[Asterisk-Users] Granstream BT100 - only partial success
George Burt
george at trueshot.com
Mon Nov 22 15:43:08 MST 2004
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 at default ; mailbox 1234 in voicemail context "default"
disallow=all ; need to disallow=all before we can use allow=
allow=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 at 10.0.0.26 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.127:5060;branch=z9hG4bK36f66772
From: "asterisk" <sip:grandstream1 at 10.0.0.127>;tag=as56de1e48
To: <sip:grandstream1 at 10.0.0.26>
Contact: <sip:grandstream1 at 10.0.0.127>
Call-ID: 5f810538049f22a9183e6e9a742c2626 at 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 at 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 at 10.0.0.127>;tag=as56de1e48
To: <sip:grandstream1 at 10.0.0.26>;tag=a84cd6fa72e72ffe
Call-ID: 5f810538049f22a9183e6e9a742c2626 at 10.0.0.127
CSeq: 102 NOTIFY
User-Agent: Grandstream BT100 1.0.4.50
Contact: <sip:grandstream1 at 10.0.0.26>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0
10 headers, 0 lines
Destroying call '5f810538049f22a9183e6e9a742c2626 at 10.0.0.127'
Destroying call 'cdbc92394c507afc at 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 at 10.0.0.127>;tag=72ca48fb08ef6b70
To: <sip:grandstream1 at 10.0.0.127>
Contact: *
Call-ID: cdbc92394c507afc at 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 at 10.0.0.127>;tag=72ca48fb08ef6b70
To: <sip:grandstream1 at 10.0.0.127>;tag=as5bb79d45
Call-ID: cdbc92394c507afc at 10.0.0.26
CSeq: 102 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:grandstream1 at 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 at 10.0.0.127>' failed for '10.0.0.26'
Scheduling destruction of call 'cdbc92394c507afc at 10.0.0.26' in 15000 ms
Destroying call 'cdbc92394c507afc at 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 at 10.0.0.127>;tag=72ca48fb08ef6b70
To: <sip:grandstream1 at 10.0.0.127>
Contact: *
Call-ID: cdbc92394c507afc at 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 at 10.0.0.127>;tag=72ca48fb08ef6b70
To: <sip:grandstream1 at 10.0.0.127>;tag=as0c7ffa5b
Call-ID: cdbc92394c507afc at 10.0.0.26
CSeq: 103 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:grandstream1 at 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 at 10.0.0.127>' failed for '10.0.0.26'
Scheduling destruction of call 'cdbc92394c507afc at 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 at 10.0.0.26 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.127:5060;branch=z9hG4bK13682bc7
From: "asterisk" <sip:grandstream1 at 10.0.0.127>;tag=as5e95b156
To: <sip:grandstream1 at 10.0.0.26>
Contact: <sip:grandstream1 at 10.0.0.127>
Call-ID: 0c14e1ab771463f11513857675be17e2 at 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 at 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 at 10.0.0.127>;tag=as5e95b156
To: <sip:grandstream1 at 10.0.0.26>;tag=08d9fc45d3739255
Call-ID: 0c14e1ab771463f11513857675be17e2 at 10.0.0.127
CSeq: 102 NOTIFY
User-Agent: Grandstream BT100 1.0.4.50
Contact: <sip:grandstream1 at 10.0.0.26>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0
10 headers, 0 lines
Destroying call '0c14e1ab771463f11513857675be17e2 at 10.0.0.127'
Sip read:
INVITE sip:321 at 10.0.0.127 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.26;branch=z9hG4bK9c4ce26a781a4b4f
From: "George Sip Burt" <sip:grandstream1 at 10.0.0.127>;tag=23de801d5cc2edfd
To: <sip:321 at 10.0.0.127>
Contact: <sip:grandstream1 at 10.0.0.26>
Call-ID: 75a24ffcd23bc78e at 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 at 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 at 10.0.0.127>;tag=23de801d5cc2edfd
To: <sip:321 at 10.0.0.127>;tag=as2e55e96c
Call-ID: 75a24ffcd23bc78e at 10.0.0.26
CSeq: 26512 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:321 at 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 at 10.0.0.127>;tag=23de801d5cc2edfd
To: <sip:321 at 10.0.0.127>;tag=as2e55e96c
Call-ID: 75a24ffcd23bc78e at 10.0.0.26
CSeq: 26512 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:321 at 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 at 10.0.0.127 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.26;branch=z9hG4bK9c4ce26a781a4b4f
From: "George Sip Burt" <sip:grandstream1 at 10.0.0.127>;tag=23de801d5cc2edfd
To: <sip:321 at 10.0.0.127>;tag=as2e55e96c
Contact: <sip:grandstream1 at 10.0.0.26>
Call-ID: 75a24ffcd23bc78e at 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 at 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 at 10.0.0.26 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.127:5060;branch=z9hG4bK5163f585;rport
From: <sip:321 at 10.0.0.127>;tag=as2e55e96c
To: "George Sip Burt" <sip:grandstream1 at 10.0.0.127>;tag=23de801d5cc2edfd
Contact: <sip:321 at 10.0.0.127>
Call-ID: 75a24ffcd23bc78e at 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 at 10.0.0.127>;tag=as2e55e96c
To: "George Sip Burt" <sip:grandstream1 at 10.0.0.127>;tag=23de801d5cc2edfd
Call-ID: 75a24ffcd23bc78e at 10.0.0.26
CSeq: 102 BYE
User-Agent: Grandstream BT100 1.0.4.50
Contact: <sip:grandstream1 at 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 at 10.0.0.26'
Destroying call 'cdbc92394c507afc at 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 at 10.0.0.127>;tag=72ca48fb08ef6b70
To: <sip:grandstream1 at 10.0.0.127>
Contact: *
Call-ID: cdbc92394c507afc at 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 at 10.0.0.127>;tag=72ca48fb08ef6b70
To: <sip:grandstream1 at 10.0.0.127>;tag=as2d358f61
Call-ID: cdbc92394c507afc at 10.0.0.26
CSeq: 133 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:grandstream1 at 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
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