Problem in sip call transfer

Asterisk Asterisk Support
1

Hi

i want to try call transfer.
i have established a call b/w to sip clients via asterisk.

now when i dial ‘#’ from one of the clients, asterisk returns 415 unsupported:

console logs:

[quote]INFO sip:919811098110@172.16.105.35 SIP/2.0
Via: SIP/2.0/UDP 10.203.154.137:7062;branch=z9hG4bK660580176-379
Max-Forwards: 70
P-Access-Network-Info: 3GPP-GERAN;cgi-3gpp=123456CAFEFACE
Accept: application/simple-message-summary
Accept: application/sdp
From: sip:919891300300@10.203.154.137:7062;tag=ICF_647408176-378
To: "user"sip:919811098110@172.16.105.35;tag=as4078fdcb
Call-ID: 481d99226edf324a1980d12d38fe5f93@172.16.105.35
CSeq: 2 INFO
Contact: usersip:919891300300@10.203.154.137:7062
Content-Type: application/dtmf
Allow: INVITE,ACK,CANCEL,BYE,OPTIONS
Content-Length: 25

Signal=35
Duration=160

<------------->
— (14 headers 2 lines) —
Receiving INFO!
[Jan 14 17:19:39] WARNING[30863]: chan_sip.c:11130 handle_request_info: Unable to parse INFO message from 481d99226edf324a1980d12d38fe5f93@172.16.105.35. Content

<— Transmitting (no NAT) to 10.203.154.137:7062 —>
SIP/2.0 415 Unsupported media type
Via: SIP/2.0/UDP 10.203.154.137:7062;branch=z9hG4bK660580176-379;received=10.203.154.137
From: sip:919891300300@10.203.154.137:7062;tag=ICF_647408176-378
To: "user"sip:919811098110@172.16.105.35;tag=as4078fdcb
Call-ID: 481d99226edf324a1980d12d38fe5f93@172.16.105.35
CSeq: 2 INFO
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

[/quote]

extensions.conf:

[quote]
exten => 919891300300,1,Verbose(1|Extension 919891300300)
exten => 919891300300,n,Dial(SIP/a@10.203.154.137:9060,20,rt)
exten => 919891300300,n,Hangup()[/quote]

2

Seems the client you’re using send the # dtmf using the INFO method in a way not understood by Asterisk, try use rfc2833 as the method to transfer the dtmfs, in Asterisk and in the client.

Cheers.

Marco Bruni

3

I have also set dtmfmode=info in sip.conf, but still it does not work

INFO sip:919811098110@172.16.105.35 SIP/2.0
Via: SIP/2.0/UDP 10.203.154.137:8060;branch=z9hG4bK941265256-2787
Max-Forwards: 70
From: usersip:919811098110@ssf.com:8060;tag=ICF_939547256-2785
To: sip:919811098110@172.16.105.35;tag=as08ea273b
Call-ID: 939547256-2783
CSeq: 2 INFO
Contact: usersip:919811098110@10.203.154.137:8060
P-Access-Network-Info: 3GPP-GERAN;cgi-3gpp=123456CAFEFACE
P-Preferred-Identity: sip:919811098110@ssf.com
Content-Type: application/dtmf
Allow: INVITE,ACK,CANCEL,BYE,OPTIONS
Content-Length: 25

Signal=35
Duration=160

sip.conf[quote]
[general]
dtmfmode=info
sendrpid = yes
relaxdtmf=yes

[919811098110]
type=friend
context=phones
host=dynamic
threewaycalling=yes
dtmfmode=info
relaxdtmf=yes

[919891300300]
type=friend
host=dynamic
context=phones
threewaycalling=yes[/quote]

extensions.conf

[quote]exten => 919811098110,1,Verbose(1|Unrouted call handler)
exten => 919811098110,n,Answer()
exten => 919811098110,n,Wait(5)
exten => 0,1,Dial(SIP/919891300300@10.203.154.137:7062,rt)
[/quote][/quote]

4

Don’t set dtmfmode=info, you’ve to set dtmfmode=rfc2833.

Cheers.

Marco Bruni

5

I also have similar problem. I try to create a voice menu and then call from a normal phone to a SIP softphone (eyeBeam). When IM asked to press extension number, everytime I press a button on the phone, this message came out:

WARNING[16400]: chan_sip.c:11152 handle_request_info: Unable to parse INFO message from AAisKX__DOsAABeyetS7ow@xener.com. Content

I set dtmfmode=rfc2833 to [general] in Asterik sip.conf file but no success. Should it be change at the DID company (which provide conversion between VoIP call and PSTN call)?

Please help me, I am quite new with Asterisk

6

Please, anyone knows about this. I reinstalled the entire system and still have this problem. So desperate now…