Hi,
I am connecting to my own * server in the same LAN using a java client. The conection is working. I tried adding some functionality to send DTMF by SIP INFO but the server seems to not understand it. I get this:
– Playing ‘vm-helpexit’ (language ‘en’)
sip debug
SIP Debugging enabled
*CLI>
<-- SIP read from 192.168.0.83:9552:
INFO sip:192.168.0.117:5060;transport=udp SIP/2.0
Call-ID: e0b44a5765c4285ae821b76ade69c477@192.168.0.83
CSeq: 1 INFO
From: sip:mcr@192.168.0.117;transport=udp;tag=8143
To: sip:sip:111@192.168.0.117;transport=udp
Via: SIP/2.0/UDP 192.168.0.83:9552;branch=885592977
Max-Forwards: 70
Contact: sip:mcr@192.168.0.83:9552;transport=udp
Content-Type: application/dtmf-relay
Content-Length: 21
Signal=1
Duration=170
— (10 headers 2 lines)—
Transmitting (no NAT) to 192.168.0.83:9552:
SIP/2.0 503 Server error
Via: SIP/2.0/UDP 192.168.0.83:9552;branch=885592977;received=192.168.0.83
From: sip:mcr@192.168.0.117;transport=udp;tag=8143
To: sip:sip:111@192.168.0.117;transport=udp;tag=as0f15c3da
Call-ID: e0b44a5765c4285ae821b76ade69c477@192.168.0.83
CSeq: 1 INFO
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:111@192.168.0.117
Content-Length: 0
– Playing ‘vm-onefor’ (language ‘en’)
I couldn’t help noticing that I have no INFO in the Allow: header. Is Asterisk capable of handling INFO messages? How shall I make it do so? Can you shed some light on this issue?
in sip.conf I have:
[mcr]
type=friend
secret=mcr
context=from-sip
host=dynamic
;nat=no
canreinvite=no
dtmfmode=info
;call-limit=1
;mailbox=1234@default
disallow=all
allow=alaw
allow=ulaw
allow=gsm
;allow=g723.1
I have also tried connecting with Linphone, Twinkle and kphone. All connect ok. Linphone has the possibility to send DTMF as INFO but it does not work. I see no indication in the Allow headers that Asterisk will be able to process INFO…
Thank you,
Mircea
I use INFO to talk to my Sipura SPA 3000. Was necessary to get the DTMF working properly. It works fine.
p
I tried to send INFO DTMF from the CLI and it worked. So * knows what is that type of INFO. But when I send from my application then I get a 503. Can you send me a SIP dump from the CLI for a dialog in which you transmit DTMF from your phone, please?
Thank you.
Mircea
Here I have my message and further down is the one I sent from CLI:
INFO sip:111@192.168.0.117:5060;transport=udp SIP/2.0
Call-ID: 30c3b9216f301679109e224a12d69fc2@192.168.0.83
CSeq: 1 INFO
From: sip:mcr@192.168.0.117;transport=udp;tag=2731
To: sip:111@192.168.0.117;transport=udp
Via: SIP/2.0/UDP 192.168.0.83:2285;branch=-1019483072
Max-Forwards: 70
Contact: sip:mcr@192.168.0.83:2285;transport=udp
Content-Type: application/dtmf-relay
Content-Length: 23
Signal=1
Duration=270
INFO sip:mcr@192.168.0.83:5987;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.117:5060;branch=z9hG4bK7fdb5d55;rport
From: “asterisk” sip:asterisk@192.168.0.117;tag=as2abdec9e
To: sip:mcr@192.168.0.83:5987;transport=udp
Contact: sip:asterisk@192.168.0.117
Call-ID: 6170e68f645a187e1b7cdebe6ecd3c8c@192.168.0.117
CSeq: 126 INFO
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Type: application/dtmf-relay
Content-Length: 24
Signal=1
Duration=250
Hi,
I have fixed a problem regarding sip dialogs and now the INFO message seems to get to the destination ok. The problem is the server answers ok but doesn’t seem to do anything… Any ideea what the problem might be?
Thank you
Mircea
– Playing ‘vm-opts’ (language ‘en’)
<-- SIP read from 192.168.0.83:4459:
INFO sip:111@192.168.0.117:5060;transport=udp SIP/2.0
Call-ID: 9c8f601ad3b1716689df84bc19d3ccbb@192.168.0.83
CSeq: 2 INFO
From: sip:mcr@192.168.0.117;transport=udp;tag=2714
To: sip:111@192.168.0.117;transport=udp
Via: SIP/2.0/UDP 192.168.0.83:4459;branch=z9hG4bKdf81f28bcc117bb9c26ff66a7cfa792b
Max-Forwards: 70
Contact: sip:mcr@192.168.0.83:4459;transport=udp
Content-Type: application/dtmf-relay
Content-Length: 23
Signal=1
Duration=270
— (10 headers 3 lines)—
Transmitting (no NAT) to 192.168.0.83:4459:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.83:4459;branch=z9hG4bKdf81f28bcc117bb9c26ff66a7cfa792b;received=192.168.0.83
From: sip:mcr@192.168.0.117;transport=udp;tag=2714
To: sip:111@192.168.0.117;transport=udp;tag=as30b6bf37
Call-ID: 9c8f601ad3b1716689df84bc19d3ccbb@192.168.0.83
CSeq: 2 INFO
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:111@192.168.0.117
Content-Length: 0
-- Playing 'vm-helpexit' (language 'en')
-- Playing 'vm-onefor' (language 'en')
Found the problem. The application was retrieving the wrong sip dialog in order to perform the operation.