asterisk users - Oct 2003 - No sound with SIP Phones on the Internet

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Are you using NAT? Is nat=yes in your sip.conf? canreinvite=no,
reinvite=no ?

 

-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Chris Hariga
Sent: Sunday, October 12, 2003 10:42 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] No sound with SIP Phones on the Internet

 

Hi,

 

I need some help with my sip phones. I have a Xten softphone and a Budge
Tone 101 from Grandstream.

If I'm connected from my LAN all is fine but from the Internet I connect
the phone but I don't have the sound.

Asterisk SLI show me this when I try to call my voicemail:

 

localhost*CLI>

    -- Executing VoiceMailMain("SIP/chariga-c067", "105") in
new stack

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

    -- Playing 'vm-password'

  == Spawn extension (internal, 205, 1) exited non-zero on
'SIP/chariga-c067'

    -- Unregistered SIP 'chariga'

localhost*CLI>

 

Any help is welcome.

 

Best regards,

 

Chris Hariga

 

 

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is your SIP phone behind a NAT?  is * behind a NAT?
Uriel
  -----Original Message-----
  From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Chris Hariga
  Sent: Sunday, October 12, 2003 10:42 PM
  To: asterisk-users@lists.digium.com
  Subject: [Asterisk-Users] No sound with SIP Phones on the Internet


  Hi,



  I need some help with my sip phones. I have a Xten softphone and a Budge
Tone 101 from Grandstream.

  If I'm connected from my LAN all is fine but from the Internet I connect
the phone but I don't have the sound.

  Asterisk SLI show me this when I try to call my voicemail:



  localhost*CLI>

      -- Executing VoiceMailMain("SIP/chariga-c067", "105")
in new stack

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

      -- Playing 'vm-password'

    == Spawn extension (internal, 205, 1) exited non-zero on
'SIP/chariga-c067'

      -- Unregistered SIP 'chariga'

  localhost*CLI>



  Any help is welcome.



  Best regards,



  Chris Hariga





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Excellent points in the printed world.  I am not certain that from mail to
eMail I would use the same principles.
Uriel

-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Tilghman
Lesher
Sent: Tuesday, October 14, 2003 6:53 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] [OT] Proper quoting (was: NAT, SIP (was:
No sound with SIP Phones on the Internet))


On Tuesday 14 October 2003 18:15, Uriel Carrasquilla
wrote:
> I have to tell you, at the expense of offending you, that I use
> MS-Outlook and the responses go to the tope of the messages.  At work
> I use Lotus Notes and the same thing happens.  Before, I used PROFS
> (on mainframes) and the same principle applied.  All in all, 20+
> years of using this principle for e-mails at both work and home.  As
> a matter of fact, I am of the opinion that the response to E-mails
> should go at the top to save time.  However, this is not about me but
> the * group and the well being of this list.  Does anybody else have
> a strong opinion one way or the other?  If it is left to John and
> myself we have a 1:1 vote.
This is all you really need to know:

http://learn.to/quote/

-Tilghman

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