A previous poster mentioned the same thing, with no response:
http://lists.digium.com/pipermail/asterisk-users/2004-
December/080161.html
Fresh asterisk 1.0.5 install on FC3, started with make samples,
nothing fancy. It's so bland, I'm surprised the list isn't full of
people having the same trouble.
I have several Uniden UIP200 phones and a single Grandstream BudgetTone
100. Any combination does the same thing.
Calls started from within asterisk (*.call files, transfers, directory)
work fine. I've tried all combinations of codecs, with no change.
This is my first serious attempt with *, so don't be afraid to assume
I'm a moron.
Relevent config snippets and a set verbose 100 and SIP DEBUG console
dump follow.
*** sip.conf ***
[general]
context=default
port=5060
bindaddr=0.0.0.0
srvlookup=yes
[1010]
type=friend
host=dynamic
username=1010
secret=password
context=default
dtmfmode=rfc2833
1011-1019 are all basically the same as 1010
*** extensions.conf ***
[default]
exten = 1010,1,Dial(SIP/1010,20,tr)
exten = 1011,1,Dial(SIP/1011,20,tr)
etc
*** console dump of call, hold, unhold, hangup ***
*** Asterisk on 192.168.200.0, phones on 192.168.201.0,
*** connected by VPN, same thing happens when on one lan
Sip read:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.201.111;branch=z9hG4bK1ae8b35c5d8d1ae5
From: sip:[EMAIL PROTECTED];tag=9970b15421c8f59c
To: sip:[EMAIL PROTECTED]
Contact: sip:[EMAIL PROTECTED]
Supported: replaces
Call-ID: [EMAIL PROTECTED]
CSeq: 22567 INVITE
User-Agent: Grandstream BT100 1.0.5.16
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Type: application/sdp
Content-Length: 354
v=0
=1019 0 8000 IN IP4 192.168.201.111
s=SIP Call
c=IN IP4 192.168.201.111
t=0 0
m=audio 5004 RTP/AVP 0 8 4 18 2 15 99 9
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:15 G728/8000
a=rtpmap:99 iLBC/8000
a=fmtp:99 mode=20
a=rtpmap:9 G722/8000
a=ptime:20
13 headers, 17 lines
Using latest request as basis request
Sending to 192.168.201.111 : 5060 (non-NAT)
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 4
Found RTP audio format 18
Found RTP audio format 2
Found RTP audio format 15
Found RTP audio format 99
Found RTP audio format 9
Peer audio RTP is at port 192.168.201.111:5004
Found description format PCMU
Found description format PCMA
Found description format G723
Found description format G729
Found description format G726-32
Found description format G728
Found description format iLBC
Found description format G722
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x51d
(g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0xc
(ulaw|alaw)
Non-codec capabilities: us - 0x1 (g723), peer - 0x0 (nothing), combined
- 0x0 (nothing)
Reliably Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.201.111;branch=z9hG4bK1ae8b35c5d8d1ae5
From: sip:[EMAIL PROTECTED];tag=9970b15421c8f59c
To: sip:[EMAIL PROTECTED];tag=as45319780
Call-ID: [EMAIL PROTECTED]
CSeq: 22567 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:[EMAIL PROTECTED]
Proxy-Authenticate: Digest realm=asterisk, nonce=499f7907
Content-Length: 0
to 192.168.201.111:5060
Scheduling destruction of call '[EMAIL PROTECTED]' in
15000 ms
Found user '1019'
asterisk*CLI
Sip read:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.201.111;branch=z9hG4bK1ae8b35c5d8d1ae5
From: sip:[EMAIL PROTECTED];tag=9970b15421c8f59c
To: sip:[EMAIL PROTECTED];tag=as45319780
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 22567 ACK
User-Agent: Grandstream BT100 1.0.5.16
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0
11 headers, 0 lines
asterisk*CLI
Sip read:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.201.111;branch=z9hG4bK1d2ba72e99353828
From: sip:[EMAIL PROTECTED];tag=9970b15421c8f59c
To: sip:[EMAIL PROTECTED]
Contact: sip:[EMAIL PROTECTED]
Supported: replaces
Proxy-Authorization: DIGEST username=1019, realm=asterisk,
algorithm=MD5, uri=sip:[EMAIL PROTECTED], nonce=499f7907,
response=80ba81f6c2dc429b45c8bb6d57c9b7d6
Call-ID: [EMAIL PROTECTED]
CSeq: 22568 INVITE
User-Agent: Grandstream BT100 1.0.5.16
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Type: application/sdp
Content-Length: 354
v=0
o=1019 1 8000 IN IP4 192.168.201.111
s=SIP Call
c=IN IP4 192.168.201.111
t=0 0
m=audio 5004 RTP/AVP 0 8 4 18 2 15 99 9
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:15 G728/8000
a=rtpmap:99 iLBC/8000
a=fmtp:99 mode=20
a=rtpmap:9 G722/8000
a=ptime:20
14 headers, 17 lines
Using latest request as basis request
Sending to 192.168.201.111 : 5060 (non-NAT)
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio
