisk-Users] Meetme and authentication

2006-05-22 Thread Josh McAllister

Title: Meetme and authentication








Perhaps youve already figured this
out, but I posted an example dialplan and small Perl AGI that would resolve
this for you. As it happens this was posted the Friday before you sent this.
Look for a posting from me on Friday, May 12th.



Josh McAllister 













From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Herchi Silviu
Sent: Tuesday, May 16, 2006 8:41
AM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: [Asterisk-Users] Meetme
and authentication





Hi all, 

I have thoroughly read the available documentation and I can't seem to
find a workaround for my setup 

I'm trying to create a phone conference line that users would call using
a unique phone number (no matter if they are moderators or just plain users). I
use Asterisk 1.2.6

The available conferences are defined as follows: 

conf = 1000,user pin1, moderator pin1 
conf = 1001,user pin2, moderator pin2 
conf = 1002,user pin3, moderator pin3 
 
conf = 1009, user pin9, moderator pin9 

The users are prompted whether they are a moderator or a user. When they
choose, they are redirected to the conference they request:

- using options aAPsX for moderators (moderator + marked + ask PIN +
allow menu using *) 
- using options Psw for users (ask PIN + allow menu + wait for a marked
user) 

My problem is that if a user chooses the moderator option,
he can authenticate using any of the two PINs, and he can become an moderator
for the conference by knowing only the user PIN

I think using two different phone numbers (one for users and one for
moderators) is neither practical nor safe. Is there a way to authenticate users
against only one of the password? For instance, math the password provided
against only the moderator PIN, or only the user PIN.

Thank you for your help, 

Silviu 

PS. Here is the dialplan : 

[ConfStart] 
exten = s,1,Answer 
exten = s,2,Set(TIMEOUT(response)=5) 
exten = s,3,Set(LANGUAGE()=conf) 
exten = s,4,Wait(1) 
exten =
s,5,Background(welcome) ; welcome,
press * if you are a user of hold the line if you are a moderator 

exten = *,1,MeetMe(|iMPsw|) ; for regular users

exten = t,1,MeetMe(|aAiMPsX|) ; for moderators 

exten = i,1,GoTo(ConfStart,s,1) 








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RE: FW: [Asterisk-Users] WiFi / GSM VoIP Handsets..

2006-05-22 Thread Sam Tam

Well it is incorrect to say that.
In places like USA or London, a lot of areas are covered by local wifi
providers, some are free, some aren't. 
You then can use them to drop some of your local or international calls
cheaply by using wifi.

Sam

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Phillips
Sent: Monday, May 22, 2006 6:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: FW: [Asterisk-Users] WiFi / GSM VoIP Handsets..

As I understand it, the device uses either GSM or VoIP to access the
carrier? Which cell phone carrier supports GSM and VoIP in the EU?

They've been punting this thing around the shows in the US for a couple
of years now but none of the carriers support it. With GSM having such
blanket coverage I don't see many carriers going this way. I can
understand this working in Asia where coverage is only in the majorly
populated areas and even then only outside.




On Mon, 2006-05-22 at 16:53 +0800, Sam Tam wrote:
  
 
 Well I think we all need to look at something like this first.
 We will be one of the first people in Europe who will be selling this.
 If anyone is interested do drop me an email.
 
  
 
 Picture of the phone can be found here.
 
 http://cyber-telecom.net/wifi-gsm.jpg
 
  
 
 GSM / VoIP Over WiFi Dual-Mode Phone
 
 CYBER-TELECOM released the world first commercial GSM/VoIP Over WiFi
 dual-mode smart phone, in March 2006. With a tri-band GSM/GPRS (Class
 10) radio and an IEEE 802.11b WLAN chipset, It enables end customers
 to enjoy broadband multimedia services at WLAN covered homes, offices,
 hot spots/zones as well as reliable GSM/GPRS service anytime anywhere.
 It shows an outstanding performance in power management, mobility
 management, security, mobile VoIP, and voice quality, no matter what
 kind of access points it connects, as the result of CYBER-TELECOM
 Wireless's advanced technologies solving the critical problems of VoIP
 Over WiFi. the phone has passed most of regulation certification
 programs and has done interoperability testing with over 40 VoIP
 service providers, system integrators, and infrastructure equipment
 vendors worldwide. the phone is an ideal device for fixed mobile
 convergence. 
 
 Hardware Specification
 
 Intel PXA271 processor with embedded Linux
 2.4 inch TFT touch screen, QVGA, 260k Colors
 Built-in speaker/microphone, 2.4mm stereo and headset
 1.3M pixel CMOS camera
 USB slave 
 Mini SD
 1100 mAh Li-ion battery 
 
 GSM Specification
 
 Frequency bands: 900/1800/1900 MHz
 GPRS Class 10
 SMS, MMS, WAP applications
 FTA/CTA certification
 FCC/CE certification
 
 WLAN Specification
 
 IEEE 802.11b 
 RF channels: US: 11, ETSI: 13, Japan: 14
 High-gain internal antenna
 WEP 64/128 bits, WPA, 802.1x
 EAP PSK/LEAP/PEAP/TTLS/SIM
 Power saving modes
 Fast roaming between access points
 
 VoIP Specification
 
 SIP: IETF RFC 3261
 Codec: G.711, G.729a/b, G.723 
 Acoustic echo cancellation
 Dynamic jitter buffer
 Voice activity detection
 Stun-based NAT traversal
 
 Input Methods
 
 Handwriting Recognition
  English
  Chinese
  Numeric characters
 Soft Keypads
  Qwerty
  Standard phone dialpad
  Symbol
 
 Power Management Features
 
 Standby time
 100 Hours (GSM on, WLAN on)
  200 Hours (GSM on, WLAN off)
 Talk time
  VoIP Over WiFi: 3.3 Hours
  GSM: 7.8 Hours
 MP3 play time
  5.8 Hours (GSM on, WLAN on)
  6.2 Hours (GSM on, WLAN off)
 
 Fixed Mobile Convergence Features
 
 Simultaneously activated GSM and WLAN air interfaces
 Handling simultaneously GSM and VoIP Over WiFi incoming calls
 SIP-based seamless handover between GSM/VoIP Over WiFi
 Automatic/manual switch for out-going calls between GSM and VoIP Over
 WiFi
 Automatic/manual switch for data applications using GPRS or WLAN
 Unified phone book for both GSM and VoIP Over WiFi.
 Unified GUI for applications (phone, E-mail, browser, QQ)
 
 Call Features
 
 Call hold
 Call waiting
 Call mute
 Call forward
 Call transfer
 3-way conference
 Voice mail
 SMS over SIP
 Phone book - (1000 entries with photos)
 Incoming call prompt with picture
 View phonebook during call
 Enter sketch pad during call
 Adjust volume during call
 Auto-answer/flip answer
 Quick silence
 Turbo dial
 Manual/Auto/Earphone redial
 Call history (20 entries)
 
 Data Application Features
 
 POP3 E-mail client (SSL support)
  100 full E-mails with attachments up to 200KB
  Document viewer for MS-Office and PDF files
 Web browser: HTML4.01, JAVAScript1.5, SSL3.0, HTTP1.1, CSS1.0
 Instant messaging: QQ
 
 Multimedia Features
 
 Video format: MP4, 3GPP
 Audio format: MP3, WAV, MIDI, AMR
 Picture format: WBMP, BMP, JPEG, GIF
 Camcorder: QVGA, QCIF
 Media Player
  Audio: MP3 player
  Video: up to 30 frames/second QVGA MP4/3GPP
 
 PIM Features
 
 Calendar
 Schedule management
 Alarm clock
 Voice recorder
 World time
 Currency converter
 Anniversary
 
 Other Features
 
 English - Chinese dictionary
 Calculator
 World time
 Notepad
 Sketch pad
 File transfer

Re: [Asterisk-Users] Centos 4.3 Issues

2006-05-22 Thread Greg Boehnlein

On Mon, 22 May 2006, Greg Oliver wrote:

 Have you tried compiling statically on CentOS 4.2 and running on 4.3?

No. Not really in the plans either. Standard policy w/ Asterisk around 
here is to compile on the box it is going to be running on, under the 
distro it's running on.
 
 I am assuming you have made sure the dist is up to date with patches.
 We do not use 729, so I cannot try it out for you, but we do use CentOS.
 Is it only w/ SVN, or all releases of *?

This happens to be with the 1.2 SVN branch.

-- 
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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Re: [Asterisk-Users] Centos 4.3 Issues

2006-05-22 Thread Greg Boehnlein

On Mon, 22 May 2006, alist wrote:

 Greg,
 
 When I upgraded to 4.3 I experienced problems with some non-asterisk 
 RPM's that were compiled on earlier versions of CentOS 4. Once they were 
 recompiled on a fully updated 4.3 system they worked fine. Have you 
 tried recompiling everything?

We recompiled Asterisk, libpri and zaptel. The one system was an upgrade 
from Centos 4.2 to Centos 4.3, but the other two were installed w/ the 
latest Centos 4.3 ISO downloaded last week.

-- 
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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Re: FW: [Asterisk-Users] WiFi / GSM VoIP Handsets..

2006-05-22 Thread Steve Kennedy

On Tue, May 23, 2006 at 02:50:33AM +0800, Sam Tam wrote:

 Well it is incorrect to say that.
 In places like USA or London, a lot of areas are covered by local wifi
 providers, some are free, some aren't. 
 You then can use them to drop some of your local or international calls
 cheaply by using wifi.

But the point is without operator cooperation, there's no seamless
handover between GSM and WiFi, and the operators don't want to lose the
revenue on the voice, so they are unlikely to support it.

BT have an arrangement with Vodafone for their Fusion service (using an
in-premise Bluetooth basestation and a phone with GSM/Bluetooth), but
they're big enough to force an operator's hand.

For general GSM/WiFi UMA, it's unlikely the (UK) operators will allow
other providers access to their networks, as it reduces their revenues.

They're already p*ssed off enough that they're being forced to reduce
roaming charges (currently on voice - but the EU is likely to look at
data charges which can be extremely costly).

They are desperate to keep revenues.


Steve

-- 
NetTek Ltd  UK mob +44-(0)7775 755503
UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
Skype/GoogleTalk/AIM/Gizmo stevekennedyuk / MSN [EMAIL PROTECTED]
Euro Tech News Blog http://eurotechnews.blogspot.com
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Re: [Asterisk-Users] How to monitor DTMF tones in a call?

2006-05-22 Thread Moises Silva


Im unable to give you SVN instructions since Im not an SVN
experiencied user, I only issue the commands in asterisk.org to get
the latest sources to generate the patches I want to upload to
bugs.digium.com

I think the best you could do is to adapt the patches to the asterisk
version you are using. Or remind me to do it for you in the weekend ;)


Regards

On 5/21/06, Obelix [EMAIL PROTECTED] wrote:

Quoting Moises Silva [EMAIL PROTECTED]:

I downloaded and compiled this trunk version - Asterisk SVN-trunk-r28970. The
DTMF events show up in the logging system after I configured logger.conf to
output them, but they are not showing up in the Events.

On checking the SVN for the 6082 patch I saw a branch ../team/jcollie/bug6082.

I don't know what revision that branch is based on. Will compiling that branch
give me the facility?

I am not that familiar with the SVN workings, but if you give the instructions
to follow and may be a revision number or some other parameters to work with I
will be able to do the rest myself.


 You can check that info in www.asterisk.org or voip-info.org

 If you have problems applying the patch let me know, may be I can make
 you a patch for the 1.2.7.1 specially.

 Regards

 On 5/19/06, Obelix [EMAIL PROTECTED] wrote:
  Quoting Moises Silva [EMAIL PROTECTED]:
  Hi,
 
  I am ready to try out this patch, both PlayDTMF and SendDTMF and want to
 know
  which branch I should work from.
 
  I am not quite experienced with compiling from SVN directly and would like
 to
  know whether to download the latest 1.2.7.1 and apply the patch to it or
 use
  the latest from SVN.
 
  Can you give me a list of commands I should apply to SVN?
 
  /Obelix
 
 
   I have uploaded a patch for some manager events that allow to know
   when DTMF has been received or sent. Please take a look at this:
  
   http://bugs.digium.com/view.php?id=6082
  
   and if you can, test it and report feedback. Im having problems to
   call the attention of bug marshalls for comitting this change. I think
   this week i will enter to IRC in asterisk-dev to try to make that
   bugmarshalls pay attention to it.
  
   Best Regards
  
   On 4/30/06, Obelix [EMAIL PROTECTED] wrote:
   
Is there a way to monitor the DTMF tones on a channel?
   
I have a prepaid application working in asterisk. When the user dials a
   call and
wants to cancel the call before it is answered, there is now way to do
 it
without hanging up and redialling the access number.
   
Is there way to monitor a sequence of DTMF tones and cancel the call?
   
If I use a SIP gateway or proxy rather than dial asterisk directly will
   that be
possible?
   
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Re: [Asterisk-Users] Events offered by

2006-05-22 Thread Moises Silva


just execute:

grep -r 'manager_event' ./

into the asterisk source code tree and you will know. may be in
voip-info someone has documented the manager events

On 5/21/06, Obelix [EMAIL PROTECTED] wrote:


Which Actions and events to the read/write options in manager.conf give access
to, ie the options below.

read = system,call,log,verbose,command,agent,user
write = system,call,log,verbose,command,agent,user

Are they documented somewhere?

/Obelix
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Re: [Asterisk-Users] Office to Office via IAX2 problems

2006-05-22 Thread Lacy Moore - Aspendora

SInce you say it was working, I am assuming that both officea.kicks-ass.net and officeb.kicks-ass.net resolves to the real IP address and not an internal address, correct? 


Also, are you providing DNS or someone else? Is this domain registered to you? I ask that because if it is not, and you are not providing DNS, it may be resolving to another IP address. But, since you said it is the same using an IP address, this should not be the real issue.

I'm not sure this would really have anything to do with it, but, if it was me, I would not have the two offices on the same subnet. I'd use 192.168.1 for one and 192.168.2 for the other. It just keeps things a little simpler routing wise.



On 5/22/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Thanks.I had already tried going into Office B box and change host=(office A's IP address), performed iax2 reload, and this did not work either.
At 10:51 AM 5/22/2006, you wrote:Don't specify the remote side by name, specify it by IP address.Ifasterisk experiences even 1 dns failure it will not try again untila reload/restart/whatever.
[EMAIL PROTECTED] wrote:SETUP:OFFICE A:iax_additional.conf:[lpeaus]username=lpeaus-usertype=peer
secret=secretqualify=yeshost=officeb.kicks-ass.netcontext=from-internal[lpenb-user]type=usersecret=secret
host=officeb.kicks-ass.netcontext=from-internalOFFICE B:iax_additional.conf :[lpeaus-user]type=user
secret=secrethost=officea.kicks-ass.netcontext=from-internal[lpenb]username=lpenb-usertype=peer
secret=secretqualify=yeshost=officea.kicks-ass.netcontext=from-internal___
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Re: [Asterisk-Users] Office to Office via IAX2 problems