[asterisk-users] wad happen if there is nothing wrong in conf but still can't make calls?

[Cindy Tan]

may i noe wad can i do because my asterisk is working fine but the calls cannot 
proceed between 2 asterisk servers.
hope anyone can help me solve this major problem.
 
thanks a lot in advance
 
Regards
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[Asterisk-Users] Asterisk settings for roaming users

[Andy Tan]

Hi,

like to know which configurations are most suitable for roaming users
accessing from various external environments? As an example, should I
use nat=yes in sip.conf when the end device could be connecting from
behind nat with private ip or with a public ip?

Appreciate any suggestions. Thanks.

Regards
Andy Tan
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Re: [Asterisk-Users] billing with PostgreSQL

[Andy Tan]

Hi Joao,

some billing solutions are listed here -
http://www.voip-info.org/wiki/view/Open+Source+Billing+Systems

IIRC, none works with PGSQL. My opinion is that considering the
importance of billing, it's better to develop a customised solution.
That way, you would have full understanding and confidence in it.
References to other systems can be  useful also. Hope it helps.

Regards
Andy Tan

On Wed, 12 Apr 2006 11:15:24 +0100, Joao Pereira
[EMAIL PROTECTED] said:
 Hello to all
 Im looking for a billing tool for Asterisk, that works with PostgreSQL.
 All the tools I found in www.asteriskbilling.com just work with MySQL :(
 
 Do you know a nice billing tool for Asterisk with PostgreSQL?
 
 Thanks
 Joao Pereira
 
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[Asterisk-Users] Trunking Protocols

[Andy Tan]

Hi,

understand that Asterisk supports a variety of signaling protocols like
SIP, IAX2 etc. As a ITSP, which would be the best or most appropiate
protocol to use as trunk to wholesale providers? Know that IAX2 can
conserve bandwidth, but I believe media and signaling are carried with
the same channel/path. That would make off-loading bandwidth utilization
for media impossible. Appreciate any input. Thanks.

Regards
Any Tan
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[Asterisk-Users] Bandwidth Management

[Andy Tan]

Hi,

understand that the bandwidth utilized for each call is dependent on the
codec used, wonder if Asterisk can monitor the total bandwidth utilized
and restrict/reject new calls when the resource is insufficient to
support them reliably?

Regards
Andy Tan
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[Asterisk-Users] Bandwidth Management

[Andy Tan]

Hi,

understand that the bandwidth utilized for each call is dependent on the
codec used, wonder if Asterisk can monitor the total bandwidth utilized
and restrict/reject new calls when the resource is insufficient to
support them reliably?

Regards
Andy Tan
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RE: [Asterisk-Users] Bandwidth Management

[Andy Tan]

Hi Alex,

thanks for the suggestion.

Did some checks, and thought that I could set a global variable to track
the utilized bandwidth.

Wish that there are plans for support to include variables like
SIP_CODEC in other protocols.
 
Regards

On Tue, 11 Apr 2006 12:50:56 -0400, Alexander Lopez
[EMAIL PROTECTED] said:
 Out of the Box probably not but with an AGI script this is very
 doable:
 
 You can have a script that monitors active calls and the Codecs that are
 in use. The script will have to do some math to calculate the bandwidth
 in use and then using the variables in Asterisk, Namely SIP_CODEC. If
 you are using SIP. There has not been a Variable coded for the other
 Technologies at this time.
 
 Alex
  
 
  -Original Message-
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Andy Tan
  Sent: Tuesday, April 11, 2006 9:00 AM
  To: asterisk-users@lists.digium.com
  Subject: [Asterisk-Users] Bandwidth Management
  
  Hi,
  
  understand that the bandwidth utilized for each call is 
  dependent on the codec used, wonder if Asterisk can monitor 
  the total bandwidth utilized and restrict/reject new calls 
  when the resource is insufficient to support them reliably?
  
  Regards
  Andy Tan
  --
Andy Tan
[EMAIL PROTECTED]
  
  --
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[Asterisk-Users] Opensource solutions to SPIT

[Andy Tan]

Hi,

I have been listening to Blue Box: The VoIP Security Podcast -
http://www.blueboxpodcast.com, and thought that SPIT could pose a
problem if not already one. Like to know if there are any OSS solutions,
within Asterisk or can integrates well with it, that focus in this area?

Regards
Andy Tan
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[Asterisk-Users] Maximum duration of Voicemail messages

[Andy Tan]

Hi,

I had set maxmessage in voicemail.conf to 18. However, the
tt-allbusy.gsm message(8.44 sec) sent to vm was cropped at the 
beginning to 8.00 sec exactly. Appreciate any advice on how the duration
can be increased for vm recordings. Thanks.

Regards
Andy Tan
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[Asterisk-Users] Asterisk and .NET

[Alvin Tan]

Hi,

Are there any Asterisk interfaces with .NET?

Thanks,
Alvin
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[Asterisk-Users] install software version to mediatrix 1204 (how to)

[eder tan]

i'm new here and i need help on how where can i get
software version 4.0.x of the mediatrix and how can i
install it...

mediatrix unit im using has a software version of
2.4.9.57. i would like to use H.323 not SIP...

please need help asap!... hope to hear from anyone of
you soon..

thanks in advance!

--
eder



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RE: [Asterisk-Users] VOIP Service Providers

[tan]

Check the Digium web site for a number of voip providers.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bruno
Fontana
Sent: 27 May 2004 00:51
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] VOIP Service Providers


Ed Mansouri wrote:
 Hello,
 
 I am looking for a VOIP Service Provider to work with getting started 
 with Asterisk.
 
 Does anyone have any brief recommendations?  Ease of use and support 
 are the key criteria.
 
 -
 Ed Mansouri
 Ucompass - http://www.ucompass.com
 
 Make sure we stay connected to you
 Add yourself to the Ucompass Address Book 
 http://support.ucompass.com/addressbook.html
 
 Committed to Building Profitable E-Learning Enterprises
 Phone: (850) 297 1800 x 201
 FAX: (850) 553-9252
 
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I'm currently using iptel.org as SIP provider to experiment with. It's 
really easy to use (to configure in fact). I don't know anything about 
support they provide. Every time I had problems with SIP someone in the 
list was kind enough to help me.
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RE: [Asterisk-Users] 100 analog phones?? HOWTO?

[tan]

4 x Mediatrix 1124 VoIP Gateways?

http://www.voiptalk.org/products/product_info.php?cPath=31products_id=7
2

Tan


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul Mahler
Sent: 25 May 2004 03:33
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] 100 analog phones?? HOWTO?


I have had good experiences with Adit. Their customer service and
documentation are excellent. 

Paul


Paul Mahler 
[EMAIL PROTECTED]   
Signate, LLC
PO Box 60430
Palo Alto, CA
 94306

 VoIP Systems, Training  Consulting

 

 

 

 

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Jeff Gustafson
 Sent: Monday, May 24, 2004 4:21 PM
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] 100 analog phones?? HOWTO?
 
   Does anyone know the best approach to take for handling
 100 analog phones?  It seems to me that a chassis like 
 Carrier Access or Adtran would work.  The chassis would do 
 much of the hard work of converting the analog sound to data.
   Any recommendations on hardware for the chassis?
 
   ...Jeff
 
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RE: [Asterisk-Users] SNOM200 + 2.05a Firmware + G729b BUG

[tan]

If using gsm, comment out the dtmfmode line in your sip.conf entity
(i.e. take the default) and it should work fine.

Tan
Telappliant.com

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of nicolas
Sent: 13 May 2004 09:22
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] SNOM200 + 2.05a Firmware + G729b BUG


If i use the snom 200 with firmware 2.05a (not tested with 2.04) and the
G739b codec.

Then the keys on the snom do not work with gsm it is ok.

greetings
nicolas

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[Asterisk-Users] extensions in mysql

[Jeffrey A. Tan]

I have already set up a mysql server and I can already use the sip configuration
from mysql, but I'm still having problem with my extensions in mysql. I have
followed the instructions in

http://www.voip-info.org/wiki-Asterisk+extensions+from+mysql

but I still can't use the extensions in mysql. From the sip-mysql , I had to add
dbname, dbhost, dbuser and dbpass in the sip.conf under the general entity. I
also added these four in the extensions.conf, but it still doesn't work. In
using sip-mysql, i have to enable SIP_MYSQL_FRIENDS. Is there anything I have
to do like the one in sip-mysql to be able to use extensions in mysql?
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RE: [Asterisk-Users] Help choosing a UK IAX provider

[tan]

Craig,

2mb up/down with QoS doesn't mean anything, especially when you hit the
Internet. What is better is to look at the exact route of your calls and
then determine whether maybe there are some other issues. For instance,
we had a customer with Ciscos who was reporting choppy audio. However,
this was down to a bug in asterisk
(http://bugs.digium.com/bug_view_page.php?bug_id=0001374) and cvs
updating fixed the problem.

Tan
Telappliant.com

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Craig
Waddington
Sent: 21 April 2004 15:38
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Help choosing a UK IAX provider


Yes, but, I am talking about this world.

Ive got 2mb up/down with qos, just need another (good) provider.

If I can try a few and see which is best.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Kennedy
Sent: 21 April 2004 15:32
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Help choosing a UK IAX provider

On Wed, Apr 21, 2004 at 03:24:06PM +0100, Craig Waddington wrote:

 Currently using voiptalk.org and the quality is getting really bad. I 
 would like a second provider preferably in UK, anyone got any
suggestions?

That's the trouble with running VoIP over contended public Internet.
Find someone who can offer you connectivity with QoS and then has QoS
across their network for VoIP traffic.

Or find someone with infinite bandwidth.


Steve

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RE: [Asterisk-Users] Help choosing a UK IAX provider

[tan]

In the UK, with the sort of equipment that BT has in its network, you're
lucky to even get adsl going through! ISPs can only provide QoS up to a
certain boundary. After that it is out of their control!

Tan


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling
Sent: 21 April 2004 15:44
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Help choosing a UK IAX provider


On Wed, 2004-04-21 at 09:32, Steve Kennedy wrote:
 That's the trouble with running VoIP over contended public Internet.

 Find someone who can offer you connectivity with QoS and then has QoS 
 across their network for VoIP traffic.

LOL!  I've not found any providers that offer QoS on their network other
than a small regional ISP that put QoS on their network when we waved
enough money at them.

-- 
  Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a
related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss.

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RE: [Asterisk-Users] VoIP Phone Recommendations

[tan]

We are currently integration testing the wireless Zyxel Prestige 2000W,
and if all goes well we'll have it for sale in 2 weeks. Has anyone any
experience of this SIP device and asterisk?

Tan
telappliant.com


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adams, Gavin
Sent: 15 April 2004 12:28
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] VoIP Phone Recommendations


 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users- 
 [EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith
 
  AS for the WiSIP, STAY WELL AWAY!! I have 3 and they all started to 
  exhibit the same problem after only 4 weeks out of the box. They
become
  deaf and consistantly miss the call setup. The processor in the
phone is
  very slow which can be demonstrated by the painfullnes endured when 
  navigating the menu's or configuring its web interface. None of the 
  features that Pulver claims work (hold, transfer, call waiting,
second
  line etc) and he will not respond to any critisism of the product.
You
 can
  forget trying to talk to their tech support too. She sucks.
 
 I second this, although I don't have any issues with speed -- Mind you
I
 am
 not using any kind of encryption, which I hear really bogs it down.
 
 Hold works fine for me.  No transfer, call waiting or second line
though.
 I
 too have had no success in getting phone calls or email responded to.
The
 volume is very low, even when cranked up.  The standy time sucks ass, 
 although I consistently get 3h of talk time out of it.  Both the
display
 contrast and the display backlight are substandard, IMO.
 
 I'm gonna dump this phone on ebay and try one of the other wireless
SIP
 phones.

I third this. Functionally it's a terrible implementation and not
something I would ever give to users. Anyone want to buy it at a low,
low price?

I'll think twice before buying another Pulver product.

Regards,

--- Gavin
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RE: [Asterisk-Users] Zaptel/PRI problem

[tan]

Hi,

Glad that your problem was solved, but we are still exerpiencing a
similar problem but our interrupts show:

  0: 162034  XT-PIC  timer
  1:234  XT-PIC  keyboard
  2:  0  XT-PIC  cascade
  5:782  XT-PIC  eth1
  7:   7448  XT-PIC  eth0
  8:  1  XT-PIC  rtc
 10:  0  XT-PIC  usb-ohci
 11:  19330  XT-PIC  ide0, ide1
 12:468  XT-PIC  PS/2 Mouse
 14:  0  XT-PIC  ide2
 15: 967550  XT-PIC  t1xxp
NMI:  1
ERR:  0

Any ideas?
Tan


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: 10 April 2004 17:23
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Zaptel/PRI problem


To the list ...
The problem appears to be fixed.  Short answer: interrupts. When looking
at cat /proc/interrupts it was seen that the wct1xxp card was sharing
interrupts with a.o. the eth0 main ethernet driver. The solution we
simple: move the T1 card to another PCI slot. Upon checking cat
/proc/interrupts again, the card now had its very own interrupt.  So far
I have NOT seen any PRI notices or warnings.  User experience has been
limited due to a light workload on Friday, and, of course, this being
Easter weekend. Since this is such a recurring problem (from witnessing
list postings), perhaps digium should have a notice or something about
this when they ship the cards out.  Especially a relative newbie (first
T1 on an asterisk box) like me is likely to even look at this interrupts
situation. 
Especially when there are other issues (zaptel.conf and
zapata.conf) to get a functional system.  
Maybe this deserves a wiki input.  In any case,
Happy Easter
WW
 
- Original Message Follows -
 Dimitri,
 I just got off the phone with digium. Here's what I (from
 my notes) the event codes mean
 Event 4: Alarm detected
 Event 5: Alarm cleared
 Event 6: Abort HDLC Frams
 Event 8: Bad HCS
 The 6  8 which occur sporadically are possibly causing
 the observed symptoms.
 Now ... what causes 6  8 is the question.
 Interrupt conflicts was one suggested possibility.
 Another possibility is 'stuff' from the Telco which is not
 understood / mis-understood by the driver.
 I'll keep the list posted.
 Willy
 
 - Original Message Follows -
  Dear Willy
  i notice the same problem with my E100P using the latest cvs 
  zaptel driver i  have try every type of config in /etc/zaptel.conf 
  to check if i have missed something in timing conf but nothing... 
  Digium help...
  :-) thanks in advance Dimitri
  
  On Thursday 08 April 2004 23:07, [EMAIL PROTECTED]
   wrote: Chris,
   Thank you for posting this.  Since it concerns my 'production' 
   system, let me comment.  After 'downshifting' to a previous 
   release (for no good reason other than desperation and teh fact 
   that an earlier list entry had commented that it cleared up
   the problems) I am sad to report that the system
   failed again. Miscellaneous throughout the day:
   Apr  8 13:41:27 WARNING[-1210639440]: PRI: Read on 32
   failed: Unknown error 500
   Apr  8 13:41:27 NOTICE[-1210639440]: PRI got event: 8
   on span 1
   Apr  8 13:41:27 WARNING[-1210639440]: PRI: Read on 32
   failed: Unknown error 500
   Apr  8 13:41:27 NOTICE[-1210639440]: PRI got event: 6
   on span 1
   Apr  8 13:42:07 WARNING[-1210639440]: PRI: Read on 32
   failed: Unknown error 500
   Apr  8 13:42:07 NOTICE[-1210639440]: PRI got event: 8
   on span 1
   Apr  8 16:44:01 WARNING[-1210631248]: PRI: Read on 34
   failed: Unknown error 500
   Apr  8 16:44:01 NOTICE[-1210631248]: PRI got event: 6
   on span 1
   Apr  8 16:44:01 WARNING[-1210631248]: PRI: Read on 34
   failed: Unknown error 500
   Apr  8 16:44:01 NOTICE[-1210631248]: PRI got event: 6
   on span 1
  
   Then this -- possibly not related ?
  
   Apr  8 16:51:45 WARNING[-1137157200]: Maximum retries exceeded on 
   call [EMAIL PROTECTED] for seqno 102 (Request)
   Apr  8 16:51:45 WARNING[-1137157200]: Maximum retries
   exceeded on call [EMAIL PROTECTED] for
   seqno 103 (Request)
   Apr  8 16:51:45 WARNING[-1137157200]: Maximum retries
   exceeded on call [EMAIL PROTECTED] for
   seqno 104 (Request)
   Apr  8 16:51:46 WARNING[-1137157200]: Maximum retries
   exceeded on call [EMAIL PROTECTED] for
   seqno 105 (Request)
  
   And finally, I'll show you a RESTART log
  
   Apr  8 17:41:32 WARNING[-1085030272]: Ignoring port
   for now Apr  8 17:41:33 WARNING[-1085030272]: XXX I
   don't work right with non-full duplex sound cards XXX
   Apr  8 17:41:33 WARNING[-1189983312]: Read error on
   sound device: Resource temporarily unavailable
   Apr  8 17:41:33 ERROR[-1085030272]: Unable to load
   config iax1.conf
   Apr  8 17:41:38 NOTICE[-1221452880]: Alarm cleared on channel 1
   Apr  8 17:41:38 NOTICE[-1221452880]: Alarm cleared on
   channel 2
   Apr  8 17:41:38 NOTICE[-1221452880]: Alarm cleared on
   channel 3
   Apr

RE: [Asterisk-Users] Zaptel/PRI problem

[tan]

Here is what we get:

Apr 10 18:10:34 WARNING[-1179604048]: chan_zap.c:6026 zt_pri_error:
PRI: Read on 24 failed: Unknown error 500
Apr 10 18:10:34 NOTICE[-1179604048]: chan_zap.c:6740 pri_dchannel:  PRI
got event: 8 on span 1

We were getting around 5 messages per second. I turned off the usb
interface in the bios and now the messages have greatly reduced in
number e.g. a few messages each minute.

We only see the errors on the console. What frequency of messages were
you getting? Sound dropping in voice calls may be related to something
else.

Tan




-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: 10 April 2004 17:51
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Zaptel/PRI problem


Tan,
Scary ...
What we used to see was quite few (and sporadic) notices and warnings in
the /var/log/messages file reporting PRI trouble.  Especially event 6
and event 8 if I recall.  These notices and warnings have disappeared
since we resolved the interrupt issue.  Because the implementation is an
office PBX, and the office had light call volume Friday afternoon and
closed for the weekend, I do NOT know if users might still experience
sound dropping on voice calls.  
My question to you: when you state 'similar problems', you
mean dropping voice (sound), or having spurious PRI events,
or both.
Cheers,
WW

- Original Message Follows -
 Hi,
 
 Glad that your problem was solved, but we are still exerpiencing a 
 similar problem but our interrupts show:
 
   0: 162034  XT-PIC  timer
   1:234  XT-PIC  keyboard
   2:  0  XT-PIC  cascade
   5:782  XT-PIC  eth1
   7:   7448  XT-PIC  eth0
   8:  1  XT-PIC  rtc
  10:  0  XT-PIC  usb-ohci
  11:  19330  XT-PIC  ide0, ide1
  12:468  XT-PIC  PS/2 Mouse
  14:  0  XT-PIC  ide2
  15: 967550  XT-PIC  t1xxp
 NMI:  1
 ERR:  0
 
 Any ideas?
 Tan
 
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf
 Of [EMAIL PROTECTED]
 Sent: 10 April 2004 17:23
 To: [EMAIL PROTECTED]
 Subject: Re: [Asterisk-Users] Zaptel/PRI problem
 
 
 To the list ...
 The problem appears to be fixed.  Short answer:
 interrupts. When looking at cat /proc/interrupts it was
 seen that the wct1xxp card was sharing interrupts with
 a.o. the eth0 main ethernet driver. The solution we
 simple: move the T1 card to another PCI slot. Upon
 checking cat /proc/interrupts again, the card now had its very own 
 interrupt.  So far I have NOT seen any PRI notices or warnings.  User 
 experience has been limited due to a light workload on Friday, and, of

 course, this being Easter weekend. Since this is such a recurring 
 problem (from witnessing list postings), perhaps digium should
 have a notice or something about this when they ship the
 cards out.  Especially a relative newbie (first T1 on an
 asterisk box) like me is likely to even look at this
 interrupts situation. 
 Especially when there are other issues (zaptel.conf and
 zapata.conf) to get a functional system.  
 Maybe this deserves a wiki input.  In any case,
 Happy Easter
 WW
  
 - Original Message Follows -
  Dimitri,
  I just got off the phone with digium. Here's what I
  (from my notes) the event codes mean
  Event 4: Alarm detected
  Event 5: Alarm cleared
  Event 6: Abort HDLC Frams
  Event 8: Bad HCS
  The 6  8 which occur sporadically are possibly causing
  the observed symptoms.
  Now ... what causes 6  8 is the question.
  Interrupt conflicts was one suggested possibility.
  Another possibility is 'stuff' from the Telco which is
  not understood / mis-understood by the driver.
  I'll keep the list posted.
  Willy
  
  - Original Message Follows -
   Dear Willy
   i notice the same problem with my E100P using the latest cvs  
   zaptel driver i  have try every type of config in /etc/zaptel.conf

   to check if i have missed something in timing conf but nothing...

   Digium help... :-) thanks in advance Dimitri
   
   On Thursday 08 April 2004 23:07, [EMAIL PROTECTED]
wrote: Chris,
Thank you for posting this.  Since it concerns my 'production'  
system, let me comment.  After 'downshifting' to a previous  
release (for no good reason other than desperation and teh fact

that an earlier list entry had commented that it cleared up
the problems) I am sad to report that the system
failed again. Miscellaneous throughout the day: Apr 
8 13:41:27 WARNING[-1210639440]: PRI: Read on 32
failed: Unknown error 500 Apr  8 13:41:27
NOTICE[-1210639440]: PRI got event: 8 on span 1
Apr  8 13:41:27 WARNING[-1210639440]: PRI: Read on
32 failed: Unknown error 500
Apr  8 13:41:27 NOTICE[-1210639440]: PRI got event:
6 on span 1
Apr  8 13:42:07 WARNING[-1210639440]: PRI: Read on
32 failed: Unknown error 500
Apr

RE: [Asterisk-Users] Sipcall.co.uk [*]

[tan]

If you're using asterisk, then why would you use SIP? Use an IAX
provider. There are a few. Check Digium's web site where they have a
list.

Tan


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt Bridges
Sent: 31 March 2004 09:15
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Sipcall.co.uk  [*]


EEK!  Doesn't handle SIP correctly, did they explain what they meant?

Cheers

Matt 

-Original Message-
From: Dave Cotton [mailto:[EMAIL PROTECTED] 
Sent: 31 March 2004 07:19
To: Asterisk List
Subject: Re: [Asterisk-Users] Sipcall.co.uk  [*]

On Wed, 2004-03-31 at 01:33, Matt wrote:
 Hello all.
 
 Has anyone managed to get SIPCALL.co.uk's service working with the [*]
box?
 
 I've managed to register with other SIP providers but not SIPcall.
 
I spent a lot of time trying to get * to connect to SIPcall, I even got
directly in contact with the support depart of the supplier of the
hardware, who informed me that it is because * does not handle SIP
correctly, as I had no trouble connecting to SIPPhone, Nikotel, VoIPTalk
etc I decided to drop it. 
 YMMV

--
Dave Cotton [EMAIL PROTECTED]

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RE: [Asterisk-Users] PRI issues with TE410P

[tan]

Before going to gentoo or another version of linux, try the following:

1) turn hyperthreading off in the BIOS. It's probably called something
like virtual cpus.

2) Use a vanilla kernel from kernel.org e.g. 2.4.25.

We used to have times when the system would just hang. Using a
non-redhat kernel seems to have resolved that. We also use Gentoo, but I
warn you that it is not the easiest flavour to set up!

Hope that helps.

Tan
Telappliant.com
Voiptalk.org



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Azher Amin
Sent: 24 March 2004 08:49
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] PRI issues with TE410P


Dear,

Are you using Redhat9.0, coz I discussed about this yesterday with mike
in Digium, and he said that this is normal with Redhat (due to its SMP
handling problems), further he suggested to shift either to redhat 8 or
debian. You can try debian or gentoo. 

I will try gentoo tomorrow and will see this behavior again.

Regards
Azher

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of reseaux
Sent: Wednesday, March 24, 2004 2:32 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] PRI issues with TE410P

Dear 
i have found the same problem last sunday in my * box and freeze
itself. My Box is Pentium 4 HT with 2 CPU and i see it in
/proc/interrupts
-
  CPU0   CPU1   CPU2   CPU3
  0:   13088533  0  0  0IO-APIC-edge  timer
  1:  6  0  0  0IO-APIC-edge
keyboard
  2:  0  0  0  0  XT-PIC
cascade
  8:  1  0  0  0IO-APIC-edge  rtc
 12: 12  0  0  0IO-APIC-edge  PS/2
Mouse
 14:2146897  0  1  0IO-APIC-edge  ide0
 15:118  0  1  0IO-APIC-edge  ide1
 16:  0  0  0  0   IO-APIC-level
usb-uhci
 18:  0  0  0  0   IO-APIC-level
usb-uhci
 19:  0  0  0  0   IO-APIC-level
usb-uhci
 28: 118357  0  0  0   IO-APIC-level  eth0
 48:  130880536  0  0  0   IO-APIC-level  t4xxp
NMI:  0  0  0  0
LOC:   13088959   13088958   13088958   13088958
ERR:  0
MIS:  0

What we think about?
Thanks in advance
Dimitri
On Sunday 21 March 2004 15:40, Azher Amin wrote:
 Hi,

 I am having some problems mentioned below, the box is in production
live
 environment with traffic around 30 - 100 calls.

 I am running T/E410P in a Dual P4 xeon with HT disabled. I am using 
 zaptel 0.9.0 and asterisk stable 1 release. There is no gui, just
mysql,
 perl (small script) and asterisk.

 System runs very smoothly if the calls are around 40-50 and comes one
by
 one , however sometimes at immediate load of around 30 more calls ...
I
 get the following processes in the ps -ax, and asterisk starts droping

 the calls, irq misses rise and console shows lot of pri errors (which 
 donot occur in a smooth load of around 50 calls).

 Can someone explain why this happens .. however these get cleared once

 the channels are handled.

 20992 pts/3S  0:01 zttool
 21412 ?S  0:00 asterisk
 21413 ?R  0:00 asterisk
 21418 ?S  0:00 asterisk
 21419 ?S  0:00 asterisk
 21420 ?S  0:00 asterisk
 21421 ?S  0:00 asterisk
 21422 ?S  0:00 asterisk
 21423 ?S  0:00 asterisk
 21424 ?S  0:00 asterisk
 21425 ?S  0:00 asterisk
 21426 ?S  0:00 asterisk
 21427 ?S  0:00 asterisk
 21429 ?S  0:00 asterisk
 21430 ?S  0:00 asterisk
 21431 ?S  0:00 asterisk
 21432 ?S  0:00 asterisk
 21433 ?S  0:00 asterisk
 21434 ?S  0:00 asterisk
 21435 ?S  0:00 asterisk
 21436 ?S  0:00 asterisk
 21437 ?S  0:00 asterisk
 21438 ?S  0:00 asterisk
 21439 ?S  0:00 asterisk
 21440 ?S  0:00 asterisk
 21441 ?S  0:00 asterisk
 21442 ?S  0:00 asterisk
 21443 ?S  0:00 asterisk
 21444 ?S  0:00 asterisk
 21445 ?S  0:00 asterisk
 21446 ?S  0:00 asterisk
 21447 ?S  0:00 asterisk
 21448 ?S  0:00 asterisk
 21449 ?S  0:00 asterisk
 21451 ?S  0:00 asterisk
 21452 ?S  0:00 asterisk
 21453 ?S  0:00 asterisk
 21454 ?S  0:00 asterisk
 21455 ?S  0:00 asterisk
 21456 ?S  0:00 asterisk
 21457 ?S  0:00 asterisk
 21458 ?S  0:00 asterisk
 21459 pts/2R  0:00 ps -ax
 21460 ?S  0:00 asterisk



 Further I am also

RE: [Asterisk-Users] asterisk: cpu load 99%

[tan]

Make sure you have done a cvs update. We had a cpu problem where we were
hitting 99.9% on every call. An update sorted it out.

Thanks
Tan



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Iain
Stevenson
Sent: 22 March 2004 12:14
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] asterisk: cpu load 99%




--On Monday, March 22, 2004 12:51 pm +0100 Matteo Rancilio 
[EMAIL PROTECTED] wrote:

 You're right :)

 I'm using Asterisk 7.2 on a SuSE 8.2 installation.
 Hardware:
 Dual Intel PIII
 1Gb ram
 AVM Fritz! ISDN card
 SIP
 CISCO Phones
 Codec g711 (switching today to g729)


... and what applications? AGI, Festival?  Festival completely stuffed
my * 
server once.

  Iain



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RE: [Asterisk-Users] NuFone?

[tan]

Since everyone is offering their services then:

USA - £0.016 (~ 2.9c)
UK - £0.016 (~ 2.9c)
Europe - £0.02 (~ 3.6c)
UK 0800 - FREE

SIP / IAX termination. auto-provisioning, web-based billing, call
history, on-line top-up, credit-card payments.

Not US-based though :-(

Tan
www.voiptalk.org
www.iaxtalk.co.uk



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Fraizer
Sent: 17 March 2004 19:36
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] NuFone?


Doug Harris wrote:
  Hi,
 
  Seems like there arn't any alternative to NuFone either ?
 
  Toll free Number + IAX/SIP + 2.9 Cents per minute, no strings
attached. Doug

If you want SIP/IAX termination from someone other than NuFone for the
same 
price, you can contact me.  We can offer that.

John


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RE: [Asterisk-Users] Small office requirements - Can this be done?

[tan]

There is a product called the firebrick which supports bonding separate
adsl lines. Don't know how well it works.

Tan
TelAppliant

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dave Cotton
Sent: 09 March 2004 08:17
To: Asterisk List
Subject: Re: [Asterisk-Users] Small office requirements - Can this be
done?


On Tue, 2004-03-09 at 08:36, WipeOut wrote:
 Simon Coles wrote:
 
 
 
  --On Tuesday, March 2, 2004 9:49 am + Steve Kennedy
  [EMAIL PROTECTED] wrote:
 
  That's the crunch (1.5/512) ... it's actually the 512 which is 
  relevent.
  We haven't set it all up yet, but for our new UK office we've gone
  with ADSL from Andrews  Arnold (http://www.aaisp.net/) who will let

  you bond 2 ADSL lines together to get 512 upstream. We only moved in
a 
  couple of days ago so I haven't had a chance to set it all up yet
:-(
 
 
 
 Nildram also support bonding ADSL lines together, I think they 
 currently
 support up to 4 lines (1Mb upstream 2Mb downstream) and they are
looking 
 at supporting more..

This may be a complete red herring, but, instead of bonding two feeds
from the same provider has anyone tried with two, or even more, feeds
from different providers? I have a possible installation with two
incoming analogue fax lines, one already has ADSL the other could. My
thoughts are to get ADSL from a second provider and then mix them in a
Linux firewall. This should give some protection from outrages at the
provider.
-- 
Dave Cotton [EMAIL PROTECTED]

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RE: [Asterisk-Users] System freeze

[tan]

We've had 2 unexplainable system freezes. 

We have SMP and Redhat 9 2.4.20-20.9. There has been no evidence
anywhere of why our system crashed.

Tan





-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael
Welter
Sent: 09 February 2004 18:39
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] System freeze


RH9.  If it happens again I think I'll drop back to 0.7.1.

Jonathan Biggs wrote:

 Currently in progress of trying to debug similar
 problem on my own system.  Sometimes it happened
 during call transfers,  but this last time,
 it happened all by itself at 4:00 AM, no calls even
 close.  Complete system Freeze, Nothing at all
 workings, except the reset button.
 
 You setup is vastly different from mine to.
 Dual Pentium III  SMP, X100P  Dual TDM400P
 
 What type and version of Linux?
 Mine is RH9  2.4.20-8???
 
 Would love to track this one down...
 
 
 
 
 
 --- Michael Welter [EMAIL PROTECTED] wrote:
 
I have a Gigabyte K7 motherboard with an Athlon
2400+ processor.  Before
the T1 install I had two T100P cards, one for the
channel bank and the
other unused.  This ran perfect for a month.

Last week we installed a new integrated T1 into the
unused T100P (to
replace POTS lines and DSL.)  In BIOS, I disabled
some unused 
peripherals so that each T100P would find its own
unique IRQ.

I also installed the updated asterisk, libpri, and
zaptel sources.

I have seen two system freezes--one on Friday and
one this morning.  The
whole system freezes--no LAN, no phones, no console.
 During this 
morning's freeze there were no calls in progress. 
The logs say nothing.

Has anyone else seen this?  I suspect it isn't an
asterisk problem, but
I would appreciate feedback.

Thanks,
Mike


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 __
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RE: [Asterisk-Users] Re: reload problem

[tan]

We were seeing hanging symptoms when the dns entries in resolv.conf were
not reachable. Don't know if this applies to you.

Tan
telappliant.com

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Philipp von
Klitzing
Sent: 19 January 2004 17:21
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Re: reload problem


Hi!

 Has anyone experienced * hang/exit when issuing -
 asterisk -r -x reload

Yes, see also here and add your comments if applicable:
http://bugs.digium.com/bug_view_page.php?bug_id=725

Philipp

P.S.: Next time please open a new top posting when you create a new
topic 
instead of replying to a totally different issue.


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RE: [Asterisk-Users] Grandstream Quality Survey.... :P

[tan]

For the price, the Grandstream is unbeatable value for money. 

Get firmware version 1.04.26 and you should be fine. This firmware fixes
issues our customers had with phone lockups, nat problems, one-way
audio, stun problems.

Best Wishes
Tan
www.telappliant.com
www.voiptalk.org



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of rnc Info
Lists
Sent: 24 December 2003 12:09
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Grandstream Quality Survey :P



From: Brian West [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Grandstream Quality Survey :P
Reply-To: [EMAIL PROTECTED]
...


I have 2 of these phones and they work fine for my application.  Granted
its not the most intensive use and definatly not the most critical users
but... With all of the companies that are running into cash problems in
the next year I think that the demands for systems that do everything
including make coffee will decrease.  Basic functionality will take the
place of complicated functionality.   Granted GS needs to be more
responsive but if they are going to maintain a low price level we need
to be a bit understanding about the responses If GS phones don't
meet your needs then by all means spend more money on some of the other
brands.  For some of us, GS does meet the requirements and we will
continue to use them.

Robert

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RE: [Asterisk-Users] Grandstream Quality Survey.... :P

[tan]

Try it on one of the phones first. We've tested it and it seems to work
fine. Let me know offline how you get on.

http://www.telappliant.com/grandstream/1.04.26.zip

Thanks
Tan
www.telappliant.com
www.voiptalk.org



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David J
Carter
Sent: 24 December 2003 13:36
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Grandstream Quality Survey :P


Hi Tan,

Can you supply us with 1.0.4.26 firmware?

Regards

Dave

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of
[EMAIL PROTECTED]
Sent: 24 December 2003 12:53
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Grandstream Quality Survey :P

For the price, the Grandstream is unbeatable value for money.

Get firmware version 1.04.26 and you should be fine. This firmware fixes
issues our customers had with phone lockups, nat problems, one-way
audio, stun problems.

Best Wishes
Tan
www.telappliant.com
www.voiptalk.org



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of rnc Info
Lists
Sent: 24 December 2003 12:09
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Grandstream Quality Survey :P



From: Brian West [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Grandstream Quality Survey :P
Reply-To: [EMAIL PROTECTED]
...


I have 2 of these phones and they work fine for my application.  Granted
its not the most intensive use and definatly not the most critical users
but... With all of the companies that are running into cash problems in
the next year I think that the demands for systems that do everything
including make coffee will decrease.  Basic functionality will take the
place of complicated functionality.   Granted GS needs to be more
responsive but if they are going to maintain a low price level we need
to be a bit understanding about the responses If GS phones don't
meet your needs then by all means spend more money on some of the other
brands.  For some of us, GS does meet the requirements and we will
continue to use them.

Robert

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RE: [Asterisk-Users] * Party in Paris

[tan]

Mark,

We're happy to host something in London if you were dropping round these
sides.

Tan
Telappliant.com
Voiptalk.org



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of WipeOut
Sent: 30 November 2003 20:45
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] * Party in Paris


Make sure and let us know anytime you are stopping by London.. :) (Just
not between the 2nd-22nd December cos I will be away)

Later..


Mark Spencer wrote:

I'll be there until jan 5.  The 19th would definitely be too early, 
maybe the 20-22?  Possibly even after the new year, jan 2 or 3.

Mark

On Sun, 30 Nov 2003, zoa wrote:

  

Count me and one of my collegue's in.
How long are you staying in Paris ? The 19th might be a bit early for 
us, but then again maybe not :)

Zoa.



At 23:28 29/11/2003 -0600, you wrote:


I'm coming to Paris Dec 19.  I was wondering if there was any 
interest in having an Asterisk get together in Paris sometime near 
there.  Any one out there interested?  Anyone in Paris who could help

organize something like that? :)

Mark

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RE: [Asterisk-Users] SIP Express Router Asterisk

[tan]

Hi,

We will shortly launch a sip service. Architecture is:

SER: for SIP registration and IP call routing, incoming number
termination, STUN, Nat traversal etc.
Asterisk: outgoing call routing, calling card platform, billing,
extended facilities e.g. voicemail etc.

Works well.

Tan
Telappliant.com
Voiptalk.org


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton
Tinchev
Sent: 27 November 2003 13:19
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] SIP Express Router  Asterisk


There was some issues with Audiocodes MP10x - With both Asterisk and
SER. It was fixed in last firmwire release. Hope it is fixed in Mediant
too. It was general SIP issues...

Ryan Tucker wrote:

 Greetings...
 
 We've been having some interoperability issues between Asterisk and an
 AudioCodes Mediant 2000, and, well, I gotta use the Mediant 2000 
 somewhere.  So, I've been pondering using iptel.org's SIP server (SIP 
 Express Router) as a front end for PSTN calls going out to the
Mediant, 
 while using Asterisk for everything else.
 
 Has anyone done something similar, or anything at all involving SER 
 and
 Asterisk?
 
 Thanks!  -rt
 


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RE: [Asterisk-Users] Which ISDM BRI Card for Asterisk?

[tan]

For production environments we ONLY use the Eicon Diva server card
range, which supports on-board echo cancellation. However it is rather
expensive.

Tan
www.voiptalk.org


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of WipeOut
Sent: 21 November 2003 09:13
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Which ISDM BRI Card for Asterisk?


Daniel ANDRE wrote:

 Hello all,

 I wonder to have some feedback on using ISDN BRI Cards with Asterisk
 and the Echo problem.

 I have tried a simple BRI card with i4l driver and encounter huge echo
 problem. I have tried to solve it with a Sw chocanceller without 
 success. What I'd like to know is wether some of you have used other 
 BRI Cards (I have seen reference to Eicon cards on this list) and if 
 the echo disappeares with these cards?

I would recommend you dump i4l and use a CAPI card with the chan_capi 
driver.. The cheap solution is a AVM FritzPCI card(this is what I use)..

The other solution is the either the Eicon or AVM active cards..

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RE: [Asterisk-Users] ISDN Card Types for Europe

[tan]

Title: Message



We 
deploy the Eicon Diva Server range of cards for production systems as they have 
onboard echo cancellation, and work very well with chan_capi and 
asterisk.

Tan
www.voiptalk.org



  
  -Original Message-From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Ray 
  BurkholderSent: 18 November 2003 16:01To: 
  [EMAIL PROTECTED]Subject: [Asterisk-Users] ISDN Card 
  Types for Europe
  What types of ISDN BRI cards work well in Europe 
  (Guadeloupe, Martinique and France) 
  ? For example: AVM C2 or AVM C4 or eicon 
  Diva server 4 BRI? Any others? Which driver is 
  appropriate?
  Ray Burkholder 
  [EMAIL PROTECTED] 
  http://www.oneunified.net 
  704 576 5101 
  -- Scanned for viruses  dangerous content at One Unified and is believed to be clean. 

RE: [Asterisk-Users] USB handsets/headsets??

[tan]

We sell some bargain basement versions. Check www.telappliant.com


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of WipeOut
Sent: 06 November 2003 12:22
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] USB handsets/headsets??


Anyone got any pointers on where to find USB handsets or headsets that 
can be used as the audio device on a softphone?

Later..

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Re: [Asterisk-Users] Grandstream Budgetone

[Tan Aks]

RRP: $75 for 101, $85 for 102


- Original Message - 
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, August 17, 2003 1:22 PM
Subject: [Asterisk-Users] Grandstream Budgetone


Does anyone know what the Grandstream Budgetone is going for $$$ in the 
US? I didn't immediately see pricing on the phones page.
AJ

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[Asterisk-Users] Configuring iptables to allow sip and dynamically allocate rtp ports

[Tan Aks]

Hi,

We have an asterisk box, with 2 nics, one with internal addressing, 
and the other with a public address. The firewall (iptables) is configured for 
nat routing. Now we want to allow this box to receive sip registrations from the 
internet. Does anyone know if you can use iptables to allow the dynamic creation 
of rtp ports?

T

Re: [Asterisk-Users] Running Asterisk behind NAT?

[Tan Aks]

You can run SIP reliably behind nat with a SIP-aware adsl router. We have
set this up for a few customers, as well as for ourselves and it works fine.
http://www.telappliant.com/intertex_sip_aware.htm.

Tan



- Original Message - 
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, August 12, 2003 5:35 AM
Subject: RE: [Asterisk-Users] Running Asterisk behind NAT?


Yes, you can run Asterisk behind a NAT.
NO, you CAN'T (reliably, easily) run SIP behind a NAT.
For FWD think about using their behindnat and fwdproxy addresses.
Maybe a STUN would help. Also, test your setup infront of NAT also, make
sure they work, before you head behind a NAT.

--
wasim


This mail is confidential  intended solely for the use of the addressee.

On Tue, 12 Aug 2003, Terence Chan wrote:

 I would like to ask if it is possible to run Asterisk behind NAT.  I have
a
 linksys router that forwarded the port UDP 5082 to the local IP of my
 Asterisk box, I got the error 479 when I try to register my Asterisk box
 with FWD. (see detail below).

 Have anyone got Asterisk working behind NAT and successfully registers
with
 FWD?

 Any pointer or information will be appreciated.
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Re: [Asterisk-Users] OT: Grandstream power supplies..

[Tan Aks]

We do the replacement adapters for £12+VAT if interested. You'll still need
the US-to-UK adapter though. Contact me offline if you need one.

You could go to cpc as was suggested. www.cpc.co.uk

Tan
telappliant.com


- Original Message - 
From: WipeOut . [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, August 12, 2003 1:09 PM
Subject: [Asterisk-Users] OT: Grandstream power supplies..


Hi,

Quick question to all the electronics gurus out there..

I unpacked my second GS phone yesterday (had it for about a month!) and set
it up.. This morning the power supply is dead..

I have looked for a new one online (In the UK using Maplin let me know if
you know a better place.) becasue it would probbaly take too long to get one
sent from China or the US and I need to get that station operational again..

There seem to be many choices for power supplies.. Looking on the bottom of
the broken one it is a 5VDC 400mA output.. When I looked online for a new
one the choices are for regulated, unregulated and switch mode power
supplies with the regulated and switch mode ones being VERY much more
expensive than the unregulated ones..

Which kind would do the job??

Also most of the variable output adapters are 4.5v or 6v, Not the required
5v..

Any help would be appreciated..

Thanks..
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Re: [Asterisk-Users] Does Wildcard x100p support BT Caller ID in UK?

[Tan Aks]

We use it, but with no caller id.

Tan
telappliant.com



- Original Message - 
From: Dave Wilson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, August 05, 2003 1:53 PM
Subject: [Asterisk-Users] Does Wildcard x100p support BT Caller ID in UK?


Hi all,

I can't seem to find any info on this anywhere on the web, except that BT
caller ID doesnt use the standard bellcore system in use in the US. So, if
anyone here in the UK is onlist and using the x100p successfully, please let
me know.


TIA,
Dave


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Re: [Asterisk-Users] Snome-200 with Asterisk

[Tan Aks]

Hi,

Some questions:

1) Are you behind NAT?

2) If the answer to (1) is Yes, then be aware that 
if you have the latest firmware (1.16w) then you should choose the the 
appropriate setting under "NAT detection". The "Automatic" setting doesn't seem 
to work some of our customers behind nat. There are options such as UPnP and 
STUN and you will have to choose the appropriate one.

3) If you aren't behind NAT, then my guess is that 
you have a codec issue.

Tan
telappliant.com



- Original Message - 
From: denzel-infotechs 

To: [EMAIL PROTECTED] 

Sent: Friday, August 08, 2003 9:46 AM
Subject: [Asterisk-Users] Snome-200 with Asterisk

hi
 We are using snome 200 IP phone 
with *. It works OK. But after a period of time we can'thear any 
sounds for any icoming or outgoing calls. I've got two of these phones. Same 
symptoms occur to both of these( not at the same time ) and the problem remains 
untilthe phone iscompletely rebooted. Don't know whether this's * or 
Snomes' prob. Any help would be appreciated. Thanks In advance.

Denzel.

[Asterisk-Users] Bad sound quality with G729A on SNOMs

[Tan Aks]

Hi,

We are testing with G729 from remote offices to a 
central asterisk machine. With a SNOM 200 the g729 is terrible. We notice the 
following:

1) when dialling voicemail, the first part of the 
announcement is missed.

2) the sound is very quiet, and sound quality is 
terrible (tinny sound, humming in background).

Has anyone else had the same problems? 

Re: [Asterisk-Users] Grandstream Budgettone 100 102

[Tan Aks]

These guys charge £79+VAT for the 102, and that includes postage to anywhere
in the UK. The $75 doesn't include the VAT tax which has to be paid on top
if shipping to places like the uk.

Tan
(telappliant.com)


- Original Message - 
From: Skuse, Phil [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, July 31, 2003 9:48 AM
Subject: RE: [Asterisk-Users] Grandstream Budgettone 100  102


We bought two 100's for $75 each, and IIRC they charged an extra $100 or so
for shipping to the UK (which seemed a little excessive to me - I asked our
finance people to look into it).

-Original Message-
From: Reed Wade [mailto:[EMAIL PROTECTED]
Sent: 31 July 2003 06:08
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Grandstream Budgettone 100  102



With shipping, I recall my 102 came to $97. I think it was $85 but
I'd need to look it up and don't have the papers nearby.

-reed



At 06:39 PM 7/30/2003 -0500, you wrote:
I was quoted $75 and $85 USD today.

Ricardo Villa
http://www.telesip.net

- Original Message -
From: Joe Cooke [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, July 30, 2003 6:31 PM
Subject: Re: [Asterisk-Users] Grandstream Budgettone 100  102


  I was quoted the $75 and $85 USD prices from Grandstream direct about 2
  months ago.  I'm not sure if it makes a difference, but I live in the
US.
 
  - Joe
  - Original Message -
  From: marrandy [EMAIL PROTECTED]
  To: [EMAIL PROTECTED]
  Sent: Wednesday, July 30, 2003 7:17 PM
  Subject: [Asterisk-Users] Grandstream Budgettone 100  102
 
 
  
   Checking the earlier mails, it stated that the phones were $75 (100) 
$85
   (102) ref :-
  
   http://lists.digium.com/pipermail/asterisk-users/2003-June/013483.html
  
   Well, I just called Ovislink/dgtimes and was quoted $90  $100 and the
  person
   said there was no price change.
  
   Anyone on this list actually bought them at the $75  $85 rate ???
  
   Regards...Martin
   --
   Too much is just enough.
   -- Mark Twain, on whiskey
  
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[Asterisk-Users] SIP Authentication bug?

[Tan Aks]

Hi,

I don't know whether only we are experiencing this 
problem but it seems that if authentication is 
used on a couple of phones, and then the authentication is removed (i.e. remove 
the secret parameter from each of the extensions), then this isn't reflected in 
asterisk after a reload. Instead we actually have to restart asterisk for the 
authentication to be removed.

Has anyone else seen this?

Tan

Re: [Asterisk-Users] UK call termination..

[Tan Aks]

We use our own gateway for h323 and sip shortly. Contact me offline.

Tan



- Original Message - 
From: WipeOut . [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, July 21, 2003 1:43 PM
Subject: [Asterisk-Users] UK call termination..


Hi,

I am looking for call termination in the UK so that I can place calls via my
internet line instead of buying more PSTN lines.. anyone know of amy
providers in the UK.. somthing like nufone.net in the UK would be perfect..

Later..
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Re: [Asterisk-Users] New budgetone firmware

[Tan Aks]

Has anyone noticed that sometimes when you boot the phone you don't hear a
dialling tone? I've logged the issue with Grandstream nevertheless.

Tan



- Original Message - 
From: WipeOut . [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, July 15, 2003 11:17 AM
Subject: RE: [Asterisk-Users] New budgetone firmware


Thanks for that I will give it a try.. only problem is that I have some
extensions that use a # which is going to cause a problem.. I guess I will
have to rethink these.. The ability to transfer is more important..

Maybe I will be able to just replace # with * in the dialplan.. :)

Later..

 For me works, providing that:

 * 'Use # as Dial Key:' must be set to yes

 And to trasnfer you do the following:
 receive a call
 press trasnfer
 dial the exten + #

 the call is trasnferred now.

 if you simply dial the exten + hangup as stated in the manual,
 that doesn't work (also if 'Use # as Dial Key:' is set to no)

 Matteo.


 Il mar, 2003-07-15 alle 10:46, WipeOut . ha scritto:
  Yes, The External NTP issue has been around for a while now.. I was
hoping it would be fixed in this release..
 
  Also the ability transfer a call using the Transfer button is still
broken.. (Unless it requires some special configuration to make it work with
*)... Anyone know??
 
  These problems are not listed in the Known Problems section of the
release notes for .77 release..
 
 
 
   Hi
  
   I upgraded earlier today and so far have found that if the Daylight
Saving
   option is on one hour is added to the time received from the NTP
server
   regardless of date.
  
   This is using my internal NTP server but I can't get it to work with
any
   external NTP server, it simply does not download the date
  
   Other than this I have seen no change since I upgraded.
  
   Although there appears to be no Release notes with this release
  
   Regards
   Paul
  
   -Original Message-
   From: [EMAIL PROTECTED]
   [mailto:[EMAIL PROTECTED] Behalf Of Brancaleoni
   Matteo
   Sent: 14 July 2003 20:42
   To: [EMAIL PROTECTED]
   Subject: [Asterisk-Users] New budgetone firmware
  
  
   Hi.
   Has anyone experienced with the new firmware .77 ?
   There's Day Light Saving time now, but haven't
   time to play with it, till now.
  
   Matteo.
  
   --
   Matteo Brancaleoni
   Espia System Administrator - IT services
   Website : http://www.espia.it
   Email   : [EMAIL PROTECTED]
  
  
  
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Re: [Asterisk-Users] caller id

[Tan Aks]

Use SetCallerID(1234567).

Tan
telappliant.com

- Original Message - 
From: Marian Danisek [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, July 09, 2003 3:23 PM
Subject: [Asterisk-Users] caller id


Hello,

is it possible to change how are caller id on incoming call from isdn,
capi lines displayed od sip phones ? ( e.g. SNOM ) standard is 
[EMAIL PROTECTED] I just want only 1234567 to be displayed. is it
possible ?

regards

Marian





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Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic
Tel: +421-46-5430 754 # Fax: +421-46-5439 144
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Re: [Asterisk-Users] Accurate Billing

[Tan Aks]

Steve,

For analog, isn't it just a case of getting asterisk to listen out for
specific tones such as busy, or ringing. Isn't this what the
callprogress option is for in zapata.conf? I thought that it works for the
US at the moment but no where else.

Tan



- Original Message - 
From: Steven Critchfield [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, July 07, 2003 6:01 AM
Subject: RE: [Asterisk-Users] Accurate Billing


On Sun, 2003-07-06 at 23:17, Kim C. Callis wrote:
 Steve,

 What exactly would be classified as a digital ZAP device?

T1/E1 interfaces, so T100P, E100P, T400P, E400P

If you need to see examples, I could probably dig up CDR records where
busy is indicated, and where no answer is indicated and there is a
definate difference between call duration and stop-start duration.

  -Original Message-
  From: [EMAIL PROTECTED] [mailto:asterisk-users-
  [EMAIL PROTECTED] On Behalf Of Steven Critchfield
  Sent: Sunday, July 06, 2003 8:58 PM
  To: [EMAIL PROTECTED]
  Subject: Re: [Asterisk-Users] Accurate Billing
 
  On Sun, 2003-07-06 at 22:07, [EMAIL PROTECTED] wrote:
   hi everyone,
  
   I know this issue has been raised many times before, i think still
 the
   problem remains. When a call is made through a Zap channel, whether
 it
   is actually made or not (irrespective of whether, engaged, busy, or
   actually answered), asterisk logs it in CDRs as a call made. This
   makes it impossible to do an accurate billing. Has anybody found a
 way
   to overcome this problem, if yes, please let me/us know.
 
  If you are on a digital Zap interface, then it is known. If you are on
  an analog interface, then there is no way to know the other answered
 or
  not.
 
  --
  Steven Critchfield [EMAIL PROTECTED]
 
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Re: [Asterisk-Users] Direct entry to your own voice mailbox

[Tan Aks]

e.g.

exten = 8501,1,VoiceMailMain2(${CALLERIDNUM})

Tan
telappliant.com

- Original Message - 
From: Dan [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, July 07, 2003 4:47 PM
Subject: [Asterisk-Users] Direct entry to your own voice mailbox


Hi,

There is any possibility to dial a specific extension and then enter in your
own mailbox (the one defined for that specific SIP phone) without asking for
the exxtension number but only for the password?

I want to be the same extension for all phones, not a specific one for each
of them.

It is possible to have a time stamp in the recorded message? I want to know
when the  message has been recorded.

Thanks,
Dan


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[Asterisk-Users] Asterisk and Hot Desks??

[Tan Aks]

How about the logon wizard of the snom 100? I think that does something
simlar to what you want. It's designed to allow different people to login to
a single phone.


- Original Message - 
From: WipeOut . [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, July 02, 2003 11:21 AM
Subject: [Asterisk-Users] Asterisk and Hot Desks??


Hi,

Has anyone worked out a way to use Asterisk in a Hot Desk environment??

I have not been able to think of a way for the user to have control over
which IP phone will ring when that users extension is dialed without the
user needing to reconfigure the phone..

Something like this would be cool..

User dials *8555 (or similar) and is prompted to enter their extension and
then password, after successfully validating the user is then prompted for
phone number (being some IP phone ID number or an external Mobile or Home
phone number).. All calls made to that users DID number or extension are now
routed to the registered destination device.. Any calls the user makes from
any IP Phones carry the correct caller ID information as well..

Anyone got something like this or any form of user manageable extension
control or hot desk type solution working?? Or any suggestions how it could
be archived??

Later..
-- 
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http://www.linuxmail.org/
Now with e-mail forwarding for only US$5.95/yr

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Re: [Asterisk-Users] Sip call pickup ?

[Tan Aks]

Just want to know if this feature was implemented.

Also, how do I do a supervised transfer with sip phones?

Tan



- Original Message - 
From: Mark Spencer [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, March 13, 2003 5:26 PM
Subject: Re: [Asterisk-Users] Sip call pickup ?


This feature is in development currently.

Mark

On Thu, 13 Mar 2003, Matteo Brancaleoni wrote:

 Hi, I have a mixed sip-zap evironment
 in my office. I was wondering if is
 possible to do remote call pickup
 from a sip phone, like from zap.

 Any hint?

 Matteo Brancaleoni
 [EMAIL PROTECTED]
 Emmegi System Administrator

 EspiA - EMMEGI Srl - e*solution provider
 Uffici: Via Pascoli, 37
 20129 Milano - Italy
 Sede Legale: Corso Sempione 67
 20149 Milano - Italy
 Tel. +39 0270633354
 Fax. +39 0245487890
 http://www.espia.it

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Re: [Asterisk-Users] Problem with echo

[Tan Aks]

Could you provide details of which sip phones you are using. For instance,
the SNOM 200 has echo problems on firmware ver 1.16b. Upgrading to 1.16k
resolves most of the echo issue.

Tan (telappliant.com)



- Original Message - 
From: Dave Packham [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, July 01, 2003 5:15 PM
Subject: Re: [Asterisk-Users] Problem with echo


Same prob here.   15 SIP phones only get eco when going to the PSTN...

if you find something let me know


Dave

 [EMAIL PROTECTED] 7/1/2003 8:53:13 AM 
Hello,

I can't have asterisk working without echo when I place a call from IP

phone (SIP or H323) to a PSTN Phone. The called number as no problem
with echo but there is a very audible echo in the SIP phone. This
situation occurs either when connected with ISDN card thru i4linux
driver and with my openline card from voicetronix.

Do you have any suggestion fo that?

Regards,

Daniel ANDRE

-- 
Daniel ANDRE (mailto:[EMAIL PROTECTED])
IRIS Technologies - http://www.iris-tech.com
Serveur kwartz - http://www.kwartz.com


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Re: [Asterisk-Users] Minimum budget question ...

[Tan Aks]

Hi,

Yes, each port can be addressed by *, as it behaves like a separate sip /
h323 endpoint. The connector is just a way to allow the voip box to have
24 connections, and they are just standard rj11 connections. Another way is
just to use 3 x 8-port gateways on separate IP addresses.

You can use g729 and run * either with safe_asterisk, or using the screen
command e.g. screen -d -m asterisk -vvvc.

Contact me offline for pricing info.

Tan
[EMAIL PROTECTED]


- Original Message - 
From: Andy Powell [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, June 30, 2003 7:51 PM
Subject: Re: [Asterisk-Users] Minimum budget question ...


Hi Tan,

Thanks for the reply. I'll end up asking a load more questions now...

What sort of prices are we talking about for the 24 port
VoIP gateway?

I assume that each port is individually addresable by *?

As I recall the 24 port gateways tend to be terminated at the FXS side
as some 'wierd' connector (wierd in that it's not rj45/11) do you just
wire this to a patch panel?

What codec is in use to get all 24 ports 'running' at the same time..G729?
Does this cause problems since iirc * needs to run in console mode for
the G729 codec to work properly

Thanks for the info... interesting site too :D

Andy



*** REPLY SEPARATOR  ***

On 30/06/2003 at 19:21 Tan Aks wrote:

Hi,

We provide asterisk-based solutions to customers based in the uk. One of
our
customers (9 users) is trialling our low-end solution which comprises of a
box with 2 x X100P (analogue line) cards installed, and a voip carrier for
outgoing calls. This customer intends to have 13 extensions in his live
scenario. The way to use multiple analogue phones is:

1) get a T100P card and use a T1 channel bank sourced from the US
2) use a couple of TDM400P cards to give 8 extensions, and use IP
phones for the other extensions
3) use a voip gateway to provide up to 24 x analogue extensions per
IP address. VoIP gateways are commonly available and convert analogue lines
into a SIP/H323 VoIP stream.

You can get an E1 terminated with an RJ45. If you have a coax  termination
then you can use a balun to get rj45 connectivity.

Hope that helps.
Tan (telappliant.com)




- Original Message - 
From: Andy Powell [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, June 30, 2003 5:26 PM
Subject: RE: [Asterisk-Users] Minimum budget question ...


Tim,

a good comprehensive answer to the question...certainly gave me a few
things
to think about. I do have a few questions though, since I'm in Europe.

Has anyone in Europe set up something equivalent to what Tim suggested?

What sort of prices did it work out at?

How did you solve the channel bank 'issue' in Europe?

I keep reading that E1 lines are coax terminated, is this correct or do you
usually get a choice from your teleco?

Were there any other issues to contend with?

I'd certainly be interested in the experiences of anyone in Europe...

Thanks

Andy




On 30/06/2003 at 10:55 [EMAIL PROTECTED] wrote:

If this is for commercial use, especially if you are going to be selling
this solution, I would suggest that you don't even offer the choice of
analog lines except in the smallest of offices.  Unless you like to
spend a lot of unbillable time supporting them :)




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Re: [Asterisk-Users] Billsec on CDR

[Tan Aks]

Isn't there any way to make callprogress work for people in Europe? What is
it that is needed to make it work?

T



- Original Message - 
From: Martin Pycko [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, June 19, 2003 11:36 PM
Subject: Re: [Asterisk-Users] Billsec on CDR


It has to do with the fact that with analog channels like FXO
we don't have a way to tell whether the call has been answered or not.
So after the interfaces sends the called number we assume that the
call got answered. This happens unless you have callprogress=yes
in zapata.conf. But it's designed to be working only in US.

Martin

On Thu, 19 Jun 2003, Dan Fernandez wrote:

 I have an X100P and when I place calls to the PSTN which are not answered,
the Billsec field of the CDR still logs the seconds that the phone rang.

 Can someone please confirm that this has to do with the ringcadance of the
indications.conf file? Is there anything else I need to check ?

 Thanks in advance


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Re: [Asterisk-Users] Active ISDN PCMCIA card

[Tan Aks]

We use and sell the AVM B1 PCI V4.0 card. It seems to work well with
asterisk apart from slight echo that I noticed when receiving an isdn --
* -- remote sip phone call.

Tan



- Original Message - 
From: Michael Manousos [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, June 20, 2003 12:28 PM
Subject: [Asterisk-Users] Active ISDN PCMCIA card



Are there any suggestions for active ISDN CAPI PCMCIA cards
that are known to work with Asterisk?

Thanks,
Michael.



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Re: [Asterisk-Users] where to get adsi phones in europe ?

[Tan Aks]

We sell the CE approved versions of the PT390. Contact me offline and I'll
give you details.

Tan

- Original Message - 
From: Thomas Haeger [EMAIL PROTECTED]
To: Asterisk User [EMAIL PROTECTED]
Sent: Friday, June 20, 2003 4:33 PM
Subject: [Asterisk-Users] where to get adsi phones in europe ?


Hi all,

have anybody an idea where to get adsi phones in europe ?



Thanks,

Thomas.

***
beroNet technologies GmbH
Dipl.- Ing. Thomas Häger
Potsdamer Str. 18 A
14513 Teltow

FON:+49 (0) 3328 3077731
FAX:+49 (0) 3328 334779
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