[Asterisk-Users] E100P Leds

2004-04-13 Thread Jon Shamash
Hi,

Can anyone please tell me what the leds (and colours ) mean on an E110P

Thanks Jon

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[Asterisk-Users] NTL PRI Config required please

2004-04-08 Thread Jon Shamash
Hi,

I've today received an E100P.

Can anyone help me out with the config of this card please. We have an
NTL PRI circuit installed with 10 channels active.


Also what do the LEDs on the back of the card mean ?


Many thanks

Jon

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Re: [Asterisk-Users] 2 Linphones communicating through Asterisk?

2004-03-04 Thread Jon Shamash
Hi...

Being very new to A* myself I understand your fustrations with the manuals
:)

It looks like you've made a typo in your extensions.conf

quote [sip]
extern = 66,1,Dial(SIP/66)
extern = 61,1,Dial(SIP/61)
extern = 60,1,Dial(SIP/60)


it should be
exten = 66,1,Dial(SIP/66)

Hope that helps

Jnn

- Original Message - 
From: Thomas Sparr [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, March 04, 2004 1:46 PM
Subject: [Asterisk-Users] 2 Linphones communicating through Asterisk?


 Hi all,

 I'm experimenting with the following setup:
 An Asterisk server at 192.168.0.10.
 2 Linphones at 192.168.0.60 and 192.168.0.66.
 The Linphones register themselves at the Asterisk as sip:[EMAIL PROTECTED]
 and sip:[EMAIL PROTECTED]
 If my understanding is correct they should be available on the Asterisk
 as sip:[EMAIL PROTECTED] and sip:[EMAIL PROTECTED]
 However, if I try to call sip:[EMAIL PROTECTED] from the other Linphone
 the Asterisk debug says:

 Looking for 66 in sip
 Transmitting (no NAT):
 SIP/2.0 404 Not Found

 Are there any merciful soul on this list who can point me in the rigth
 direction?
 If your answer are RTFM, please tell me which FM to R.
 Asterisk sip debug follow below.
 Also attaching config files for Asterisk and Linphone I have messed
 with. All others are from make samples in asterisk.
 versions: Linphone 0.12.1, libosip 0.9.7, asterisk 2 days old from cvs.

 Regards

 Thomas


 Sip read:
 REGISTER sip:192.168.0.10 SIP/2.0
 Via: SIP/2.0/UDP 192.168.0.60:5060;branch=z9hG4bK3085308593
 From: sip:[EMAIL PROTECTED];tag=680816676
 To: sip:[EMAIL PROTECTED];tag=680816676
 Call-ID: [EMAIL PROTECTED]
 CSeq: 0 REGISTER
 Contact: sip:[EMAIL PROTECTED]
 max-forwards: 10
 expires: 3600
 user-agent: oSIP/Linphone-0.12.1
 Content-Length: 0


 11 headers, 0 lines
 Using latest request as basis request
 Sending to 192.168.0.60 : 5060 (non-NAT)
 Transmitting (no NAT):
 SIP/2.0 100 Trying
 Via: SIP/2.0/UDP 192.168.0.60:5060;branch=z9hG4bK3085308593
 From: sip:[EMAIL PROTECTED];tag=680816676
 To: sip:[EMAIL PROTECTED];tag=680816676
 Call-ID: [EMAIL PROTECTED]
 CSeq: 0 REGISTER
 User-Agent: Asterisk PBX
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
 Contact: sip:[EMAIL PROTECTED]
 Content-Length: 0


  to 192.168.0.60:5060
 -- Registered SIP '60' at 192.168.0.60 port 5060 expires 3600
 Transmitting (no NAT):
 SIP/2.0 200 OK
 Via: SIP/2.0/UDP 192.168.0.60:5060;branch=z9hG4bK3085308593
 From: sip:[EMAIL PROTECTED];tag=680816676
 To: sip:[EMAIL PROTECTED];tag=680816676
 Call-ID: [EMAIL PROTECTED]
 CSeq: 0 REGISTER
 User-Agent: Asterisk PBX
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
 Expires: 3600
 Contact: sip:[EMAIL PROTECTED];expires=3600
 Date: Thu, 04 Mar 2004 13:45:14 GMT
 Content-Length: 0

 Sip read:
 INVITE sip:[EMAIL PROTECTED] SIP/2.0
 Via: SIP/2.0/UDP 192.168.0.60:5060;branch=z9hG4bK962844352
 From: sip:[EMAIL PROTECTED];tag=4243659372;tag=4075507534
 To:  sip:[EMAIL PROTECTED]
 Call-ID: [EMAIL PROTECTED]
 CSeq: 20 INVITE
 Contact: sip:[EMAIL PROTECTED]
 max-forwards: 10
 user-agent: oSIP/Linphone-0.12.1
 Content-Type: application/sdp
 Content-Length:   367

 v=0
 o=60 123456 654321 IN IP4 192.168.0.60
 s=A conversation
 c=IN IP4 192.168.0.60
 t=0 0
 m=audio 7078 RTP/AVP 0 8 3 110 111 115 101
 b=AS:110 20
 b=AS:111 28
 a=rtpmap:0 PCMU/8000/1
 a=rtpmap:8 PCMA/8000/1
 a=rtpmap:3 GSM/8000/1
 a=rtpmap:110 speex/8000/1
 a=rtpmap:111 speex/16000/1
 a=rtpmap:115 1015/8000/1
 a=rtpmap:101 telephone-event/8000
 a=fmtp:101 0-11

 11 headers, 16 lines
 Using latest request as basis request
 Sending to 192.168.0.60 : 5060 (non-NAT)
 Found audio format UNKN
 Found audio format ALAW
 Found audio format UNKN
 Found audio format UNKN
 Found audio format UNKN
 Found audio format UNKN
 Found audio format UNKN
 Found description format PCMU
 Found description format PCMA
 Found description format GSM
 Found description format speex
 Found description format speex
 Found description format 1015
 Found description format telephone-event
 Capabilities: us - 12, them - 526/0, combined - 12
 Non-codec capabilities: us - 1, them - 1, combined - 1
 Looking for 66 in sip
 Transmitting (no NAT):
 SIP/2.0 404 Not Found
 Via: SIP/2.0/UDP 192.168.0.60:5060;branch=z9hG4bK962844352
 From: sip:[EMAIL PROTECTED];tag=4243659372;tag=4075507534
 To: sip:[EMAIL PROTECTED];tag=as7e281fb9
 Call-ID: [EMAIL PROTECTED]
 CSeq: 20 INVITE
 User-Agent: Asterisk PBX
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
 Contact: sip:[EMAIL PROTECTED]
 Content-Length: 0


  to 192.168.0.60:5060


 Sip read:
 ACK sip:[EMAIL PROTECTED] SIP/2.0
 Via: SIP/2.0/UDP 192.168.0.60:5060;branch=z9hG4bK962844352
 From: sip:[EMAIL PROTECTED];tag=4243659372;tag=4075507534
 To: sip:[EMAIL PROTECTED];tag=as7e281fb9
 Call-ID: [EMAIL PROTECTED]
 CSeq: 20 ACK
 Content-Length: 0





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