date:20050923

RE: [Asterisk-Users] SNOM 190 '486/Busy here' after upgrade to firmware 3.60s

2005-09-23 Thread Christian Stredicke

Looks like this phone has redirection or DND set. Anything on the
display? If it still a mystery send us the settings of the phone, then
it should become clear.

BTW if you have a snom trouble ticket, you can also go to
http://www.snom.com/onlinesupport.html (scroll down to set up an
account).

CS 

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Colin Anderson
 Sent: Friday, September 23, 2005 1:31 AM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: [Asterisk-Users] SNOM 190 '486/Busy here' after 
 upgrade to firmware 3.60s
 
 I have a Snom 190 that refuses to accept calls after upgrade 
 to firmware 3.60s, latest. I get SIP 486/Busy Here. No 
 change in the dialplan, nor settings in the phone. Calls out 
 fine. I did 30 other phones yesterday with 3.60s with no 
 problem, this is the only one. In the phone's log I get:
 
 [5]22/9/2005 17:01:00: timeout::callback: Registering with 
 timeout of 0 ms
 [5]22/9/2005 17:05:45: Match challenge for user=9002, realm=asterisk
 [2]22/9/2005 17:07:31: Denying call id=-4 
 reason=unconditional --wtf does this mean?
 [5]22/9/2005 17:07:31: Dialog -4/2 going to terminated
 [5]22/9/2005 17:07:31: timeout::callback: Registering with 
 timeout of 0 ms
 
 Any Snom factory guys on the list care to comment?
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Re: [Asterisk-Users] Problems with sipura 1001's and 2002's

2005-09-23 Thread Peter Wemm

On Wednesday 21 September 2005 12:31 pm, Phil Allred wrote:
 I'm having lots of problems with sipura spa1001's and spa2002's. 
 Asterisk claims they are busy when they aren't.  Other times, it
 claims to be ringing them, but they aren't really ringing.  I have
 done the following to try to resolve the problem:

   1)   I upgraded all my spa1001's and 2002's to their latest
 firmware (3.1.5).  This lessened, but did not resolve completely the
 problem.

Make sure you use the dual line firmware for the 1001.  A few months ago 
when I was tearing my hair out with phantom rings, I discovered that I 
had to reflash the firmware to the special dual-line one.  The default 
firmware wouldn't generate an actual ring.

-- 
Peter Wemm - [EMAIL PROTECTED]; [EMAIL PROTECTED]; [EMAIL PROTECTED]
All of this is for nothing if we don't go to the stars - JMS/B5
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Re: [Asterisk-Users] Asterisk + GNUGK + Asterisk-Addons ooh323

2005-09-23 Thread brett

On 9/23/2005, Brian C. Fertig [EMAIL PROTECTED] wrote:

 I am having a slight issue.  I am trying to register 2 asterisk boxes with
 GNUGK and when I try to add the 2nd it gets denied cause of it saying its
 a duplicate.  How do I change the configs to allow more than one asterisk
 box register to the same GK?
 
 brian

Don't 'quote me' on this but...  Look in the h323.conf/s and see if you
have two different h323id strings for the servers.  I think it defaults
to 'Asterisk PBX' but the current sample makes it 'ObjSysAsterisk'. I
am
pretty sure they have to have different names or GNUGK is going to think
they are the same.

Brett
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RE: [Asterisk-Users] custom ring tone

2005-09-23 Thread Marko Rakar

but how to take advantage of this within asterisk?



Two atoms bump into each other. One says 
I think I lost an electron! The other 
asks, Are you sure?, to which the 
first replies, I'm positive.

mailto:[EMAIL PROTECTED]
http://printel.hr 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex
Vishnev
Sent: Friday, September 23, 2005 3:47 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] custom ring tone


Yes, sometime audio is both ways. Sometimes, it is just one way. This
only works in digital network (T1/E1 to T1/E1 (CAS handoff) - the window
is much shorter even in feature group D. or T1/E1 to T1/E1 PRI
signaling where the window could be as large as 90 seconds. Again, that
depends on country, provider, switch software. You can't get this if you
are calling POTS lines.

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[Asterisk-Users] CPU load

2005-09-23 Thread Anders Svensson









Hi!

Here comes a newbi question.

I now that transcoding of codecs take a lot of cpu
load. But if I want to receive all traffic as IAX and then want to send it out
as SIP. Is it the same? Requires a lot of CPU and RAM?







Regards

Anders Svensson








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[Asterisk-Users] Keytouch without effect

2005-09-23 Thread Hans-Peter Straub

Hello anybody,

i have a problem on connecting an innovaphone ip202 to theAsterisk-PBX. When i 
dial in the PBX with the standard (make samples) configuration with the ip202 
the connection is fine, but to push any Key on the keypad dosn't take any 
effect. Is for H323-Phones a special DTMF config necessary?

Thanks

Hans-Peter Straub


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[Asterisk-Users] Hangup when dial via Mobile Interface

2005-09-23 Thread Thawat Mohchit


  
  

  I have Rhio CB24 8FXS/16 FXO which connects to Digium T100P card on [EMAIL PROTECTED] 1.3. There are 2 FXOs of the channel bank connect to the Mobile Interface which the box that insert Mobiles SIM card and it acts like a normal mobile phone. I can dial via these ports but if the destination answers this call it hangs up immediately. The problem does not appear when I use the FXO port of Digium TDM400P.


  How do I fix this problem


  


  Regards


  Thawat

  

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RE: [Asterisk-Users] sipuras 841 bad sound

2005-09-23 Thread Anton Krall

Seems upgrading the firmware fixed the problems with internal calls. 

Im still having problems sith those phones and unicall r2mfc for making
outside calls. 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of 
|Anton Krall
|Sent: Martes, 20 de Septiembre de 2005 06:47 p.m.
|To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
|Subject: [Asterisk-Users] sipuras 841 bad sound
|
|Hi Guys!
|
|I have a problems with some sipuras 841 and asterisk 1.0.9. 
|
|Im using 841 with asterisk 1.0.9 with a digium card (single e1 
|span) with steve's unicall.
|
|Everything compiled fine and in fact I can make and receive 
|calls but I have a problem with bad sound when the sipuras 
|call the outside E1's lines. I can listen to the caller 
|without problems but they heard me with a choppy sound as if 
|you were losing frames or cutting off. Calls between internal 
|sipuras sound good (eventhough the speaker and headset sound 
|comes and goes, for example, when you start talking, seems as 
|if the sipuras takes a few seconds to catch up with you on 
|volume so the remote user listen to you as if the first words 
|and the last were at low volume and the conversation in the 
|middle sound good, any had that problem?)
|
|So, internal calls sound good between 841's but sound volume 
|is weird at the start and end of a sentence. 
|Calling the outside lines via E1's, I can listen to people 
|without problems but they heard me as choppy or cut off.
|
|Anybody had issues like this? Is it asterisk or the phones or what?
|
|Hope you can help Guys, Im really banging my head against the 
|wall here.
|
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Re: [Asterisk-Users] SIP exten to PSTN calls

2005-09-23 Thread Nil S

Hello,

I have read your email.

I found that you have configured X100P card and established a call from SIP exten. to SIP exten and PSTN to SIP exten.

I have done the first part i.e. SIP exten to SIP exten and would like to do a second part. So please help me regarding this.

I have installed Asterisk on Linux machine. So from here please guide me how i should proceed. What are the requirements? and some other details.

Your help will be much appriciated.

Thanks,
Nil.Appan KH [EMAIL PROTECTED] wrote:
Hi,I had configured Asterisk with the following1). X100P - Card2). Two -Greadstream100 SIP Phones.I am able to make calls from SIP Ext to SIP Ext and PSTN calls from outside to SIP Extn.But I am not able to make calls from SIP Extn to PSTN out going calls-it gives BT error message- The number you had dialled not recognised.The SIP extn is not sending the correct number.I will be thank full if some solutions is suggested.appan kh___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [Asterisk-Users] SIP exten to PSTN calls

2005-09-23 Thread Nil S

Hello,

I have read your email.

I found that you have configured X100P card and established a call from SIP exten. to SIP exten and PSTN to SIP exten.

I have done the first part i.e. SIP exten to SIP exten and would like to do a second part. So please help me regarding this.

I have installed Asterisk on Linux machine. So from here please guide me how i should proceed. What are the requirements? and some other details.

Your help will be much appriciated.

Thanks,
Nil.Appan KH [EMAIL PROTECTED] wrote:
Hi,I had configured Asterisk with the following1). X100P - Card2). Two -Greadstream100 SIP Phones.I am able to make calls from SIP Ext to SIP Ext and PSTN calls from outside to SIP Extn.But I am not able to make calls from SIP Extn to PSTN out going calls-it gives BT error message- The number you had dialled not recognised.The SIP extn is not sending the correct number.I will be thank full if some solutions is suggested.appan kh___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users__Do You Yahoo!?Tired of spam? Yahoo! Mail h
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