Sean Cheesman
2004-Jan-03 10:14 UTC
[Asterisk-Users] Newbie - getting two local phones tocommunicate would be a good start :)
Hi John,
Try adding username=5702 and username=5703 to each of the configs in
sip.conf. I recall I had this problem with the Grandstreams.
-----Original Message-----
From: John Coll [mailto:john.coll@csoft.co.uk]
Sent: Saturday, January 03, 2004 11:56 AM
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] Newbie - getting two local phones
tocommunicate would be a good start :)
Steven - thanks for that. OK I will try and ask "interesting and
directed questions" :)
I appreciate the support from several people. Rich Adamson encouraged me
to hang in there so I've been back at the shell prompt and edited
configuration files down to the bare essentials and still get the same.
I would appreciate any suggestions ....
To sumarise:
Asterisk and 2 Grandstrem phones on a LAN. The * box (liza) is RH9 and
happens to have a firewall connected to the outside but * and the SIP
phones are all on the same LAN. The * machine can ping both SIP phones
fine. The SIP phones can call each other's IP directly and establish a
voice path - but not via *
Just in case here is the config of one of the SIP phones
Login password xxx
MAC 00.0B.82.00.4B.57
IP 10.0.1.202
Subnet 255.255.255.0 (all machines are /24 on this LAN)
Default router 10.0.1.198
DNS server #1 10.0.1.198
DNS Server #2 158.152.1.43
SIP Server: 10.0.1.198
Outbound Proxy:
SIP User ID: 5702
Authenticate ID: 5702
Authenticate Password:
Name: John Coll 5702
Timezone: GMT
SIP User ID is phone number: yes
I've experimented with adding an Authenticate Password and adding
secret=xxx to sip.conf - but that did not help.
Here are the main asterisk configuration files:
------------------------------------------------------------------------
----
-
;
; liza:/etc/asterisk/sip.conf
;
[general]
port = 5060
bindaddr = 0.0.0.0
externip = 10.0.1.198
[5702]
type=friend
host=dynamic
context=johnhome
reinvite=no
canreinvite=no
qualify=300
callerid="John workroom #1" <5702>
mailbox=5702
nat=yes
[5703]
type=friend
host=dynamic
context=johnhome
reinvite=no
canreinvite=no
qualify=300
callerid="John workroom #2" <5703>
mailbox=5703
nat=yes
------------------------------------------------------------------------
----
-
;
; liza:/etc/asterisk/extensions.conf
;
[general]
static=yes
writeprotect=no
;
[globals]
CONSOLE=Console/dsp
[johnhome]
exten => 5702,1,Dial(SIP/5702,20,Ttr)
exten => 5702,2,Voicemail(u5702)
exten => 5702,102,Voicemail(b5702)
exten => 5702,103,Hangup
exten => 5703,1,Dial(SIP/5703,20,Ttr)
exten => 5703,2,Voicemail(u5703)
exten => 5703,102,Voicemail(b5703)
exten => 5703,103,Hangup
------------------------------------------------------------------------
----
-
I have created voicemail boxes for 5702 and 5703 using context johnhome.
[root@liza johnhome]# pwd /var/spool/asterisk/voicemail/johnhome
[root@liza johnhome]# l
total 16
drwxr-xr-x 4 root root 4096 Jan 3 15:41 .
drwxr-xr-x 4 root root 4096 Jan 3 15:41 ..
drwxr-xr-x 3 root root 4096 Jan 3 15:41 5702
drwxr-xr-x 3 root root 4096 Jan 3 15:41 5703
[root@liza johnhome]#
------------------------------------------------------------------------
----
-
Other configuration files exist in /etc/asterisk
[root@liza asterisk]# ls
adsi.conf cdr_pgsql.conf johncoll modem.conf
phone.conf sample vpb.conf
adtranvofr.conf enum.conf john_todd modules.conf
privacy.conf sip.conf z2.conf
agents.conf extensions.conf logger.conf musiconhold.conf
queues.conf skinny.conf zapata.conf
alsa.conf festival.conf manager.conf orig
rpt.conf
sprackett
asterisk.adsi iax.conf meetme.conf oss.conf
rtp.conf
telcordia-1.adsi
asterisk.conf indications.conf mgcp.conf parking.conf
s2.conf
voicemail.conf
[root@liza asterisk]#
------------------------------------------------------------------------
----
-
one other that might have some influence perhaps
;
; liza:/etc/asterisk/manager.conf
;
[general]
enabled = no
port = 5038
bindaddr = 0.0.0.0
------------------------------------------------------------------------
----
-
I believe that liza:/etc/asterisk/zapata.conf and liza:/etc/zaptel.conf
are not relevant but they exist
------------------------------------------------------------------------
----
-
Starting asterisk up - not too verbose....
[root@liza asterisk]# asterisk -vnc
Asterisk CVS-12/12/03-17:13:48, Copyright (C) 1999-2001 Linux Support
Services, Inc. Written by Mark Spencer <markster@linux-support.net>
=======================================================================Asterisk
Event Logger Started /var/log/asterisk/event_log Asterisk PBX
Core Initializing Registering builtin applications: [AbsoluteTimeout]
[Answer] [BackGround] [Busy] [Congestion] [DigitTimeout] [Goto]
[GotoIf] [GotoIfTime] [Hangup] [NoOp] [Prefix] [ResetCDR]
[ResponseTimeout] [Ringing] [SayNumber] [SayDigits] [SetAccount]
[SetGlobalVar] [SetLanguage] [SetVar] [StripMSD] [Suffix] [Wait]
Asterisk Dynamic Loader Starting: [chan_modem.so] => (Generic Voice
Modem Driver) => (A/Open (Rockwell Chipset) ITU-2 VoiceModem Driver)
[res_musiconhold.so] => (Music On Hold Resource) [res_adsi.so] => (ADSI
Resource) [res_parking.so] => (Call Parking Resource) [res_crypto.so]
=> (Cryptographic Digital Signatures) [res_indications.so] =>
(Indications Configuration)
-- Registered indication country 'us'
-- Registered indication country 'au'
-- Registered indication country 'fr'
-- Registered indication country 'de'
-- Registered indication country 'nl'
-- Registered indication country 'uk'
-- Registered indication country 'fi'
-- Registered indication country 'no'
-- Registered indication country 'br'
-- Setting default indication country to 'uk' [res_monitor.so]
=>
(Call Monitoring Resource) [chan_iax.so] => (Inter Asterisk eXchange)
[chan_sip.so] => (Session Initiation Protocol (SIP))
-- SIP Seeding '5702' at 10.0.1.202:5060 for 3600
-- SIP Seeding '5703' at 10.0.1.203:5060 for 3600
[chan_modem_bestdata.so] => (BestData (Conexant V.90 Chipset) VoiceModem
Driver)
[chan_modem_i4l.so] => (ISDN4Linux Emulated Modem Driver)
[chan_agent.so] => (Agent Proxy Channel) [chan_mgcp.so] => (Media
Gateway Control Protocol (MGCP)) [chan_iax2.so] => (Inter Asterisk
eXchange (Ver 2))
WARNING[16384]: File chan_iax2.c, Line 5465 (set_config): Ignoring port
for now [chan_local.so] => (Local Proxy Channel) [chan_skinny.so] =>
(Skinny Client Control Protocol (Skinny)) [chan_oss.so] => (OSS Console
Channel Driver)
WARNING[16384]: File chan_oss.c, Line 429 (soundcard_init): Unable to
open
/dev/dsp: No such device
[chan_phone.so] => (Linux Telephony API Support)
[chan_zap.so] => (Zapata Telephony w/PRI)
[pbx_config.so] => (Text Extension Configuration) [pbx_wilcalu.so] =>
(Wil Cal U (Auto Dialer)) [pbx_spool.so] => (Outgoing Spool Support)
/var/spool/asterisk/outgoing [app_dial.so] => (Dialing Application)
[app_playback.so] => (Trivial Playback Application) [app_voicemail.so]
=> (Comedian Mail (Voicemail System)) [app_directory.so] => (Extension
Directory) [app_mp3.so] => (Silly MP3 Application) [app_system.so] =>
(Generic System() application) [app_echo.so] => (Simple Echo
Application) [app_record.so] => (Trivial Record Application)
[app_image.so] => (Image Transmission Application) [app_url.so] =>
(Send URL Applications) [app_disa.so] => (DISA (Direct Inward System
Access) Application) [app_agi.so] => (Asterisk Gateway Interface (AGI))
[app_qcall.so] => (Call from Queue) [app_adsiprog.so] => (Asterisk ADSI
Programming Application) [app_getcpeid.so] => (Get ADSI CPE ID)
[app_milliwatt.so] => (Digital Milliwatt (mu-law) Test Application)
[app_zapateller.so] => (Block Telemarketers with Special Information
Tone) [app_datetime.so] => (Date and Time) [app_setcallerid.so] =>
(Set CallerID Application) [app_festival.so] => (Simple Festival
Interface) [app_queue.so] => (True Call Queueing) [app_senddtmf.so] =>
(Send DTMF digits Application) [app_parkandannounce.so] => (Call
Parking and Announce Application) [app_striplsd.so] => (Strip trailing
digits) [app_setcidname.so] => (Set CallerID Name)
[app_lookupcidname.so] => (Look up CallerID Name from local database)
[app_substring.so] => (Save substring digits in a given variable)
[app_macro.so] => (Extension Macros) [app_authenticate.so] =>
(Authentication Application) [app_softhangup.so] => (Hangs up the
requested channel) [app_lookupblacklist.so] => (Look up Caller*ID
name/number from blacklist
database)
[app_waitforring.so] => (Waits until first ring after time)
[app_privacy.so] => (Require phone number to be entered, if no CallerID
sent)
[app_db.so] => (Database access functions for Asterisk extension logic)
[app_chanisavail.so] => (Check if channel is available)
[app_enumlookup.so] => (ENUM Lookup) [app_transfer.so] => (Transfer)
[app_setcidnum.so] => (Set CallerID Number) [app_cdr.so] => (Make sure
asterisk doesn't save CDR for a certain call) [app_hasnewvoicemail.so]
=> (Indicator for whether a voice mailbox has new
messages.)
[app_sayunixtime.so] => (Say time)
[app_cut.so] => (Cuts up variables)
[app_read.so] => (Read Variable Application)
[skipping app_intercom.so]
[app_zapras.so] => (Zap RAS Application)
[app_meetme.so] => (Simple MeetMe conference bridge) [app_flash.so] =>
(Flash zap trunk application) [app_zapbarge.so] => (Barge in on Zap
channel application) [codec_ilbc.so] => (iLBC/PCM16 (signed linear)
Codec Translator) [codec_gsm.so] => (GSM/PCM16 (signed linear) Codec
Translator) [codec_lpc10.so] => (LPC10 2.4kbps (signed linear) Voice
Coder) [codec_adpcm.so] => (Adaptive Differential PCM Coder/Decoder)
[codec_ulaw.so] => (Mu-law Coder/Decoder) [codec_alaw.so] => (A-law
Coder/Decoder) [codec_a_mu.so] => (A-law and Mulaw direct
Coder/Decoder) [format_gsm.so] => (Raw GSM data) [format_wav.so] =>
(Microsoft WAV format (8000hz Signed Linear)) [format_wav_gsm.so] =>
(Microsoft WAV format (Proprietary GSM)) [format_vox.so] => (Dialogic
VOX (ADPCM) File Format) [format_pcm.so] => (Raw uLaw 8khz Audio
support (PCM)) [format_g729.so] => (Raw G729 data)
[format_pcm_alaw.so] => (Raw aLaw 8khz PCM Audio support)
[format_h263.so] => (Raw h263 data) [format_jpeg.so] => (JPEG (Joint
Picture Experts Group) Image Format) [cdr_csv.so] => (Comma Separated
Values CDR Backend) Asterisk Ready.
------------------------------------------------------------------------
----
-
Just two warnings in the above and they sound benign....
And now dial 5703 from 5702. 5703 rings but when 5703 is taken off hook
no voice path is established and both phones give rapid beep beep beep
after a few seconds. The following has been cut a bit but I hope I've
left something useful in there....
*CLI>
*CLI> sip debug
SIP Debugging Enabled
Sip read:
INVITE sip:5703@10.0.1.198;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.0.1.202
+++++++++++++++++++++++++ divider inserted to aid reading ++++++++++++++
From: "John Coll 5702"
<sip:5702@10.0.1.198;user=phone>;tag=633d95e4-ede7-3b06-2c26-bdb01931bb9
5
To: <sip:5703@10.0.1.198;user=phone>
Contact: <sip:5702@10.0.1.202;user=phone>
Call-ID: 12555207-9a00-2c4a-1df3-38ba95e427aa@10.0.1.202
CSeq: 24755 INVITE
User-Agent: Grandstream SIP UA 1.0.4.17
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Content-Length: 253
v=0
o=5702 0 0 IN IP4 10.0.1.202
s=-
c=IN IP4 10.0.1.202
t=0 0
m=audio 5004 RTP/AVP 0 8 4 18 2 15
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=ptime:20
12 headers, 13 lines
Using latest request as basis request
Sending to 10.0.1.202 : 5060 (non-NAT)
Found audio format UNKN
<cut>
Found description format PCMU
<cut>
Capabilities: us - 524302, them - 285/0, combined - 12 Non-codec
capabilities: us - 1, them - 0, combined - 0 Looking for 5703 in
johnhome
list_route: hop: <sip:5702@10.0.1.202;user=phone>
Transmitting (NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.1.202;received=10.0.1.202
+++++++++++++++++++++++++ divider inserted to aid reading ++++++++++++++
From: "John Coll 5702"
<sip:5702@10.0.1.198;user=phone>;tag=633d95e4-ede7-3b06-2c26-bdb01931bb9
5
To: <sip:5703@10.0.1.198;user=phone>;tag=as24ca7ae1
Call-ID: 12555207-9a00-2c4a-1df3-38ba95e427aa@10.0.1.202
CSeq: 24755 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:5703@10.0.1.198>
Content-Length: 0
to 10.0.1.202:5060
We're at 10.0.1.198 port 18360
Answering with capability 2
Answering with capability 4
Answering with capability 8
Answering with non-codec capability 1
11 headers, 11 lines
Reliably Transmitting:
INVITE sip:10.0.1.203 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.198:5060;branch=z9hG4bK7b229a46
+++++++++++++++++++++++++ divider inserted to aid reading ++++++++++++++
From: "John workroom #1" <sip:5702@10.0.1.198>;tag=as3e1081f0
To: <sip:10.0.1.203>
Contact: <sip:5702@10.0.1.198>
Call-ID: 26e0e4083e0dcc9202a36bc566f60607@10.0.1.198
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 232
v=0
o=root 24767 24767 IN IP4 10.0.1.198
s=session
c=IN IP4 10.0.1.198
t=0 0
m=audio 18360 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
(NAT) to 10.0.1.203:5060
Transmitting (NAT):
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.0.1.202;received=10.0.1.202
+++++++++++++++++++++++++ divider inserted to aid reading ++++++++++++++
From: "John Coll 5702"
<sip:5702@10.0.1.198;user=phone>;tag=633d95e4-ede7-3b06-2c26-bdb01931bb9
5
To: <sip:5703@10.0.1.198;user=phone>;tag=as24ca7ae1
Call-ID: 12555207-9a00-2c4a-1df3-38ba95e427aa@10.0.1.202
CSeq: 24755 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:5703@10.0.1.198>
Content-Length: 0
to 10.0.1.202:5060
Sip read:
SIP/2.0 100 trying
Via: SIP/2.0/UDP 10.0.1.198:5060;branch=z9hG4bK7b229a46
From: "John workroom #1" <sip:5702@10.0.1.198>;tag=as3e1081f0
To: <sip:10.0.1.203>
Call-ID: 26e0e4083e0dcc9202a36bc566f60607@10.0.1.198
CSeq: 102 INVITE
User-Agent: Grandstream SIP UA 1.0.4.17
Content-Length: 0
8 headers, 0 lines
Sip read:
SIP/2.0 180 ringing
Via: SIP/2.0/UDP 10.0.1.198:5060;branch=z9hG4bK7b229a46
+++++++++++++++++++++++++ divider inserted to aid reading ++++++++++++++
From: "John workroom #1" <sip:5702@10.0.1.198>;tag=as3e1081f0
To: <sip:10.0.1.203>;tag=f42d82b9-1d86-dbf7-c9f6-e69f5f0e25d0
Call-ID: 26e0e4083e0dcc9202a36bc566f60607@10.0.1.198
CSeq: 102 INVITE
User-Agent: Grandstream SIP UA 1.0.4.17
Content-Length: 0
8 headers, 0 lines
We're at 10.0.1.198 port 12168
Answering with capability 2
Answering with capability 4
Answering with capability 8
Transmitting (NAT):
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.0.1.202;received=10.0.1.202 =======================this next
chunk is repeated 3 times ================From: "John Coll 5702"
<sip:5702@10.0.1.198;user=phone>;tag=633d95e4-ede7-3b06-2c26-bdb01931bb9
5
To: <sip:5703@10.0.1.198;user=phone>;tag=as24ca7ae1
Call-ID: 12555207-9a00-2c4a-1df3-38ba95e427aa@10.0.1.202
CSeq: 24755 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:5703@10.0.1.198>
Content-Type: application/sdp
Content-Length: 176
v=0
o=root 24767 24767 IN IP4 10.0.1.198
s=session
c=IN IP4 10.0.1.198
t=0 0
m=audio 12168 RTP/AVP 3 0 8
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
to 10.0.1.202:5060
Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.1.198:5060;branch=z9hG4bK7b229a46
+++++++++++++++++++++++++ divider inserted to aid reading ++++++++++++++
From: "John workroom #1" <sip:5702@10.0.1.198>;tag=as3e1081f0
To: <sip:10.0.1.203>;tag=f42d82b9-1d86-dbf7-c9f6-e69f5f0e25d0
Call-ID: 26e0e4083e0dcc9202a36bc566f60607@10.0.1.198
CSeq: 102 INVITE
User-Agent: Grandstream SIP UA 1.0.4.17
Contact: <sip:5703@10.0.1.203;user=phone>
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Content-Length: 126
v=0
o=5703 0 0 IN IP4 10.0.1.203
s=-
c=IN IP4 10.0.1.203
t=0 0
m=audio 5004 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
11 headers, 8 lines
Found audio format UNKN
Found description format PCMU
Capabilities: us - 524302, them - 4/0, combined - 4
Non-codec capabilities: us - 1, them - 0, combined - 0
list_route: hop: <sip:5703@10.0.1.203;user=phone>
set_destination: Parsing <sip:5703@10.0.1.203;user=phone> for
address/port to send to
set_destination: set destination to 10.0.1.203, port 5060
Transmitting:
ACK sip:5703@10.0.1.203 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.198:5060;branch=z9hG4bK7b229a46
+++++++++++++++++++++++++ divider inserted to aid reading ++++++++++++++
From: "John workroom #1" <sip:5702@10.0.1.198>;tag=as3e1081f0
To: <sip:10.0.1.203>;tag=f42d82b9-1d86-dbf7-c9f6-e69f5f0e25d0
Contact: <sip:5702@10.0.1.198>
Call-ID: 26e0e4083e0dcc9202a36bc566f60607@10.0.1.198
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0
(NAT) to 10.0.1.203:5060
We're at 10.0.1.198 port 12168
Answering with capability 2
Answering with capability 4
Answering with capability 8
Reliably Transmitting (NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.1.202;received=10.0.1.202
======================================= end of repeated chunk
=============
+++++++++++++++++++++++++ divider inserted to aid reading ++++++++++++++
From: "John Coll 5702"
<sip:5702@10.0.1.198;user=phone>;tag=633d95e4-ede7-3b06-2c26-bdb01931bb9
5
To: <sip:5703@10.0.1.198;user=phone>;tag=as24ca7ae1
Call-ID: 12555207-9a00-2c4a-1df3-38ba95e427aa@10.0.1.202
CSeq: 24755 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:5703@10.0.1.198>
Content-Type: application/sdp
Content-Length: 176
v=0
o=root 24767 24768 IN IP4 10.0.1.198
s=session
c=IN IP4 10.0.1.198
t=0 0
m=audio 12168 RTP/AVP 3 0 8
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
to 10.0.1.202:5060
WARNING[81926]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries
exceeded on call 12555207-9a00-2c4a-1df3-38ba95e427aa@10.0.1.202 for
seqno 24755 (Response)
set_destination: Parsing <sip:5703@10.0.1.203;user=phone> for
address/port to send to
set_destination: set destination to 10.0.1.203, port 5060 Reliably
Transmitting: BYE sip:5703@10.0.1.203 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.198:5060;branch=z9hG4bK7b229a46
+++++++++++++++++++++++++ divider inserted to aid reading ++++++++++++++
From: "John workroom #1" <sip:5702@10.0.1.198>;tag=as3e1081f0
To: <sip:10.0.1.203>;tag=f42d82b9-1d86-dbf7-c9f6-e69f5f0e25d0
Contact: <sip:5702@10.0.1.198>
Call-ID: 26e0e4083e0dcc9202a36bc566f60607@10.0.1.198
CSeq: 103 BYE
User-Agent: Asterisk PBX
Content-Length: 0
(NAT) to 10.0.1.203:5060
set_destination: Parsing <sip:5702@10.0.1.202;user=phone> for
address/port to send to
set_destination: set destination to 10.0.1.202, port 5060 Reliably
Transmitting: BYE sip:5702@10.0.1.202 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.198:5060;branch=z9hG4bK1437f2b8
+++++++++++++++++++++++++ divider inserted to aid reading ++++++++++++++
From: <sip:5703@10.0.1.198;user=phone>;tag=as24ca7ae1
To: "John Coll 5702"
<sip:5702@10.0.1.198;user=phone>;tag=633d95e4-ede7-3b06-2c26-bdb01931bb9
5
Contact: <sip:5703@10.0.1.198>
Call-ID: 12555207-9a00-2c4a-1df3-38ba95e427aa@10.0.1.202
CSeq: 102 BYE
User-Agent: Asterisk PBX
Content-Length: 0
(NAT) to 10.0.1.202:5060
Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.1.198:5060;branch=z9hG4bK7b229a46
+++++++++++++++++++++++++ divider inserted to aid reading ++++++++++++++
From: "John workroom #1" <sip:5702@10.0.1.198>;tag=as3e1081f0
To: <sip:10.0.1.203>;tag=f42d82b9-1d86-dbf7-c9f6-e69f5f0e25d0
Call-ID: 26e0e4083e0dcc9202a36bc566f60607@10.0.1.198
CSeq: 103 BYE
User-Agent: Grandstream SIP UA 1.0.4.17
Contact: <sip:5703@10.0.1.203;user=phone>
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE
Content-Length: 0
10 headers, 0 lines
Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.1.198:5060;branch=z9hG4bK1437f2b8
+++++++++++++++++++++++++ divider inserted to aid reading ++++++++++++++
From: <sip:5703@10.0.1.198;user=phone>;tag=as24ca7ae1
To: "John Coll 5702"
<sip:5702@10.0.1.198;user=phone>;tag=633d95e4-ede7-3b06-2c26-bdb01931bb9
5
Call-ID: 12555207-9a00-2c4a-1df3-38ba95e427aa@10.0.1.202
CSeq: 102 BYE
User-Agent: Grandstream SIP UA 1.0.4.17
Contact: <sip:5702@10.0.1.202;user=phone>
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE
Content-Length: 0
10 headers, 0 lines
Message is BYE
10 headers, 0 lines
Reliably Transmitting:
OPTIONS sip:10.0.1.203 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.198:5060;branch=z9hG4bK76827705
+++++++++++++++++++++++++ divider inserted to aid reading ++++++++++++++
From: "asterisk" <sip:asterisk@10.0.1.198>;tag=as7437a2f3
To: <sip:10.0.1.203>
Contact: <sip:asterisk@10.0.1.198>
Call-ID: 46780f326bc759660a9a0ae7686ba6bb@10.0.1.198
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Length: 0
(no NAT) to 10.0.1.203:5060
10 headers, 0 lines
Reliably Transmitting:
OPTIONS sip:10.0.1.202 SIP/2.0
Via: SIP/2.0/UDP 10.0.1.198:5060;branch=z9hG4bK4ac25865
+++++++++++++++++++++++++ divider inserted to aid reading ++++++++++++++
From: "asterisk" <sip:asterisk@10.0.1.198>;tag=as0c8f9b3f
To: <sip:10.0.1.202>
Contact: <sip:asterisk@10.0.1.198>
Call-ID: 5ad587c34824ebb9180be9b724247378@10.0.1.198
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Length: 0
(no NAT) to 10.0.1.202:5060
Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.1.198:5060;branch=z9hG4bK76827705
+++++++++++++++++++++++++ divider inserted to aid reading ++++++++++++++
From: "asterisk" <sip:asterisk@10.0.1.198>;tag=as7437a2f3
To: <sip:10.0.1.203>;tag=48c5ac3e-dbf7-f42d-e69f-1d8625d0c9f6
Call-ID: 46780f326bc759660a9a0ae7686ba6bb@10.0.1.198
CSeq: 102 OPTIONS
User-Agent: Grandstream SIP UA 1.0.4.17
Contact: <sip:5703@10.0.1.203;user=phone>
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE
Content-Length: 0
10 headers, 0 lines
Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.1.198:5060;branch=z9hG4bK4ac25865
+++++++++++++++++++++++++ divider inserted to aid reading ++++++++++++++
From: "asterisk" <sip:asterisk@10.0.1.198>;tag=as0c8f9b3f
To: <sip:10.0.1.202>;tag=ede73b06-2c26-bdb0-1931-bb9512555207
Call-ID: 5ad587c34824ebb9180be9b724247378@10.0.1.198
CSeq: 102 OPTIONS
User-Agent: Grandstream SIP UA 1.0.4.17
Contact: <sip:5702@10.0.1.202;user=phone>
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE
Content-Length: 0
+++++++++++++++++++++++++ divider inserted to aid reading ++++++++++++++
-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Steven
Critchfield
Sent: 03 January 2004 14:39
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] Newbie - getting two local phones
tocommunicate would be a good start :)
On Sat, 2004-01-03 at 06:31, John Coll wrote:> SW: Thanks a million for the statement that I only need these two
> files
and> they can be just about empty !
>
> David Carter: many thanks for those files which I will study
>
> Rich Adamson: That is so re-assuring! That may sound odd but its realy
> helpful to have the problems I am facing acknowledged and makes me
> feel
that> others really see the need for, in effect, intuitive docs to get the
novice> on-board. I used to write code, now I leave it to my staff, but I
> guess I can go there. What I am doing is evaluating * to see if we as
> a company should use and support it rather than just buying in Quintum
> boxes or whatever. No doubt many others are doing the same.
Be aware, that the documentation is getting better. There is info going
into the wiki daily and there is a book being written. This kind of
documentation is only needed when we get over run by people who aren't
willing to take their time learning and willing to spend a lot of
effort. We built up a large group of developers and supporters without
much documentation. This allowed us to to move forward at a pretty
decent pace. Documentation is usually neglected during periods of fast
growth.
> As a company we write software for end-users and I insist that an
> average
16> year old must be able to make it work, at the basic level, without
> grief - it must be intuitive. OK make that "an average linux
> administrator" for */VOIP but again it really needs to be intuitive -
> but I guess I am preaching to the convereted.
I think you need to go meet and get to know some pbx installers. Maybe
make some calls to your local CLECS and requests sales support by a
engineer there. You will eventually learn that telephony is a large
field that takes quite a bit of effort to understand. Your expectations
that telephony be easy will probably not be met unless you have the
prerequisite knowledge to begin with.
During my last job, I was able to talk to the man who the company hired
to install their Intertel pbx. I ended up with the distinct feeling from
him that the industry has been progressing much as old trades did.
Basically it seemed that one had to study by being a go-fer for a person
who knew what they where doing for some time before you could pick up
the required knowledge to do simple installs.
So while we haven't improved the amount of knowledge you will need to
acquire to begin a decent install, but we have developed a community
that will help those willing to help themselves. Those who aren't
willing to put forth the effort have the ability to pay for the support
they need.
> I would like to offer to try and do that in the wiki - but
> realistically I don't have the time. Still I am feeling a bit guilty
> now having got such solid support.
The biggest time consumer is the amount you must learn before you can
start documenting. There are many people here who are contributing to
documentation. What you can do to help now is to ask interesting and
directed questions that will be answered by members of this group. When
it is sufficiently answered, Olle tends to get it incorporated into the
wiki.
--
Steven Critchfield <critch@basesys.com>
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SW
2004-Jan-03 10:29 UTC
[Asterisk-Users] Newbie - getting two local phones tocommunicate would be a good start :)
John, Obviousely, this would not work. Look at my example before; [5702] <=====type=friend username=5702 <=====context=internal dtmfmode=info username and context should match. Better get it working in a simple LAN first, why NAT, why voicemail ...... Go to basics :), SW From: "John Coll" <john.coll@csoft.co.uk> To: <asterisk-users@lists.digium.com> Subject: RE: [Asterisk-Users] Newbie - getting two local phones tocommunicate would be a good start :) Date: Sat, 3 Jan 2004 16:55:45 -0000 Reply-To: asterisk-users@lists.digium.com ; liza:/etc/asterisk/sip.conf ; [general] port = 5060 bindaddr = 0.0.0.0 externip = 10.0.1.198 [5702] type=friend host=dynamic context=johnhome reinvite=no canreinvite=no qualify=300 callerid="John workroom #1" <5702> mailbox=5702 nat=yes [5703] type=friend host=dynamic context=johnhome reinvite=no canreinvite=no qualify=300 callerid="John workroom #2" <5703> mailbox=5703 nat=yes