RE: [Asterisk-Users] Speaking of ring tones...

2004-03-19 Thread Kevin Pearcey
Has anyone figured out how to change the volume of the ring tone on the snom
200?

Its pretty loud in our office when a group of these things start to ring
together.

Kev

 I kinda like it .. ;) 
 Nice  conservative.
 OTOH, the new snom 200 I just got today has some reeeaaally 
 weird ring tones (and nothing really 'traditional'). Now, 
 maybe we should take a lesson from the cell-phone people, and 
 talk manufacturers into letting us download ringtone(s). Cheers, 
 WW 

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RE: [Asterisk-Users] SIP behind NAT - use of externip option

2004-01-29 Thread Kevin Pearcey
I've just recently got this working with asterisk 0.7.1.

You also need to add values for:
Localnet  localmask to indicate the values that do not get externip set.

However, this did not fully solve the problem for me. External phones would
register just fine, but then I kept getting lots of retry failures. It
appears that asterisk changes the Call-ID value between what is sent and
what is stored internally!

Once a phone registers I sip debug shows asterisk sending a NOTIFY message
wit a Call-ID value using the externip address, however the 200 OK response
back from the phone has a call id with the same random characters but has
the localip address instead.

I've currently solved this but removing the ip address from build_callid()
until I try to see where asterisk gets this wrong. Has anyone else seen
this?

Having made this fix I now have SIP going NAT - NAT - asterisk.

Kevin

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Patrick Lidstone (Personal E-mail)
 Sent: 26 January 2004 18:31
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] SIP behind NAT - use of externip option
 
 
 I am having difficulty configuring SIP behind NAT (using latest CVS).
 
 Using sip.conf:
 
 [general]
 port=5060   ; Port to bind to
 externip=ww.xx.yy.zz
 bindaddr=0.0.0.0
 nat=yes
 register=[userid]:[EMAIL PROTECTED]/2000
 [voiptalk.org]
 nat=yes
 externip=ww.xx.yy.zz
 type=friend
 secret=[password]
 nat=yes
 reinvite=no
 canreinvite=no
 
 I fail to register. SIP Debug gives:
 
 SIP Debugging Enabled
 Jan 26 18:20:04 NOTICE[9226]: chan_sip.c:3126 
 sip_reg_timeout: Registration for  '[EMAIL PROTECTED]' 
 timed out, trying again 11 headers, 0 lines  Reliably 
 Transmitting: REGISTER sip:voiptalk.org SIP/2.0
 Via: SIP/2.0/UDP 192.168.0.89:5060;branch=z9hG4bK02c0373f
 From: sip:[EMAIL PROTECTED];tag=as5548d275
 To: sip:[EMAIL PROTECTED]
 Call-ID: [EMAIL PROTECTED]
 CSeq: 103 REGISTER
 User-Agent: Asterisk PBX
 Expires: 120
 Contact: sip:[EMAIL PROTECTED]
 Event: registration
 Content-length: 0
 
  (no NAT) to 82.145.32.73:5060
 Retransmitting #1 (no NAT):
 REGISTER sip:voiptalk.org SIP/2.0
 Via: SIP/2.0/UDP 192.168.0.89:5060;branch=z9hG4bK02c0373f
 From: sip:[EMAIL PROTECTED];tag=as5548d275
 To: sip:[EMAIL PROTECTED]
 Call-ID: [EMAIL PROTECTED]
 CSeq: 103 REGISTER
 User-Agent: Asterisk PBX
 Expires: 120
 Contact: sip:[EMAIL PROTECTED]
 Event: registration
 Content-length: 0
 
 
  to 82.145.32.73:5060
 Retransmitting #2 (no NAT):
 ... as previous frame ...
 
 I would expect (perhaps wrongly?) that the externip would be 
 reflected in the Via: header (it seems to be when I look at 
 the traces from a SNOM 100 successfully registery with 
 voiptalk behind the same firewall). And the (no NAT) comments 
 in the debug trace also look suspicious, given the explicit 
 nat=yes in the config.
 
 Any hints? I guess the externip might be in the wrong place 
 or conflicting with one of my other options, but neither the 
 docs don't seem to offer much by way of advice (I've checked 
 the wiki, googled etc). Thanks Patrick
 
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