[Asterisk-Users] ISDN debugging and SIP dial-in issue]

[Marc SCHAEFER]

(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 at 192.168.1.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.190
From: "Martial Guex"
<sip:17476691152 at 192.168.1.10>;tag=7adc221a-d23b-5289-93ff-261810e5291c
To: <sip:2 at 192.168.1.10>
Contact: <sip:17476691152 at 192.168.1.190>
Call-ID: b5288d54-a46c-9e16-ff7c-ec43221a71b2 at 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 at 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 at 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 at 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