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Francesco Peeters

2009-Sep-02 23:18 UTC

[asterisk-users] Voipbuster not ringing, other SIP peers are ringing...

Does anybody else see the same behavior for VoipBuster connections?

When I trace one of the other SIP peers, I see it sends this message:
----------------------------------------------------------------------
<--- SIP read from 82.101.62.99:5060 --->
SIP/2.0 180 Ringing
Allow: INVITE,ACK,BYE,CANCEL,PRACK,SUBSCRIBE,NOTIFY,UPDATE
Call-ID: 740540ee64fa957513ce89f03ef5e3f2 at sip.xs4all.nl
Contact: <sip:82.101.62.99:5060>
Content-Type: application/sdp
CSeq: 103 INVITE
From: "**********" <sip:**********@sip.xs4all.nl>;tag=as70e84199
Record-Route:
<sip:82.101.62.115;lr;r2=on;ftag=as70e84199>,<sip:82.101.63.5;lr;r2=on;ftag=as70e84199>
Server: Cirpack/v4.41b (gw_sip)
To: <sip:0031********@sip.xs4all.nl>;tag=00-08168-044b6f36-245cd72c7
Via: SIP/2.0/UDP
***.***.***.***:5060;received=***.***.***.***;rport=5060;branch=z9hG4bK07c2ed92
Content-Length: 182

v=0
o=cp10 125193221174 125193221174 IN IP4 82.101.62.66
s=SIP Call
c=IN IP4 194.109.8.2
t=0 0
m=audio 36984 RTP/AVP 8
b=AS:64
a=rtpmap:8 PCMA/8000/1
a=ptime:20
a=sendrecv

<------------->
--- (12 headers 10 lines) ---
Found RTP audio format 8
Peer audio RTP is at port 194.109.8.2:36984
Found audio description format PCMA for ID 8
Capabilities: us - 0x10a (gsm|alaw|g729), peer - audio=0x8
(alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing),
combined - 0x0 (nothing)
Peer audio RTP is at port 194.109.8.2:36984
    -- SIP/*********-089ca9b8 is ringing
    -- SIP/*********-089ca9b8 is making progress passing it to
IAX2/2104-2287
Scheduling destruction of SIP dialog
'740540ee64fa957513ce89f03ef5e3f2 at sip.xs4all.nl' in 6400 ms (Method:
INVITE)
Reliably Transmitting (NAT) to 82.101.62.99:5060:
CANCEL sip:0031*********@sip.xs4all.nl SIP/2.0
Via: SIP/2.0/UDP ***.***.***.***:5060;branch=z9hG4bK07c2ed92;rport
From: "**********" <sip:*********@sip.xs4all.nl>;tag=as70e84199
To: <sip:0031*********@sip.xs4all.nl>
Call-ID: 740540ee64fa957513ce89f03ef5e3f2 at sip.xs4all.nl
CSeq: 103 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

----------------------------------------------------------------------


However when I dial exactly the same from VoipBuster, I see this instead:


----------------------------------------------------------------------
<--- SIP read from 77.72.169.129:5060 --->
SIP/2.0 183 Session progress
Via: SIP/2.0/UDP 195.164.89.135:5060;branch=z9hG4bK6d7efb43;rport
From: "*********"
<sip:*********@sip.voipbuster.com>;tag=as1374705a
To: <sip:0031*********@sip.voipbuster.com>;tag=120113ac4a54a269af9e2c
Contact: sip:0031*********@77.72.169.129:5060
Call-ID: 1949e0303d52a19b1b4f91f16ff94297 at sip.voipbuster.com
CSeq: 103 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Type: application/sdp
Content-Length: 162

v=0
o=********* 1251932194 1251932194 IN IP4 194.221.62.33
s=SIP Call
c=IN IP4 194.221.62.33
t=0 0
m=audio 8958 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20

<------------->
--- (11 headers 8 lines) ---
Found RTP audio format 0
Peer audio RTP is at port 194.221.62.33:8958
Found audio description format PCMU for ID 0
Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0
(nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing),
combined - 0x0 (nothing)
Peer audio RTP is at port 194.221.62.33:8958
    -- SIP/********-089dc538 is making progress passing it to IAX2/2104-8077
  == Connect attempt from '127.0.0.1' unable to authenticate
Scheduling destruction of SIP dialog
'1949e0303d52a19b1b4f91f16ff94297 at sip.voipbuster.com' in 6400 ms
(Method: INVITE)
Reliably Transmitting (NAT) to 77.72.169.129:5060:
CANCEL sip:0031*********@sip.voipbuster.com SIP/2.0
Via: SIP/2.0/UDP ***.***.***.***:5060;branch=z9hG4bK6d7efb43;rport
From: "**********"
<sip:**********@sip.voipbuster.com>;tag=as1374705a
To: <sip:0031*********@sip.voipbuster.com>
Call-ID: 1949e0303d52a19b1b4f91f16ff94297 at sip.voipbuster.com
CSeq: 103 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
----------------------------------------------------------------------

As you can see, there are different packets being sent, and in the 2nd
case, there is no "is ringing" message, which is rather irritating...

Any suggestions would be appreciated...

TIA
-- 
FP

Francesco Peeters

2009-Sep-02 23:26 UTC

head link

[asterisk-users] Voipbuster not ringing, other SIP peers are ringing...

Francesco Peeters wrote:> Does anybody else see the same behavior for VoipBuster connections?
>
> When I trace one of the other SIP peers, I see it sends this message:
> ----------------------------------------------------------------------
> <--- SIP read from 82.101.62.99:5060 --->
> SIP/2.0 180 Ringing
> Allow: INVITE,ACK,BYE,CANCEL,PRACK,SUBSCRIBE,NOTIFY,UPDATE
> Call-ID: 740540ee64fa957513ce89f03ef5e3f2 at sip.xs4all.nl
> Contact: <sip:82.101.62.99:5060>
> Content-Type: application/sdp
> CSeq: 103 INVITE
> From: "**********"
<sip:**********@sip.xs4all.nl>;tag=as70e84199
> Record-Route:
>
<sip:82.101.62.115;lr;r2=on;ftag=as70e84199>,<sip:82.101.63.5;lr;r2=on;ftag=as70e84199>
> Server: Cirpack/v4.41b (gw_sip)
> To: <sip:0031********@sip.xs4all.nl>;tag=00-08168-044b6f36-245cd72c7
> Via: SIP/2.0/UDP
>
***.***.***.***:5060;received=***.***.***.***;rport=5060;branch=z9hG4bK07c2ed92
> Content-Length: 182
>
> v=0
> o=cp10 125193221174 125193221174 IN IP4 82.101.62.66
> s=SIP Call
> c=IN IP4 194.109.8.2
> t=0 0
> m=audio 36984 RTP/AVP 8
> b=AS:64
> a=rtpmap:8 PCMA/8000/1
> a=ptime:20
> a=sendrecv
>
> <------------->
> --- (12 headers 10 lines) ---
> Found RTP audio format 8
> Peer audio RTP is at port 194.109.8.2:36984
> Found audio description format PCMA for ID 8
> Capabilities: us - 0x10a (gsm|alaw|g729), peer - audio=0x8
> (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
> Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing),
> combined - 0x0 (nothing)
> Peer audio RTP is at port 194.109.8.2:36984
>     -- SIP/*********-089ca9b8 is ringing
>     -- SIP/*********-089ca9b8 is making progress passing it to
> IAX2/2104-2287
> Scheduling destruction of SIP dialog
> '740540ee64fa957513ce89f03ef5e3f2 at sip.xs4all.nl' in 6400 ms
(Method: INVITE)
> Reliably Transmitting (NAT) to 82.101.62.99:5060:
> CANCEL sip:0031*********@sip.xs4all.nl SIP/2.0
> Via: SIP/2.0/UDP ***.***.***.***:5060;branch=z9hG4bK07c2ed92;rport
> From: "**********"
<sip:*********@sip.xs4all.nl>;tag=as70e84199
> To: <sip:0031*********@sip.xs4all.nl>
> Call-ID: 740540ee64fa957513ce89f03ef5e3f2 at sip.xs4all.nl
> CSeq: 103 CANCEL
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Content-Length: 0
>
> ----------------------------------------------------------------------
>
>
> However when I dial exactly the same from VoipBuster, I see this instead:
>
>
> ----------------------------------------------------------------------
> <--- SIP read from 77.72.169.129:5060 --->
> SIP/2.0 183 Session progress
> Via: SIP/2.0/UDP 195.164.89.135:5060;branch=z9hG4bK6d7efb43;rport
> From: "*********"
<sip:*********@sip.voipbuster.com>;tag=as1374705a
> To: <sip:0031*********@sip.voipbuster.com>;tag=120113ac4a54a269af9e2c
> Contact: sip:0031*********@77.72.169.129:5060
> Call-ID: 1949e0303d52a19b1b4f91f16ff94297 at sip.voipbuster.com
> CSeq: 103 INVITE
> Server: (Very nice Sip Registrar/Proxy Server)
> Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
> Content-Type: application/sdp
> Content-Length: 162
>
> v=0
> o=********* 1251932194 1251932194 IN IP4 194.221.62.33
> s=SIP Call
> c=IN IP4 194.221.62.33
> t=0 0
> m=audio 8958 RTP/AVP 0
> a=rtpmap:0 PCMU/8000
> a=ptime:20
>
> <------------->
> --- (11 headers 8 lines) ---
> Found RTP audio format 0
> Peer audio RTP is at port 194.221.62.33:8958
> Found audio description format PCMU for ID 0
> Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0
> (nothing), combined - 0x4 (ulaw)
> Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing),
> combined - 0x0 (nothing)
> Peer audio RTP is at port 194.221.62.33:8958
>     -- SIP/********-089dc538 is making progress passing it to
IAX2/2104-8077
>   == Connect attempt from '127.0.0.1' unable to authenticate
> Scheduling destruction of SIP dialog
> '1949e0303d52a19b1b4f91f16ff94297 at sip.voipbuster.com' in 6400 ms
> (Method: INVITE)
> Reliably Transmitting (NAT) to 77.72.169.129:5060:
> CANCEL sip:0031*********@sip.voipbuster.com SIP/2.0
> Via: SIP/2.0/UDP ***.***.***.***:5060;branch=z9hG4bK6d7efb43;rport
> From: "**********"
<sip:**********@sip.voipbuster.com>;tag=as1374705a
> To: <sip:0031*********@sip.voipbuster.com>
> Call-ID: 1949e0303d52a19b1b4f91f16ff94297 at sip.voipbuster.com
> CSeq: 103 CANCEL
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Content-Length: 0
> ----------------------------------------------------------------------
>
> As you can see, there are different packets being sent, and in the 2nd
> case, there is no "is ringing" message, which is rather
irritating...
>
> Any suggestions would be appreciated...
>
> TIA
>   BTW: I am talking about the ringtone the caller should hear... The other
side is ringing, and calls are established just fine, but it is very
irritating to hear nothing until the call either fails or gets picked up...

-- 
FP

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