Ringing tone

I have asterisk with h323 trunk. When I dial from PSTN phone through h323 trunk to sip clients, not hear ringing tone (tooo,tooo,tooo). The SIP phone rings. extension is:

exten => 0276,1,Progress
exten => 0276,2,Ringing
exten => 0276,3,Dial(SIP/111,r)

When I changed extension :

exten => 0276,1,Progress
exten => 0276,2,Ringing
exten => 0276,3,Playback(some file)
exten => 0276,4,Dial(SIP/111,r)

I hear now playback and after that ringing tone too.
How can I hear ringing without play some file?
Plz help
regards kajana.

Hi

try adding progressinband=yes in the general section of the sip.conf

or
exten => 0276,1,Answer
exten => 0276,2,Dial(SIP/111,r)

Ian

With progressinband=yes in sip.conf not works, but width exten => 0276,1,Answer works. Thanks a lot.

I have billing on asterisk and with exten => 0276,1,Answer, it beginns count seconds if user not answers. Isn’t other way to hear ringing? :frowning:
(I noticed that with oh323 driver it works fine.)
regards kajana.

hi
I changed H323 trunk with SIP but still the same problem.
this is my dialplan:
[default]
exten => 55,1,Progress
exten => 55,2,Dial(SIP/55)
sip debug :
<-- SIP read from XXX :smile: (myprovider):5061:
INVITE sip:55@10.16.0.34SIP/2.0
Via: SIP/2.0/UDP
XXX:5061;rport;branch=z9hG4bK-6665340-279391521-469762176-4122611908
From:
sip:58147750@XXX:5061;tag=6664900-279391521-469762176-4122611908
To: sip:55@10.16.0.34
Call-ID: e2b26500212da7108000001cc410baf5@XXX
CSeq: 1 INVITE
Contact: sip:58147750@XXX:5061
Max-Forwards: 10
User-Agent: MERA MSIP v.3.0
Cisco-Guid: 1581549417-3529773532-2196674106-2860857446
P-Asserted-Identity: sip:58147750@XXX:5061
Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, REGISTER
Content-Type: application/sdp
Content-Length: 211

v=0
o=- 1202138462 1202138462 IN IP4 XXX
s=-
c=IN IP4 XXX
t=0 0
m=audio 10916 RTP/AVP 8 0 96
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15

— (14 headers 10 lines) —
Using INVITE request as basis request -
e2b26500212da7108000001cc410baf5@XXX
Sending to XXX : 5061 (NAT)
Found no matching peer or user for XXX:5061’
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 96
Peer audio RTP is at port XXX:10916
Found description format PCMA
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer -
audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc
(ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Looking for 55 in default (domain 10.16.0.34)
list_route: hop: sip:58147750@XXX:5061
Transmitting (NAT) to XXX:5061:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
XXX:5061;branch=z9hG4bK-6665340-279391521-469762176-4122611908;received=XXX;rport=5061
From:
sip:58147750@XXX:5061;tag=6664900-279391521-469762176-4122611908
To: sip:55@10.16.0.34
Call-ID: e2b26500212da7108000001cc410baf5@XXX
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:55@10.16.0.34
Content-Length: 0
— (14 headers 10 lines) —
Using INVITE request as basis request -
e2b26500212da7108000001cc410baf5@XXX
Sending to XXX : 5061 (NAT)
Found no matching peer or user for XXX:5061’
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 96
Peer audio RTP is at port XXX:10916
Found description format PCMA
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer -
audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc
(ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Looking for 55 in default (domain 10.16.0.34)
list_route: hop: sip:58147750@XXX:5061
Transmitting (NAT) to XXX:5061:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
XXX:5061;branch=z9hG4bK-6665340-279391521-469762176-4122611908;received=XXX;rport=5061
From:
sip:58147750@XXX:5061;tag=6664900-279391521-469762176-4122611908
To: sip:55@10.16.0.34
Call-ID: e2b26500212da7108000001cc410baf5@XXX
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:55@10.16.0.34
Content-Length: 0


-- Executing Progress("SIP/5061-08186f08", "") in new stack

We’re at 10.16.0.34 port 14914
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Transmitting (NAT) to XXX:5061:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP
XX:5061;branch=z9hG4bK-6665340-279391521-469762176-4122611908;received=XXX;rport=5061
From:
sip:58147750@XXX:5061;tag=6664900-279391521-469762176-4122611908
To: sip:55@10.16.0.34;tag=as65295176
Call-ID: e2b26500212da7108000001cc410baf5@XXX
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:55@10.16.0.34
Content-Type: application/sdp
Content-Length: 233

v=0
o=root 29507 29507 IN IP4 10.16.0.34
s=session
c=IN IP4 10.16.0.34
t=0 0
m=audio 14914 RTP/AVP 0 8 96
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=silenceSupp:off - - - -


-- Executing Dial("SIP/5061-08186f08", "SIP/111") in new stack

We’re at 10.16.0.34 port 16626
Adding codec 0x4 (ulaw) to SDP
13 headers, 8 lines
Reliably Transmitting (no NAT) to 192.168.240.2 :smile: (my sip phone) :smile: :5060:
INVITE sip:111@192.168.240.2:5060 SIP/2.0
Via: SIP/2.0/UDP 10.16.0.34:5060;branch=z9hG4bK49ce176d;rport
From: “58147750” sip:58147750@10.16.0.34;tag=as0d328e59
To: sip:111@192.168.240.2:5060
Contact: sip:58147750@10.16.0.34
Call-ID: 739c6d440e49cca27e95475b7a68fc3c@10.16.0.34
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 04 Feb 2008 15:29:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 156

v=0
o=root 29507 29507 IN IP4 10.16.0.34
s=session
c=IN IP4 10.16.0.34
t=0 0
m=audio 16626 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -


-- Called 111

<-- SIP read from 192.168.240.2:5060:
SIP/2.0 100 Trying
To: sip:111@192.168.240.2:5060
From: “58147750” sip:58147750@10.16.0.34;tag=as0d328e59
Call-ID: 739c6d440e49cca27e95475b7a68fc3c@10.16.0.34
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.16.0.34:5060;branch=z9hG4bK49ce176d
Server: Linksys/PAP2T-3.1.15(LS)
Content-Length: 0

— (8 headers 0 lines) —

<-- SIP read from 192.168.240.2:5060:
SIP/2.0 180 Ringing
To: sip:111@192.168.240.2:5060;tag=370be070900112a7i0
From: “58147750” sip:58147750@10.16.0.34;tag=as0d328e59
Call-ID: 739c6d440e49cca27e95475b7a68fc3c@10.16.0.34
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.16.0.34:5060;branch=z9hG4bK49ce176d
Server: Linksys/PAP2T-3.1.15(LS)
Content-Length: 0

— (8 headers 0 lines) —
– SIP/111-081c6088 is ringing
Transmitting (NAT) to XXX:5061:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP
XXX:5061;branch=z9hG4bK-6665340-279391521-469762176-4122611908;received=XXX;rport=5061
From:
sip:58147750@XXX:5061;tag=6664900-279391521-469762176-4122611908
To: sip:55@10.16.0.34;tag=as65295176
Call-ID: e2b26500212da7108000001cc410baf5@XXX
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:55@10.16.0.34
Content-Length: 0


<-- SIP read from XXX:5061:
CANCEL sip:55@10.16.0.34 SIP/2.0
Via: SIP/2.0/UDP
XXX:5061;rport;branch=z9hG4bK-6665340-279391521-469762176-4122611908
From:
sip:58147750@XXX:5061;tag=6664900-279391521-469762176-4122611908
To: sip:55@10.16.0.34
Call-ID: e2b26500212da7108000001cc410baf5@XXX
CSeq: 1 CANCEL
Max-Forwards: 10
User-Agent: MERA MSIP v.3.0
Reason: Q.850;cause=16;text="Normal call clearing"
Content-Length: 0

can anyone help me?
regards kajana.