administrator tootai
2004-Nov-22 12:29 UTC
[Asterisk-Users] Granstream BT100 - only partial success
George Burt a ?crit :>[...] >host=10.0.0.26 ; we have a static but private IP address > >Here static>[...] > to 10.0.0.26:5060 >Nov 22 17:24:28 NOTICE[3035]: chan_sip.c:7616 handle_request: Registration >from '<sip:grandstream1@10.0.0.127>' failed for '10.0.0.26 > >Here you try to register. Or you put host=dynamic in your sip.conf or you don't ask your BT to register -- Daniel
George Burt
2004-Nov-22 15:43 UTC
[Asterisk-Users] Granstream BT100 - only partial success
We are having many successes with Asterisk and starting to get the hang of
it.
But, I am still having problems getting my Budgetone BT100 (firmware
1.0.4.50) to work fully. I can receive calls, but cannot make them.
We have the latest version of Asterisk, Fedora Core 3, Digium TDM400P with
one FXO and one FXS card configured and working well. We have a PSTN line
going into the Digium card, standard phone going out. Also, we have the
Grandstream phone.
I have included a) extensions.conf, b) sip.conf, c) debug sip console output
and d) the settings for my web-based GS settings.
I also have some comments under the "~~~~~" below the extensions.conf
listed
next.
<--extensions.conf-->
[general]
static=yes
writeprotect=no
autofallthrough=yes
[globals]
CONSOLE=Console/dsp ; Console interface for demo
IAXINFO=guest ; IAXtel username/password
TRUNK=Zap/1 ; Trunk interface
TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
[default]
include => incoming
[incoming]
exten => s,1,Answer()
exten => s,2,NoOp(${CALLERID})
exten => s,3,Dial(SIP/Grandstream1)
exten => 123,1,Answer
exten => 123,2,Dial(SIP/Grandstream1)
exten => 321,1,Answer
exten => 321,2,Dial(Zap/1)
~~~~~~~~~~~~~~~~~~~~~~~~~~~~
When I use the Analog phone to dial "123" The Grandstream1 rings
and
answers and works fine.
But, when I pickup the Grandstream1 handset and dial
<sip.conf>
[general]
port=5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=10.0.0.127 ; IP address to bind to (0.0.0.0 binds to all)
context=default ; Default context for incoming calls
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF.
Default: rfc2833
[grandstream1]
type=friend ; either "friend" (peer+user),
"peer" or "user"
context=incoming
fromuser=grandstream1 ; overrides the callerid, e.g. required by
FWD
username=grandstream1
callerid=John Doe <1234>
host=10.0.0.26 ; we have a static but private IP address
nat=no ; there is not NAT between phone and Asterisk
canreinvite=yes ; allow RTP voice traffic to bypass Asterisk
dtmfmode=rfc2833 ; either RFC2833 or INFO for the BudgeTone
;incominglimit=1 ; permit only 1 outgoing call at a time
; from the phone to asterisk
mailbox=1234@default ; mailbox 1234 in voicemail context
"default"
disallow=all ; need to disallow=all before we can use allowallow=ulaw
; Note: In user sections the order of codecs
; listed with allow= does NOT matter!
allow=alaw
allow=ilbc
allow=g723.1 ; Asterisk only supports g723.1 pass-thru!
allow=g729 ; Pass-thru only unless g729 license obtained
----------------------------
11 headers, 2 lines
Reliably Transmitting:
NOTIFY sip:grandstream1@10.0.0.26 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.127:5060;branch=z9hG4bK36f66772
From: "asterisk" <sip:grandstream1@10.0.0.127>;tag=as56de1e48
To: <sip:grandstream1@10.0.0.26>
Contact: <sip:grandstream1@10.0.0.127>
Call-ID: 5f810538049f22a9183e6e9a742c2626@10.0.0.127
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 42
Messages-Waiting: no
Voice-Message: 0/0
(no NAT) to 10.0.0.26:5060
Scheduling destruction of call
'5f810538049f22a9183e6e9a742c2626@10.0.0.127'
in 15000 ms
Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.127:5060;branch=z9hG4bK36f66772
From: "asterisk" <sip:grandstream1@10.0.0.127>;tag=as56de1e48
To: <sip:grandstream1@10.0.0.26>;tag=a84cd6fa72e72ffe
Call-ID: 5f810538049f22a9183e6e9a742c2626@10.0.0.127
CSeq: 102 NOTIFY
User-Agent: Grandstream BT100 1.0.4.50
Contact: <sip:grandstream1@10.0.0.26>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0
10 headers, 0 lines
Destroying call '5f810538049f22a9183e6e9a742c2626@10.0.0.127'
Destroying call 'cdbc92394c507afc@10.0.0.26'
Sip read:
REGISTER sip:10.0.0.127 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.26;branch=z9hG4bK969a4b237e56781d
From: "George Sip Burt"
<sip:grandstream1@10.0.0.127>;tag=72ca48fb08ef6b70
To: <sip:grandstream1@10.0.0.127>
Contact: *
Call-ID: cdbc92394c507afc@10.0.0.26
CSeq: 102 REGISTER
Expires: 0
User-Agent: Grandstream BT100 1.0.4.50
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0
12 headers, 0 lines
Using latest request as basis request
Sending to 10.0.0.26 : 5060 (non-NAT)
Nov 22 17:24:28 NOTICE[3035]: chan_sip.c:4814 register_verify: Peer
'grandstream1' is trying to register, but not configured as host=dynamic
Transmitting (no NAT):
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 10.0.0.26;branch=z9hG4bK969a4b237e56781d
From: "George Sip Burt"
<sip:grandstream1@10.0.0.127>;tag=72ca48fb08ef6b70
To: <sip:grandstream1@10.0.0.127>;tag=as5bb79d45
Call-ID: cdbc92394c507afc@10.0.0.26
CSeq: 102 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:grandstream1@10.0.0.127>
Content-Length: 0
to 10.0.0.26:5060
Nov 22 17:24:28 NOTICE[3035]: chan_sip.c:7616 handle_request: Registration
from '<sip:grandstream1@10.0.0.127>' failed for
'10.0.0.26'
Scheduling destruction of call 'cdbc92394c507afc@10.0.0.26' in 15000 ms
Destroying call 'cdbc92394c507afc@10.0.0.26'
Sip read:
REGISTER sip:10.0.0.127 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.26;branch=z9hG4bK2ba7e96c0bbdf023
From: "George Sip Burt"
<sip:grandstream1@10.0.0.127>;tag=72ca48fb08ef6b70
To: <sip:grandstream1@10.0.0.127>
Contact: *
Call-ID: cdbc92394c507afc@10.0.0.26
CSeq: 103 REGISTER
Expires: 0
User-Agent: Grandstream BT100 1.0.4.50
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0
12 headers, 0 lines
Using latest request as basis request
Sending to 10.0.0.26 : 5060 (non-NAT)
Nov 22 17:24:45 NOTICE[3035]: chan_sip.c:4814 register_verify: Peer
'grandstream1' is trying to register, but not configured as host=dynamic
Transmitting (no NAT):
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 10.0.0.26;branch=z9hG4bK2ba7e96c0bbdf023
From: "George Sip Burt"
<sip:grandstream1@10.0.0.127>;tag=72ca48fb08ef6b70
To: <sip:grandstream1@10.0.0.127>;tag=as0c7ffa5b
Call-ID: cdbc92394c507afc@10.0.0.26
CSeq: 103 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:grandstream1@10.0.0.127>
Content-Length: 0
to 10.0.0.26:5060
Nov 22 17:24:45 NOTICE[3035]: chan_sip.c:7616 handle_request: Registration
from '<sip:grandstream1@10.0.0.127>' failed for
'10.0.0.26'
Scheduling destruction of call 'cdbc92394c507afc@10.0.0.26' in 15000 ms
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
This "schedule destruction of call" happens over and over, even though
I
have not done anything except load asterisk and turn on "sip debug"
and
power cycle the BT100.
Next, I pickup the BT100 handset and dial 321. The analog phone clicks as
if it is starting to ring for an instant, then the BT100 plays a busy
signal. I pickup the Analog handset and get a dial tone.
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
-- Added extension '123' priority 1 to incoming
-- Added extension '123' priority 2 to incoming
-- Added extension '321' priority 1 to incoming
-- Added extension '321' priority 2 to incoming
-- Reloading module 'pbx_dundi.so' (Distributed Universal Number
Discovery (DUNDi))
== Parsing '/etc/asterisk/dundi.conf': Found
-- Reloading module 'app_voicemail.so' (Comedian Mail (Voicemail
System))
== Parsing '/etc/asterisk/voicemail.conf': Found
-- Reloading module 'cdr_csv.so' (Comma Separated Values CDR
Backend)
-- Reloading module 'app_txtcidname.so' (TXTCIDName)
== Parsing '/etc/asterisk/enum.conf': Found
-- Reloading module 'app_enumlookup.so' (ENUM Lookup)
== Parsing '/etc/asterisk/enum.conf': Found
-- Reloading module 'app_queue.so' (True Call Queueing)
== Parsing '/etc/asterisk/queues.conf': Found
-- Reloading module 'cdr_manager.so' (Asterisk Call Manager CDR
Backend)
== Parsing '/etc/asterisk/cdr_manager.conf': Found
11 headers, 2 lines
Reliably Transmitting:
NOTIFY sip:grandstream1@10.0.0.26 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.127:5060;branch=z9hG4bK13682bc7
From: "asterisk" <sip:grandstream1@10.0.0.127>;tag=as5e95b156
To: <sip:grandstream1@10.0.0.26>
Contact: <sip:grandstream1@10.0.0.127>
Call-ID: 0c14e1ab771463f11513857675be17e2@10.0.0.127
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 42
Messages-Waiting: no
Voice-Message: 0/0
(no NAT) to 10.0.0.26:5060
Scheduling destruction of call
'0c14e1ab771463f11513857675be17e2@10.0.0.127'
in 15000 ms
Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.127:5060;branch=z9hG4bK13682bc7
From: "asterisk" <sip:grandstream1@10.0.0.127>;tag=as5e95b156
To: <sip:grandstream1@10.0.0.26>;tag=08d9fc45d3739255
Call-ID: 0c14e1ab771463f11513857675be17e2@10.0.0.127
CSeq: 102 NOTIFY
User-Agent: Grandstream BT100 1.0.4.50
Contact: <sip:grandstream1@10.0.0.26>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0
10 headers, 0 lines
Destroying call '0c14e1ab771463f11513857675be17e2@10.0.0.127'
Sip read:
INVITE sip:321@10.0.0.127 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.26;branch=z9hG4bK9c4ce26a781a4b4f
From: "George Sip Burt"
<sip:grandstream1@10.0.0.127>;tag=23de801d5cc2edfd
To: <sip:321@10.0.0.127>
Contact: <sip:grandstream1@10.0.0.26>
Call-ID: 75a24ffcd23bc78e@10.0.0.26
CSeq: 26512 INVITE
User-Agent: Grandstream BT100 1.0.4.50
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Type: application/sdp
Content-Length: 169
v=0
o=grandstream1 8000 8000 IN IP4 10.0.0.26
s=SIP Call
c=IN IP4 10.0.0.26
t=0 0
m=audio 5004 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
12 headers, 9 lines
Using latest request as basis request
Sending to 10.0.0.26 : 5060 (non-NAT)
Found RTP audio format 0
Found RTP audio format 8
Peer audio RTP is at port 10.0.0.26:5004
Found description format PCMU
Found description format PCMA
Capabilities: us - 0x8000e(GSM|ULAW|ALAW|H263), peer -
audio=0xc(ULAW|ALAW)/video=0x0(EMPTY), combined - 0xc(ULAW|ALAW)
Non-codec capabilities: us - 0x1(G723), peer - 0x0(EMPTY), combined -
0x0(EMPTY)
Found user 'grandstream1'
Looking for 321 in incoming
list_route: hop: <sip:grandstream1@10.0.0.26>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.0.26;branch=z9hG4bK9c4ce26a781a4b4f
From: "George Sip Burt"
<sip:grandstream1@10.0.0.127>;tag=23de801d5cc2edfd
To: <sip:321@10.0.0.127>;tag=as2e55e96c
Call-ID: 75a24ffcd23bc78e@10.0.0.26
CSeq: 26512 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:321@10.0.0.127>
Content-Length: 0
to 10.0.0.26:5060
-- Executing Answer("SIP/grandstream1-3f43", "") in new
stack
We're at 10.0.0.127 port 11680
Answering with capability 0x1(G723)
Answering with capability 0x4(ULAW)
Answering with capability 0x8(ALAW)
Answering with capability 0x100(G729A)
Answering with capability 0x400(ILBC)
Answering with non-codec capability 0x1(G723)
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.26;branch=z9hG4bK9c4ce26a781a4b4f
From: "George Sip Burt"
<sip:grandstream1@10.0.0.127>;tag=23de801d5cc2edfd
To: <sip:321@10.0.0.127>;tag=as2e55e96c
Call-ID: 75a24ffcd23bc78e@10.0.0.26
CSeq: 26512 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:321@10.0.0.127>
Content-Type: application/sdp
Content-Length: 310
v=0
o=root 3035 3035 IN IP4 10.0.0.127
s=session
c=IN IP4 10.0.0.127
t=0 0
m=audio 11680 RTP/AVP 4 0 8 18 97 101
a=rtpmap:4 G723/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
to 10.0.0.26:5060
-- Executing Dial("SIP/grandstream1-3f43", "Zap/1") in
new stack
-- Called 1
-- Zap/1-1 is ringing
Sip read:
ACK sip:321@10.0.0.127 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.26;branch=z9hG4bK9c4ce26a781a4b4f
From: "George Sip Burt"
<sip:grandstream1@10.0.0.127>;tag=23de801d5cc2edfd
To: <sip:321@10.0.0.127>;tag=as2e55e96c
Contact: <sip:grandstream1@10.0.0.26>
Call-ID: 75a24ffcd23bc78e@10.0.0.26
CSeq: 26512 ACK
User-Agent: Grandstream BT100 1.0.4.50
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0
11 headers, 0 lines
Nov 22 17:32:57 NOTICE[3035]: channel.c:1731 ast_set_read_format: Unable to
find a path from G723 to ULAW
Nov 22 17:32:57 NOTICE[3035]: channel.c:1698 ast_set_write_format: Unable to
find a path from SLINR to G723
Nov 22 17:32:57 WARNING[3035]: chan_sip.c:1831 sip_write: Asked to transmit
frame type 4, while native formats is 1 (read/write = 4/64)
Nov 22 17:32:57 WARNING[3035]: chan_zap.c:4239 zt_write: Cannot handle
frames in 1 format
Nov 22 17:32:57 WARNING[3035]: app_dial.c:424 wait_for_answer: Unable to
forward voice
-- Hungup 'Zap/1-1'
== No one is available to answer at this time
Nov 22 17:32:57 NOTICE[3035]: channel.c:1698 ast_set_write_format: Unable to
find a path from ULAW to G723
== Auto fallthrough, channel 'SIP/grandstream1-3f43' status is
'NOANSWER'
set_destination: Parsing <sip:grandstream1@10.0.0.26> for address/port to
send to
set_destination: set destination to 10.0.0.26, port 5060
Reliably Transmitting:
BYE sip:grandstream1@10.0.0.26 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.127:5060;branch=z9hG4bK5163f585;rport
From: <sip:321@10.0.0.127>;tag=as2e55e96c
To: "George Sip Burt"
<sip:grandstream1@10.0.0.127>;tag=23de801d5cc2edfd
Contact: <sip:321@10.0.0.127>
Call-ID: 75a24ffcd23bc78e@10.0.0.26
CSeq: 102 BYE
User-Agent: Asterisk PBX
Content-Length: 0
(no NAT) to 10.0.0.26:5060
Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.127:5060;branch=z9hG4bK5163f585;rport
From: <sip:321@10.0.0.127>;tag=as2e55e96c
To: "George Sip Burt"
<sip:grandstream1@10.0.0.127>;tag=23de801d5cc2edfd
Call-ID: 75a24ffcd23bc78e@10.0.0.26
CSeq: 102 BYE
User-Agent: Grandstream BT100 1.0.4.50
Contact: <sip:grandstream1@10.0.0.26>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0
10 headers, 0 lines
Message is BYE
Destroying call '75a24ffcd23bc78e@10.0.0.26'
Destroying call 'cdbc92394c507afc@10.0.0.26'
sip
Sip read:
REGISTER sip:10.0.0.127 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.26;branch=z9hG4bK66e067077adc074a
From: "George Sip Burt"
<sip:grandstream1@10.0.0.127>;tag=72ca48fb08ef6b70
To: <sip:grandstream1@10.0.0.127>
Contact: *
Call-ID: cdbc92394c507afc@10.0.0.26
CSeq: 133 REGISTER
Expires: 0
User-Agent: Grandstream BT100 1.0.4.50
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0
12 headers, 0 lines
Using latest request as basis request
Sending to 10.0.0.26 : 5060 (non-NAT)
Nov 22 17:33:01 NOTICE[3035]: chan_sip.c:4814 register_verify: Peer
'grandstream1' is trying to register, but not configured as host=dynamic
Transmitting (no NAT):
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 10.0.0.26;branch=z9hG4bK66e067077adc074a
From: "George Sip Burt"
<sip:grandstream1@10.0.0.127>;tag=72ca48fb08ef6b70
To: <sip:grandstream1@10.0.0.127>;tag=as2d358f61
Call-ID: cdbc92394c507afc@10.0.0.26
CSeq: 133 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:grandstream1@10.0.0.127>
Content-Length: 0
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
MAC Address: 00.0B.82.00.33.1D
Product Model: BT100
Software Version: Program--1.0.4.50 Bootloader--1.0.0.14
HTML--1.0.0.22
Admin Password: ***** (password to configure this IP phone)
IP Address:
statically configured as:
IP Address: 10.0.0.26
Subnet Mask: 255.255.255.0
Default Router: 10.0.0.1
DNS Server 1: 166.102.165blah
DNS Server 2: blah blah
SIP Server: 10.0.0.127 (e.g., sip.mycompany.com, or IP address)
Outbound Proxy: 10.0.0.127 (e.g., proxy.myprovider.com, or IP address,
if any)
SIP User ID: grandstream1 (the user part of an SIP address)
Authenticate ID: grandstream1 (can be identical to or different from SIP
User ID)
Authenticate Password: (none)
Name: George Sip Burt
Advanced Options:
Preferred Vocoder:
(in listed order) choice 1: PCMU
choice 2: PCMA
choice 3: PCMU
choice 4: PCMU
choice 5: PCMU
choice 6: PCMU
G723 rate: (selected) 6.3kbps encoding rate 5.3kbps encoding rate
Silence Suppression: No
Voice Frames per TX: 2 (up to 10/20/32/64 for G711/G726/G723/other codecs
respectively)
Layer 3 QoS: 48 (Diff-Serv or Precedence value)
Layer 2 QoS: 0 802.1Q/VLAN Tag 0 802.1p priority value (0-7)
User ID is phone number: No
Dial Plan: (empty) (dial plan prefix string)
SIP Registration: Yes
Unregister On Reboot: Yes
Register Expiration: 60 (in minutes. default 1 hour, max 45 days)
Early Dial: No (use "Yes" only if proxy supports 484 response)
Use # as Dial Key: No Yes (if set to Yes, "#" will function as
the
"(Re-)Dial" key)
local SIP port: 5060 (default 5060)
local RTP port: 5004 (1024-65535, default 5004)
Use random port: No
NAT Traversal: No
keep-alive interval: 30 (in seconds, default 20 seconds)
TFTP Server: 168.75.215.189 (for remote software upgrade and
configuration)
Voice Mail UserID: (User ID/extension for 3rd party voice mail system)
Auto Answer: No
Offhook Auto-Dial: empty (User ID/extension to dial automatically when
offhook)
Send DTMF: via SIP INFO
DTMF Payload Type: 101
Send Flash Event: No
NTP Server: time.nist.gov (URI or IP address)
Time Zone: GMT-5
Daylight Savings Time: No
Send Anonymous: No
Lock Menu button: No
~~~~~~~~~~~~~~~~~~~~~~~~~~~~
Any help would be appreciated and I will surely pass the favor on to those
that come behind me.
Also, if there is any reference material on how to read the debug screens,
that would be good to know. I couldn't find any.
George
Stephen R. Besch
2004-Nov-23 07:55 UTC
[Asterisk-Users] Re: Granstream BT100 - only partial success
George Burt wrote: <snip>> [grandstream1] > host=10.0.0.26 ; we have a static but private IP address > canreinvite=yes ; allow RTP voice traffic to bypass Asterisk<snip>> IP Address: > statically configured as: > IP Address: 10.0.0.26 > Subnet Mask: 255.255.255.0 > Default Router: 10.0.0.1 > SIP Registration: YesComments: 1) You can't ask asterisk to register your phone if you have a fixed IP address specified as host= in sip.conf. Either the phone sends the address (i.e., host=dynamic), or you enter it as an IP address. It's OK to be fixed at the phone and dynamic in asterisk, but that isn't rational - just adds net traffic. Turn off the sip registration option on the phone. 2) Unless I am mistaken, you are not going to be able to use re-invites without NAT. It will work on your calls to analog phones handled by Asterisk and to other IP phones on the local network. However, as soon as you connect to an outbound/inbound service, the reinvite will fail and you will lose your media stream. 3) Don't know if it will make a difference, but I always set the router field to 0.0.0.0. There is no such thing as a valid router IP on a private network - they are not routable by design. I had quite an argument with Grandstream about this when I first purchased the phones. As a result, the firmware was modified to accept a null router entry for use with private IP ranges. Sincerely, Stephen R. Besch