asterisk users - Nov 2003 - ISDN debugging and SIP dial-in issue]

(I have some problems with my mailing-list alias, I hope this
 doesn't get sent twice)

On Sat, Nov 15, 2003 at 04:35:20PM +0100, Philipp von Klitzing wrote:

Thank you for your comments Philipp:
> >    - with a SIP phone configured as 192.168.1.190, and with its SIP
> >      server being 192.168.1.190
> 
> That doesn't look right. Do you have another "SIP server"
installed on
> your client machine - shouldn't that rather be *, or did you - which I 
> guess - just mistype the IP? Which SIP phone are you using 
Mis-typed, yes. The SIP server is the Asterisk server and is
192.168.1.10
> (hardware/software, brand, version)?
Grandstream BudgeTone-100
   - can dial 1-800-CALL-ATT and talk with an operator through the
     sipphone.com SIP proxy, quality is adequate (changed the
     SIP server to sip01.sipphone.com of course)
   - when the SIP server is Asterisk, can be dialed from ISDN without
     any problem (maybe a slight delay), quality is good both
     directions.
   - can dial to Asterisk, in that case Asterisk's debug shows the call,
     but fails. Nothing is hearable on the BudgetTone except a busy
     tone.

Software:
   Program--1.0.3.81    Bootloader--1.0.0.7    HTML--1.0.0.18

Call examples: (this time with `sip debug' I just found about)

SIP phone dials '2'

Sip read:
INVITE sip:2@192.168.1.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.190
From: "Martial Guex"
<sip:17476691152@192.168.1.10>;tag=7adc221a-d23b-5289-93ff-261810e5291c
To: <sip:2@192.168.1.10>
Contact: <sip:17476691152@192.168.1.190>
Call-ID: b5288d54-a46c-9e16-ff7c-ec43221a71b2@192.168.1.190
CSeq: 53320 INVITE
User-Agent: Grandstream SIP UA 1.0.3.81
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Content-Length: 266

[ ... ]

Sending to 192.168.1.190 : 5060 (non-NAT)

[ ... ]

Capabilities: us - 524302, them - 285/0, combined - 12
Non-codec capabilities: us - 1, them - 0, combined - 0
DEBUG[5126]: File chan_sip.c, Line 3965 (check_user): Setting NAT on RTP to 0
Reliably Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required

[ ... ]
ACK sip:2@192.168.1.10 SIP/2.0

[ ... ]

DEBUG[5126]: File chan_sip.c, Line 565 (__sip_ack): Stopping
retransmission on 'b5288d54-a46c-9e16-ff7c-ec43221a71b2@192.168.1.190'
of Response 53320: Found

[ ... ]

DEBUG[5126]: File chan_sip.c, Line 991 (find_user): Call from user
'17476691152' is 1 out of 0
Looking for 2 in localphones

DEBUG[5126]: File chan_sip.c, Line 3369 (build_route): build_route:
Contact hop: <sip:17476691152@192.168.1.190>

    -- Executing Playback("SIP/17476691152-a52e",
"publicar-extbusy|skip") in new stack

*CLI> some time ... a few seconds
No such command 'some' (type 'help' for help)
*CLI>     -- Timeout on SIP/17476691152-a52e
  == CDR updated on SIP/17476691152-a52e
    -- Executing Hangup("SIP/17476691152-a52e", "") in new
stack
  == Spawn extension (localphones, t, 1) exited non-zero on
'SIP/17476691152-a52e'
DEBUG[15376]: File chan_sip.c, Line 1068 (sip_hangup):
find_user(17476691152) - decrement inUse counter
Reliably Transmitting (no NAT):
SIP/2.0 403 Forbidden
> >    - dial-in from ISDN, then transfer to ISDN on the secondary
channel:
> >      doesn't work (more details below)
> 
> I assume with "transfer" you mean that you are trying to
"dial out" on
> the 2nd channel. So who are you trying to call? If you are trying to call 
I call from a mobile phone to a mobile phone:

   mobile -> ISDN in -> ISDN out -> mobile

this setup works with software I developped (a modified isdn2h323
which can connect the two streams by byte-copying, plus conferencing
and control software).
> Not sure, but: You might want to look into the isdn4linux documentation 
> and use its tools like isdnlog (?) etc.
I added some printf()'s in channels/*modems*.c and the adequate AT
commands are sent, something wrong is happening but it's not Asterisk's
fault.
> If that is not it: Check your context setup: The incoming call must be  
> in a context that is allowed to dial out again.
There is no immediate error, looking like some attempt is made.
> Please provide (the relevant parts of) your extensions.conf.
[xfertomobile]
exten => s,1,Wait,1                     ; Wait a second, just for fun
exten => s,2,Answer                     ; Answer the line
exten => s,3,Background(transfer)    ; schaefer
exten => s,4,Dial,Modem/g1:079xxxxxxx|60|r
exten => s,5,Playback(extbusy,skip) ; schaefer
exten => s,6,Hangup                     ; schaefer

[localphones]
exten => s,1,Wait,1                     ; Wait a second, just for fun
exten => s,2,Answer                     ; Answer the line

exten => s,3,DigitTimeout,5             ; Set Digit Timeout to 5 seconds
exten => s,4,ResponseTimeout,10         ; Set Response Timeout to 10 seconds

exten => 1,1,Goto(demo,s,1)
exten => 2,1,Playback(extbusy,skip)
exten => 3,1,Goto(xfertomobile,s,1)

exten => t,1,Hangup
exten => i,1,Playback(invalid)          ; "That's not valid, try
again"

[default]
include => xfertomobile
> - check rtp.conf
I will need help here. Configuration on the SIP phone is local port
5004 and don't use random port.

/etc/rtp.conf:

   [general]
   rtpstart=10000
   rtpend=20000
> - any firewall (personal firewall?) or NAT in between SIP client and 
> Asterisk?
no, an Ethernet switch.
> - show us your sip.conf
[general]
port = 5060                     ; Port to bind to
bindaddr = 0.0.0.0              ; Address to bind to
context = default               ; Default for incoming calls

[17476691152]
type=friend
context=localphones
host=dynamic
secret=XXXXX
username=17476691152
dtmfmode=inband         ; Choices are inband, rfc2833, or info
qualify=1000

Try this - change 
exten => s,3,Background(transfer)    ; schaefer

to
exten => s,3,Playback(transfer)    ; schaefer

and then dial 3 from your GS.

You should also add to sip.conf for [17476691152]:
disallow=all
allow=ulaw
allow=alaw

Are you sure you need the dtmfmode=inband for the GS? I don't have a GS, 
so look for GS samples on this list. In order to prevent codec problems 
and to allow transcoding you might also want to add canreinvite=no.

Also: Check your stripmsd= setting in modem.conf to make sure you are 
really dialing the mobile number you want to dial. Next to that I'd 
rather use the "new" syntax for the Dial application like 
"Dial(Modem/g1/012345,20,rt)".

Cheers, Philipp