[Asterisk-video] Problems with a loopback scenario, initial negotiation fails.

Thu Jan 8 09:37:31 CST 2009

Hi, I'm having problems establishing an h324m call. We don't have access
yet to an external PRI connection so we bought a Digium TE205P with 2
PRI and connected them with a crossed cable to make a loopback (one acts
like a net and the other as a cpe). 

My idea is to make a SIP call to asterisk, forward it with h324m_call
through the span 1, receive it with h324m_gw from the span 2 and then
make a new SIP call to another phone (or use the h324m_loopback). The
problem is that the negotiation stays in the state SETUP and thus the
initial SIP phone stays in "calling". I've think the problem might be
either in the configuration of the PRI loopback, in my selection of
install steps or in the weird dialplan, so i include some of the things
I've tried to see if I'm missing something.

Selection of install steps:

*I've decided to use Dahdi instead of Zap (haven't seen any post on this
but I don't think it should be a problem). I use the version 2.1.0.3

*At first I've tried with Asterisk 1.6.0.1, but there were 3 problems:

    1. I had to manually relink app_h324m.so because of Makefile
differences with version 1.4.X

    2. Had to comment line 1560 of app_h324m.c
"ast_cli_register(&cli_debug);" because it causes a segfault at loading.

    3. Had to modify AMR code in several places to compile.

* Then I've changed to Asterisk 1.4.22 and those problems went away.

* I'm using latest version from svn, revision 241. Don't know how stable
this is.

* I'm using libpri-1.4.8 without patching with
http://bugs.digium.com/view.php?id=10217 (I think this could be a
problem, but I haven't seen a tutorial that says the patch its required)

Configuration of the PRI loopback.

I include at the bottom of the mail the configuration im currently
using. In general all examples follow the same logic. With the prefix
201 i can make a voice call through the span 1, with the prefix 301 I
create an h324 pseudo channel and then go through the span 1. All
incoming calls from the span 2 are answered with different dialplans in
the context [from-pstn-cpe], some of them expect normal voice, and other
expects h324m. I have two SIP phones connected to asterisk, 402 is a
physical and 403 is a soft phone.

* If from 402 I call 2011234, the loopback works and I can hear 1, 2, 3,
4.

* If from 403 I call 201402 or if from 402 I call 201403 it works two,
but at some point I had audio problems between in the first scenario
because of alaw and ulaw. 

* From any SIP phone I call 3016XX it does not work, the SIP phone stays
on the state "calling" and from what I've seen debugging the app, both
h324m_call and h324m_gw stays in state 1 (SETUP).

* Strange things I've noticed in the loopback looking at core show
channel/s are:

    * Different law handling, I couldn't hardcore chan_dahdi.c to make
it always ulaw.

    * TRANSFERCAPABILITY=SPEECH, I don't know exactly what this is, but
couldn't make both DIGITAL, just the receiving end.

    * Using dahdi_monitor I could dump the pri channel, but couldn't
find any sequence i would recognize from the mailing list.

---------------------- Configuration files---------------------

/etc/asterisk/extensions.conf

[from-sip]
exten => 402,1,Dial(SIP/402)
exten => 403,1,Dial(SIP/403)

;;;;exten => _201.,1,Set(CHANNEL(transfercapability)=DIGITAL)
;;;;exten => _201.,n,Dial(DAHDI/1/${EXTEN:3})

exten => _201.,1,Dial(DAHDI/1/${EXTEN:3})

exten => _301.,1,h324m_call(201${EXTEN:3}@from-internal-custom)

[from-pstn-cpe]
exten => 1234,1,Answer()
exten => 1234,n,SayDigits(${EXTEN})

exten => 402,1,Dial(SIP/402)
exten => 403,1,Dial(SIP/403)

exten => 666,1,Answer()
exten => 666,n,h324m_loopback(v)

exten => 667,1,h324m_gw_answer()
exten => 667,n,h324m_loopback()

exten => 668,1,h324m_gw_answer()
exten => 668,n,Playback(tt-monkeys)

exten => 669,1,Set(CHANNEL(transfercapability)=DIGITAL)
exten => 669,n,h324m_gw(migw at from-pstn-cpe)

exten => 670,1,Answer()
exten => 670,n,h324m_gw(migw at from-pstn-cpe)

exten => 671,1,Answer()
exten => 671,n,h324m_gw(migw2 at from-pstn-cpe <mailto:migw2 at from-pstn-cpe>
)

exten => migw,1,h324m_gw_answer()
exten => migw,2,Echo()

exten => migw2,1,h324m_gw_answer()
exten => migw2,2,Dial(SIP/402)

----------------------------------------------------------

/etc/dahdi/system.conf

span=1,0,0,ccs,hdb3,crc4
# termtype: te
bchan=1-15,17-31
dchan=16
mulaw=1-15
mulaw=17-31
echocanceller=mg2,1-15,17-31

# Span 2: TE2/0/2 "T2XXP (PCI) Card 0 Span 2"
span=2,1,0,ccs,hdb3,crc4
# termtype: te
bchan=32-46,48-62
dchan=47
mulaw=32-46
mulaw=48-62
echocanceller=mg2,32-46,48-62

# Global data

loadzone        = us
defaultzone     = us

----------------------------------------------------------------

/etc/asterisk/chan_dahdi.conf

[channels]

; Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" (MASTER)
group=11
context=from-pstn-net
switchtype = euroisdn
signalling = pri_net
transfer=yes
;threewaycalling=yes
;cancallforward=yes
facilityenable = yes
channel => 1-15,17-31

; Span 2: TE2/0/2 "T2XXP (PCI) Card 0 Span 2"
group=12
context=from-pstn-cpe
switchtype = euroisdn
signalling = pri_cpe
transfer=yes
;threewaycalling=yes
;cancallforward=yes
facilityenable = yes
channel => 32-46,48-62

----------------------------------------------------------------

/etc/asterisk/sip.conf

[general]

context=default

allowoverlap=no

bindport=5060           

bindaddr=0.0.0.0        

srvlookup=yes           

videosupport=yes

disable=all
allow=ulaw
allow=h263
allow=h263p

[402]
type=friend
qualify=no
port=5060
nat=never
host=dynamic
dtmfmode=rfc2833
context=from-internal-custom
canreinvite=yes
callerid="zultys 402" <402>

[403]
type=friend
secret=403
qualify=no
port=5060
nat=never
host=dynamic
dtmfmode=rfc2833
context=from-internal-custom
canreinvite=yes
callerid="SOFT PHONE 403" <403>

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