Assignment 2: H323 (by Fabio & Anneke)

We use gnomemeeting softphones and a gnugk gatekeeper.

Configuration of Gnomemeeting

Edit -> Preferences

• General: You can fill in your personal data and leave everything default
• H323 Settings: Gatekeeper:

Registering method: Gatekeeper host
Gatekeeper host: 129.125.71.171:1719
Gatekeeper alias: 601

Gatekeeper host is the IP address and port of the gatekeeper. Port 1720 is the standard h323 port .

Gatekeeper alias is the primary alias with the gatekeeper, also the extension number in Asterisk to keep things simple.

Tick the "Register this alias as the primary alias with the gatekeeper" box and Apply (sorry, you need to do this after you've installed the gk) :).

• H323 Settings: Gateway:
  • For H323 to H323 - Do not use a gateway, these calls will only go via the gatekeeper, not the trixbox
  • However for H323 to SIP a gateway to the trixbox (so trixbox ip) is needed but this messes up the h323 to h323 settings. Right now we don't have another solution than manually check or uncheck the box depending on what kind of call you want to make :S
• Codecs: Audio Codecs - We enabled all available codecs.
• Codecs: Video Codecs - We have not been able to spend time on this
• Devices:
  • Audio Devices:
    • Audio Plugin ALSA
    • Output device: Intel ICH5
    • Input device: Intel ICH5
  • Detect Devices

(All the audio and video devices settings are of course depending on your local system, this is more for our own reference) :)

Alternative: Ekiga

To check the codec availability we decided to switch back to Ekiga since it supports SIP and H.323. We first made a SIP call and this worked fine. Switching to H.323 and calling a SIP phone and calling from the SIP phone to Ekiga(H.323) does not report any codec problems. (It seems that this is just a consequence of switching to Ekiga, no other parameters were changed).

GnuGk Gatekeeper

Packages for this gatekeeper are not available for CentOS. We found out that this gatekeeper was already installed in Etch. (System administration provided us with two machines with etch so we could get the sound working).

Almost no configuring here. Just enable PROXY.

Once the gatekeeper is installed you can register your gnomemeeting clients with the gatekeeper (see section gnomemeeting.

We are still playing with the gatekeeper so this section is under construction.

Configuration of Trixbox 1.2.3

We followed mostly the instructions from these two forums on the Trixbox site:

• here is the working configuration & tips for oh323
• H323, SIP and Gatekeeper

There are more forums where people say things about h323 but these seem to be the most helpful.

Following the suggestions we used Trixbox 1.2.3 with asterisk-addons-1.2.3. In order to do this we deleted the current addons:

rpm -qa | grep asterisk-addons
rpm -e asterisk-addons-1.2.4_1.2.12.1-1.294

And loaded the older addons: (these can be found ...............)

rpm -i asterisk-addons-1.2.3-1.219.i386.rpm

If this is really necessary we can't tell right now. We got things working when changing other things as well so later on we can try if it still works without this step.

ooh323.conf

Next thing was to edit the ooh323.conf file. This file needs to be copied from the samples first:

cp /etc/asterisk-1.2.8-samples/ooh323.conf /etc/asterisk 

Here is our ooh323.conf:

; Objective System's H323 Configuration example for Asterisk
; ooh323c driver configuration
;
; [general] section defines global parameters
;
; This is followed by profiles which can be of three types - user/peer/friend
; Name of the user profile should match with the h323id of the user device.
; For peer/friend profiles, host ip address must be provided as "dynamic" is
; not supported as of now.
;
; Syntax for specifying a H323 device in extensions.conf is
; For Registered peers/friends profiles:
;        OOH323/name where name is the name of the peer/friend profile.
;
; For unregistered H.323 phones:
;        OOH323/ip[:port] OR if gk is used OOH323/alias where alias can be any H323
;                          alias
;
; For dialing into another asterisk peer at a specific exten
;       OOH323/exten/peer OR OOH323/exten@ip
;
; Domain name resolution is not yet supported.
;
; When a H.323 user calls into asterisk, his H323ID is matched with the profile
; name and context is determined to route the call
;
; The channel driver will register all global aliases and aliases defined in
; peer profiles with the gatekeeper, if one exists. So, that when someone
; outside our pbx (non-user) calls an extension, gatekeeper will route that
; call to our asterisk box, from where it will be routed as per dial plan.


[general]
;Define the asetrisk server h323 endpoint

;The port asterisk should listen for incoming H323 connections.
;Default - 1720
;port=1720
;CHANGED
port=1720

;The dotted IP address asterisk should listen on for incoming H323
;connections
;Default - tries to find out local ip address on it's own
bindaddr=0.0.0.0

;This parameter indicates whether channel driver should register with
;gatekeeper as a gateway or an endpoint.
;Default - no
;gateway=no
;CHANGED
gateway=yes

;Whether asterisk should use fast-start and tunneling for H323 connections.
;Default - yes
;faststart=no
;h245tunneling=no
;CHANGED
h245tunneling=yes


;H323-ID to be used for asterisk server
;Default - Asterisk PBX
h323id=ObjSysAsterisk
;e164=100
;CHANGED
e164=501

;CallerID to use for calls
;Default - Same as h323id
callerid=asterisk

;Whether this asterisk server will use gatekeeper.
;Default - DISABLE
;gatekeeper = DISCOVER
;gatekeeper = DISABLE
;CHANGED
gatekeeper = 129.125.71.171

;Location for H323 log file
;Default - /var/log/asterisk/h323_log
;CHANGED
logfile=/var/log/asterisk/h323_log


;Following values apply to all users/peers/friends defined below, unless
;overridden within their client definition
 
;Sets default context all clients will be placed in.
;Default - default
context=default

;Sets rtptimeout for all clients, unless overridden
;Default - 60 seconds
;rtptimeout=60      ; Terminate call if 60 seconds of no RTP activity
                    ; when we're not on hold
;CHANGED
rtptimeout=3
;Type of Service
;Default - none (lowdelay, thoughput, reliability, mincost, none)
;tos=lowdelay

;amaflags = default

;The account code used by default for all clients.
;accountcode=h3230101

;The codecs to be used for all clients.Only ulaw and gsm supported as of now.
;Default - ulaw
; ONLY ulaw, gsm, g729 and g7231 supported as of now
;disallow=all     ;Note order of disallow/allow is important.
;allow=gsm
;allow=ulaw
;;CHANGED
;allow=g729
;allow=g723

;CHANGED
allow=all

; dtmf mode to be used by default for all clients. Supports rfc2833, q931keypad
; h245alphanumeric, h245signal.
;Default - rfc 2833
dtmfmode=rfc2833

; User/peer/friend definitions:
; User config options                    Peer config options
; ------------------                     -------------------
; context
; disallow                               disallow
; allow                                  allow
; accountcode                            accountcode
; amaflags                               amaflags
; dtmfmode                               dtmfmode
; rtptimeout                             ip
;                                        port
;                                        h323id
;                                        email
;                                        url
;                                        e164
;                                        rtptimeout

;

;Define users here
;Section header is extension
[myuser1]
type=user
context=context1
disallow=all
allow=gsm
allow=ulaw



[mypeer1]
type=peer
context=context2
ip=a.b.c.d   ; UPDATE with appropriate ip address
port=1720    ; UPDATE with appropriate port
e164=101



;[myfriend1]

[friend-Anneke]
type=friend
;CHANGED
context=from-internal
ip=129.125.71.172   ; UPDATE with appropriate ip address
;port=1820           ; UPDATE with appropriate port
port=1720
disallow=all
;allow=ulaw
;CHANGED
allow=gsm
allow=ulaw
allow=g729
allow=g723

;CHANGED
;e164=12345
e164=601

rtptimeout=3
dtmfmode=rfc2833


[friend-Fabio]
type=friend
context=default
ip=129.125.71.173   ; UPDATE with appropriate ip address
;port=1820           ; UPDATE with appropriate port
port=1720
disallow=all
;allow=ulaw
;CHANGED
allow=gsm
allow=ulaw
allow=g729
allow=g723

;CHANGED
;e164=12345

rtptimeout=3
dtmfmode=rfc2833

;[601]
;type=peer
;context=default
;ip=129.125.71.173:1719
;e164=100
;OH323/129.125.71.173:1719
;gatekeeper=ENABLE
;OOH323/601

(Still needs some cleaning up, will do later)

Comments on this file:

• All parameters that have been changed by us are preceded with a comment line ;CHANGED
• The Trixbox is going to act as gateway for the gatekeeper hence gateway = yes
• h245tunneling = yes
• For some reason it seems that you have to set e164 to the SIP extension number ??? (how to handle multiple extension???)
• We will be using a gatekeeper so gatekeeper = 129.125.71.173 (ip-address of the gatekeeper)
• The user profile section is a bit messy right now

Adding H323 extensions

A H323 extension can be added in FreePBX as a custom extension. For example:

Extension number: 600
Displayname: Anneke h323
Dial: OOH323/600@129.125.71.172 (ip-address of machine where extension can be found/maybe gatekeeper?)

Asterisk saves these extensions in the file extensions_additional.conf

General remarks

Whenever you edit anyting in the .conf files makes sure to do

amportal stop
amportal start

at you asterisk prompt. If you then do

asterisk -vvvvr

you get an CLI and here you can type

ooh323 debug
exit

to initiate the debugger

Progression

Using Gnomemeeting

• H323 -> H323: via gatekeeper only (gateway unchecked in gnomemeeting preferences)
• H323 -> SIP : rings but no connection can be establihed (gnomemeeting says: no commmon codec). These calls seem to be terminated by the trixbox.
• SIP -> H323 : SIP phone says there is a connection but receiving end gets very short signal followed by the message: call cleared by the gatekeeper. These seem to be terminated by the gatekeeper.

Using Gnomemeeting/Ekiga with GnuGK and TrixBox 1.2.3

• H323 -> H323: via gatekeeper only (gateway unchecked in gnomemeeting preferences)
• H323 -> SIP : rings but no connection can be establihed.
• SIP -> H323 : rings but no connection can be establihed.

We cannot find any errors in the gatekeeper log. The error seems to be in Asterisk. Both ways the ooh323 debug output says:

     -- Executing Macro("SIP/501-0841a378", "exten-vm|novm|602") in new stack
     -- Executing Macro("SIP/501-0841a378", "user-callerid") in new stack
     -- Executing GotoIf("SIP/501-0841a378", "0?report") in new stack
     -- Executing GotoIf("SIP/501-0841a378", "0?start") in new stack
     -- Executing Set("SIP/501-0841a378", "REALCALLERIDNUM=501") in new stack
     -- Executing NoOp("SIP/501-0841a378", "REALCALLERIDNUM is 501") in new stack
     -- Executing Set("SIP/501-0841a378", "AMPUSER=501") in new stack
     -- Executing Set("SIP/501-0841a378", "AMPUSERCIDNAME=Anneke SIP freePBX") in new stack
     -- Executing GotoIf("SIP/501-0841a378", "0?report") in new stack
     -- Executing Set("SIP/501-0841a378", "CALLERID(all)=Anneke SIP freePBX <501>") in new stack
     -- Executing NoOp("SIP/501-0841a378", "Using CallerID "Anneke SIP freePBX" <501>") in new stack
     -- Executing Set("SIP/501-0841a378", "FROMCONTEXT=exten-vm") in new stack
     -- Executing Set("SIP/501-0841a378", "VMBOX=novm") in new stack
     -- Executing Set("SIP/501-0841a378", "EXTTOCALL=602") in new stack
     -- Executing Set("SIP/501-0841a378", "CFUEXT=") in new stack
     -- Executing Set("SIP/501-0841a378", "RT=") in new stack
     -- Executing Macro("SIP/501-0841a378", "record-enable|602|IN") in new stack
     -- Executing GotoIf("SIP/501-0841a378", "0 > 0?2:4") in new stack
     -- Goto (macro-record-enable,s,4)
     -- Executing AGI("SIP/501-0841a378", "recordingcheck|20070226-192222|1172514142.0") in new stack
     -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
 PHP Warning:  Unknown(): Unable to load dynamic library '/usr/lib/php4/imap.so' - libc-client.so.0:    cannot open shared object file: No such  file or directory in Unknown on line 0 
  recordingcheck|20070226-192222|1172514142.0: Inbound recording not enabled 
     -- AGI Script recordingcheck completed, returning 0 
     -- Executing NoOp("SIP/501-0841a378", "No recording needed") in new stack
     -- Executing GotoIf("SIP/501-0841a378", "0?dolocaldial|1") in new stack
     -- Executing Macro("SIP/501-0841a378", "dial||tr|602") in new stack
     -- Executing AGI("SIP/501-0841a378", "dialparties.agi") in new stack
     -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
 PHP Warning:  Unknown(): Unable to load dynamic library '/usr/lib/php4/imap.so' - libc-client.so.0:  cannot open shared object file: No such  file or directory in Unknown on line 0 
  dialparties.agi: Starting New Dialparties.agi 
    --  dialparties.agi: priority is 1 
   dialparties.agi: Caller ID name is 'Anneke SIP freePBX' number is '501'
   dialparties.agi: Methodology of ring is  'none'
     --  dialparties.agi: Added extension 602 to extension map
     --  dialparties.agi: Extension 602 cf is disabled
     --  dialparties.agi: Extension 602 do not disturb is disabled
   == Parsing '/etc/asterisk/manager.conf': Found
   == Parsing '/etc/asterisk/manager_custom.conf': Found
   == Manager 'admin' logged on from 127.0.0.1
   == Manager 'admin' logged off from 127.0.0.1
     --  dialparties.agi: Checking CW and CFB status for extension 602
     --  dialparties.agi: DbSet CALLTRACE/602 to 501
     -- AGI Script dialparties.agi completed, returning 0
     -- Executing Dial("SIP/501-0841a378", "OOH323/602||tr") in new stack
 ---   ooh323_request - data 602 format 0x4 (ulaw)
 ---   find_peer
 +++   find_peer
 +++   ooh323_request
 ---   ooh323_call- 602
 +++   ooh323_call
     -- Called 602
 ---   onNewCallCreated ooh323c_o_1
 ---   find_call
 +++   find_call
 setting callid number 501
  Outgoing call 602(ooh323c_o_1) - Codec prefs - ()
         Adding capabilities to call(outgoing, ooh323c_o_1)
 ---   configure_local_rtp
 +++   configure_local_rtp
 +++   onNewCallCreated ooh323c_o_1
 --- onAlerting ooh323c_o_1
 ---   find_call
 +++   find_call
 +++ onAlerting ooh323c_o_1
     -- OOH323/602-ed80 is ringing
 ---   onCallEstablished ooh323c_o_1
 ---   find_call
 +++   find_call
 +++   onCallEstablished ooh323c_o_1
     -- OOH323/602-ed80 answered SIP/501-0841a378
 ---   onCallCleared ooh323c_o_1
 ---   find_call
 +++   find_call
 ---   ooh323_hangup
     hanging 602
 +++   ooh323_hangup
   == Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/501-0841a378' in macro 'dial'
   == Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/501-0841a378' in macro 'exten-vm'
   == Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/501-0841a378'
 ---   ooh323_destroy
  Destroying 602
 +++   ooh323_destroy

This output describes the situation where SIP extension 501 tries to call the H323 extension 602. In this experiment the callee's softphone rings, but after accepting the connection on the softphone, the connection is mysteriously dropped.

Because the telephone do ring, we can conclude that the H323 name resolution part is correctly carried out by the H323 gatekeeper GnuGK.

The next step for a h323 terminal then is to route a connection through the gateway, in this case the h323 channel driver implemented in Asterisk.

By carefully looking at the ooh323c channel driver's output, we notice that for both extensions the data declared in freePBX is found and a connection is set up.

When the called softphone tries to answer, the connection seems to get established; then the sudden hangup seems to originate in the macro-dial but the exact cause is uncertain (exited non-zero on "SIP/501-0841a378" ).

Using Ekiga with GnuGK and TrixBox 1.2.3 - Update

Another trial was done with almost the same setup: we suspect that the inbound/outbound routes do not play any role when routing calls locally on a Trixbox, even if between SIP and H323. Moreover, we applied a couple of changes:

1.in ooh323.conf check;

h245tunneling=no
faststart=no

2.in sip.conf under the line "tos=0x68" check;

notransfer=yes
canreinvite=no

We also activated both SIP and ooh323c debug messages on Asterisk; unluckily the end result is the very same: the phones ring but the voice connection is dropped when answering. This is the combined debug log:

<-- SIP read from 129.125.71.172:5060: INVITE sip:602@129.125.21.241 SIP/2.0 Via: SIP/2.0/UDP 129.125.71.172:5060;rport;branch=z9hG4bK608ACE08141F5E878FB9EED02E9157D0 From: "Anneke X-lite@wingtip112" <sip:501@129.125.21.241>;tag=738462257 To: <sip:602@129.125.21.241> Contact: <sip:501@129.125.71.172:5060> Call-ID: 443E55E1-9961-48C7-B5AA-D58BA21F6B66@129.125.71.172 CSeq: 18831 INVITE Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-Lite release 1105d Content-Length: 310

v=0 o=501 232538645 232538662 IN IP4 129.125.71.172 s=X-Lite c=IN IP4 129.125.71.172 t=0 0 m=audio 8000 RTP/AVP 0 8 3 98 97 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv

--- (11 headers 14 lines)--- Using INVITE request as basis request - 443E55E1-9961-48C7-B5AA-D58BA21F6B66@129.125.71.172 Sending to 129.125.71.172 : 5060 (NAT) Reliably Transmitting (no NAT) to 129.125.71.172:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 129.125.71.172:5060;rport;branch=z9hG4bK608ACE08141F5E878FB9EED02E9157D0;received=129.125.71.172 From: "Anneke X-lite@wingtip112" <sip:501@129.125.21.241>;tag=738462257 To: <sip:602@129.125.21.241>;tag=as7d1966a3 Call-ID: 443E55E1-9961-48C7-B5AA-D58BA21F6B66@129.125.71.172 CSeq: 18831 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:602@129.125.21.241> Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4333ac79" Content-Length: 0

--- Scheduling destruction of call '443E55E1-9961-48C7-B5AA-D58BA21F6B66@129.125.71.172' in 15000 ms Found user '501'

<-- SIP read from 129.125.71.172:5060: ACK sip:602@129.125.21.241 SIP/2.0 Via: SIP/2.0/UDP 129.125.71.172:5060;rport;branch=z9hG4bK608ACE08141F5E878FB9EED02E9157D0 From: "Anneke X-lite@wingtip112" <sip:501@129.125.21.241>;tag=738462257 To: <sip:602@129.125.21.241>;tag=as7d1966a3 Contact: <sip:501@129.125.71.172:5060> Call-ID: 443E55E1-9961-48C7-B5AA-D58BA21F6B66@129.125.71.172 CSeq: 18831 ACK Max-Forwards: 70 Content-Length: 0

--- (9 headers 0 lines)---

<-- SIP read from 129.125.71.172:5060: INVITE sip:602@129.125.21.241 SIP/2.0 Via: SIP/2.0/UDP 129.125.71.172:5060;rport;branch=z9hG4bK54C0A2207F8FC2C3D514805AC6CF3749 From: "Anneke X-lite@wingtip112" <sip:501@129.125.21.241>;tag=738462257 To: <sip:602@129.125.21.241> Contact: <sip:501@129.125.71.172:5060> Call-ID: 443E55E1-9961-48C7-B5AA-D58BA21F6B66@129.125.71.172 CSeq: 18832 INVITE Proxy-Authorization: Digest username="501",realm="asterisk",nonce="4333ac79",response="68a8dd8168ca34280e2fb56aceb8cdc9",uri="sip:602@129.125.21.241",algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-Lite release 1105d Content-Length: 310

v=0 o=501 232538645 232538662 IN IP4 129.125.71.172 s=X-Lite c=IN IP4 129.125.71.172 t=0 0 m=audio 8000 RTP/AVP 0 8 3 98 97 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv

--- (12 headers 14 lines)--- Using INVITE request as basis request - 443E55E1-9961-48C7-B5AA-D58BA21F6B66@129.125.71.172 Sending to 129.125.71.172 : 5060 (NAT) Found user '501' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 98 Found RTP audio format 97 Found RTP audio format 101 Peer audio RTP is at port 129.125.71.172:8000 Found description format pcmu Found description format pcma Found description format gsm Found description format iLBC Found description format speex Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x60e (gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 602 in from-internal (domain 129.125.21.241) list_route: hop: <sip:501@129.125.71.172:5060> Transmitting (no NAT) to 129.125.71.172:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 129.125.71.172:5060;rport;branch=z9hG4bK54C0A2207F8FC2C3D514805AC6CF3749;received=129.125.71.172 From: "Anneke X-lite@wingtip112" <sip:501@129.125.21.241>;tag=738462257 To: <sip:602@129.125.21.241> Call-ID: 443E55E1-9961-48C7-B5AA-D58BA21F6B66@129.125.71.172 CSeq: 18832 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:602@129.125.21.241> Content-Length: 0

---

   -- Executing Macro("SIP/501-09b4e060", "exten-vm|novm|602") in new stack
   -- Executing Macro("SIP/501-09b4e060", "user-callerid") in new stack
   -- Executing GotoIf("SIP/501-09b4e060", "0?report") in new stack
   -- Executing GotoIf("SIP/501-09b4e060", "0?start") in new stack
   -- Executing Set("SIP/501-09b4e060", "REALCALLERIDNUM=501") in new stack
   -- Executing NoOp("SIP/501-09b4e060", "REALCALLERIDNUM is 501") in new stack
   -- Executing Set("SIP/501-09b4e060", "AMPUSER=501") in new stack
   -- Executing Set("SIP/501-09b4e060", "AMPUSERCIDNAME=Anneke SIP freePBX") in new stack
   -- Executing GotoIf("SIP/501-09b4e060", "0?report") in new stack
   -- Executing Set("SIP/501-09b4e060", "CALLERID(all)=Anneke SIP freePBX <501>") in new stack
   -- Executing NoOp("SIP/501-09b4e060", "Using CallerID "Anneke SIP freePBX" <501>") in new stack
   -- Executing Set("SIP/501-09b4e060", "FROMCONTEXT=exten-vm") in new stack
   -- Executing Set("SIP/501-09b4e060", "VMBOX=novm") in new stack
   -- Executing Set("SIP/501-09b4e060", "EXTTOCALL=602") in new stack
   -- Executing Set("SIP/501-09b4e060", "CFUEXT=") in new stack
   -- Executing Set("SIP/501-09b4e060", "RT=") in new stack
   -- Executing Macro("SIP/501-09b4e060", "record-enable|602|IN") in new stack
   -- Executing GotoIf("SIP/501-09b4e060", "0 > 0?2:4") in new stack
   -- Goto (macro-record-enable,s,4)
   -- Executing AGI("SIP/501-09b4e060", "recordingcheck|20070301-155204|1172760724.6") in new stack
   -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck

PHP Warning: Unknown(): Unable to load dynamic library '/usr/lib/php4/imap.so' - libc-client.so.0: cannot open shared object file: No such file or directory in Unknown on line 0

 recordingcheck|20070301-155204|1172760724.6: Inbound recording not enabled
   -- AGI Script recordingcheck completed, returning 0
   -- Executing NoOp("SIP/501-09b4e060", "No recording needed") in new stack
   -- Executing GotoIf("SIP/501-09b4e060", "0?dolocaldial|1") in new stack
   -- Executing Macro("SIP/501-09b4e060", "dial||tr|602") in new stack
   -- Executing AGI("SIP/501-09b4e060", "dialparties.agi") in new stack
   -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi

PHP Warning: Unknown(): Unable to load dynamic library '/usr/lib/php4/imap.so' - libc-client.so.0: cannot open shared object file: No such file or directory in Unknown on line 0

 dialparties.agi: Starting New Dialparties.agi
   --  dialparties.agi: priority is 1
 dialparties.agi: Caller ID name is 'Anneke SIP freePBX' number is '501'
 dialparties.agi: Methodology of ring is  'none'
   --  dialparties.agi: Added extension 602 to extension map
   --  dialparties.agi: Extension 602 cf is disabled
   --  dialparties.agi: Extension 602 do not disturb is disabled
 == Parsing '/etc/asterisk/manager.conf': Found
 == Parsing '/etc/asterisk/manager_custom.conf': Found
 == Manager 'admin' logged on from 127.0.0.1
 == Manager 'admin' logged off from 127.0.0.1
   --  dialparties.agi: Checking CW and CFB status for extension 602
   --  dialparties.agi: DbSet CALLTRACE/602 to 501
   -- AGI Script dialparties.agi completed, returning 0
   -- Executing Dial("SIP/501-09b4e060", "OOH323/602||tr") in new stack

--- ooh323_request - data 602 format 0x4 (ulaw) --- find_peer +++ find_peer +++ ooh323_request --- ooh323_call- 602 +++ ooh323_call

   -- Called 602

Transmitting (no NAT) to 129.125.71.172:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 129.125.71.172:5060;rport;branch=z9hG4bK54C0A2207F8FC2C3D514805AC6CF3749;received=129.125.71.172 From: "Anneke X-lite@wingtip112" <sip:501@129.125.21.241>;tag=738462257 To: <sip:602@129.125.21.241>;tag=as4303820b Call-ID: 443E55E1-9961-48C7-B5AA-D58BA21F6B66@129.125.71.172 CSeq: 18832 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:602@129.125.21.241> Content-Length: 0

--- --- onNewCallCreated ooh323c_o_4 --- find_call +++ find_call setting callid number 501

Outgoing call 602(ooh323c_o_4) - Codec prefs - ()
       Adding capabilities to call(outgoing, ooh323c_o_4)

--- configure_local_rtp +++ configure_local_rtp +++ onNewCallCreated ooh323c_o_4 --- onAlerting ooh323c_o_4 --- find_call +++ find_call +++ onAlerting ooh323c_o_4

   -- OOH323/602-bf43 is ringing

<-- SIP read from 129.125.71.172:5060:

--- (0 headers 0 lines) Nat keepalive --- --- onCallEstablished ooh323c_o_4 --- find_call +++ find_call +++ onCallEstablished ooh323c_o_4

   -- OOH323/602-bf43 answered SIP/501-09b4e060

We're at 129.125.21.241 port 17612 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 129.125.71.172:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 129.125.71.172:5060;rport;branch=z9hG4bK54C0A2207F8FC2C3D514805AC6CF3749;received=129.125.71.172 From: "Anneke X-lite@wingtip112" <sip:501@129.125.21.241>;tag=738462257 To: <sip:602@129.125.21.241>;tag=as4303820b Call-ID: 443E55E1-9961-48C7-B5AA-D58BA21F6B66@129.125.71.172 CSeq: 18832 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:602@129.125.21.241> Content-Type: application/sdp Content-Length: 244

v=0 o=root 11555 11555 IN IP4 129.125.21.241 s=session c=IN IP4 129.125.21.241 t=0 0 m=audio 17612 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - -

---

<-- SIP read from 129.125.71.172:5060: ACK sip:602@129.125.21.241 SIP/2.0 Via: SIP/2.0/UDP 129.125.71.172:5060;rport;branch=z9hG4bK2453C8D95BA5E050AEB9D10880A040C9 From: "Anneke X-lite@wingtip112" <sip:501@129.125.21.241>;tag=738462257 To: <sip:602@129.125.21.241>;tag=as4303820b Contact: <sip:501@129.125.71.172:5060> Call-ID: 443E55E1-9961-48C7-B5AA-D58BA21F6B66@129.125.71.172 CSeq: 18832 ACK Max-Forwards: 70 Content-Length: 0

--- (9 headers 0 lines)--- --- onCallCleared ooh323c_o_4 --- find_call +++ find_call --- ooh323_hangup

   hanging 602

+++ ooh323_hangup

 == Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/501-09b4e060' in macro 'dial'
 == Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/501-09b4e060' in macro 'exten-vm'
 == Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/501-09b4e060'

Scheduling destruction of call '443E55E1-9961-48C7-B5AA-D58BA21F6B66@129.125.71.172' in 32000 ms set_destination: Parsing <sip:501@129.125.71.172:5060> for address/port to send to set_destination: set destination to 129.125.71.172, port 5060 Reliably Transmitting (no NAT) to 129.125.71.172:5060: BYE sip:501@129.125.71.172:5060 SIP/2.0 Via: SIP/2.0/UDP 129.125.21.241:5060;branch=z9hG4bK02074b63 From: <sip:602@129.125.21.241>;tag=as4303820b To: "Anneke X-lite@wingtip112" <sip:501@129.125.21.241>;tag=738462257 Contact: <sip:602@129.125.21.241> Call-ID: 443E55E1-9961-48C7-B5AA-D58BA21F6B66@129.125.71.172 CSeq: 102 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0

--- --- ooh323_destroy

Destroying 602

+++ ooh323_destroy

<-- SIP read from 129.125.71.172:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 129.125.21.241:5060;branch=z9hG4bK02074b63 From: <sip:602@129.125.21.241>;tag=as4303820b To: "Anneke X-lite@wingtip112" <sip:501@129.125.21.241>;tag=738462257 Contact: <sip:501@129.125.71.172:5060> Call-ID: 443E55E1-9961-48C7-B5AA-D58BA21F6B66@129.125.71.172 CSeq: 102 BYE Server: X-Lite release 1105d Content-Length: 0

==== Common codecs among Ekiga and X-Lite ====

As recommended on the TrixBox.org fora, we try to limit the codecs set to the common ones. to do so, we register Ekiga as a SIP phone to the trixBox and make a call; the Ekiga log shows then the following:

10:07:42 Started Ekiga 2.0.3 for user fabio
10:07:47 Registered to 129.125.21.241
10:07:57 Call from Anneke SIP freePBX
10:07:59 Answering incoming call
10:08:00 Opened Intel ICH5 for recording with plugin ALSA
10:08:00 Opened codec PCMU for transmission
10:08:00 Opened Intel ICH5 for playing with plugin ALSA
10:08:00 Opened codec PCMU for reception
10:08:00 Connected with Anneke SIP freePBX using Asterisk PBX
10:08:10 Closed codec PCMU which was opened for transmission
10:08:10 Closed codec PCMU which was opened for reception
10:08:11 Local user cleared the call

The X-Lite unluckily doesn't report any codec information.

From this log we can conclude that the used codec is the Ekiga's PCMU, which is likely to be X-Lite's G711u