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Today's Topics:
1. Problem: Asterisk is ignoring INVITE Messages
(alexander.deuschlingerchello.at)
2. Re: 185 ringing / 199 trying (Malay Si)
3. Re: 185 ringing / 199 trying (mwilliam prusty)
------------------------------------------------------------
----------
Message: 1
Date: Fri, 25 Apr 2008 12:29:47 +0200
From: <alexander.deuschlingerchello.at>
Subject: [SIPForum-discussion] Problem: Asterisk is ignoring
INVITE
Messages
To: discussionsipforum.org
Message-ID: <22050687.1209119387742.JavaMail.rootviefep32>
Content-Type: text/plain; charset=utf-8
Hi!
We are experementing with SIP and Asterisk in school.
We had no big problems so far, but now we have a problem
with sending an INVITE-Message to another host, because
Asterisk ignores them.
And we cannot figure out where this failure comes from.
So, please help us to get this thing done.
Here is the SIP debug:
The Registration-Part is function successfully for the users
2000 and 2001.
INVITE-Part:
<--- SIP read from 192.168.100.50:5070 --->
INVITE sip:2000192.168.100.32 SIP/2.0
Via: SIP/2.0/UDP
192.168.100.50:5070;branch=z9hG4bK-eE6hF;rport=
From: <sip:2001192.168.100.32>
To: <sip:2000192.168.100.32>
Call-ID: eE6h192.168.100.32
CSeq: 3 INVITE
Contact: <sip:2001192.168.100.50>
User-Agent: TwinSIP Communicator
Content-Type: application/sdp
Content-Length: 99
v=0
o=andy 2634 2634 IN IP4 192.168.100.50
s=test von alex
t=0 0
m=audio 9050 udp PCM/32000/8/1
<------------->
--- (10 headers 5 lines) ---
Sending to 192.168.100.50 : 5070 (no NAT)
Using INVITE request as basis request - eE6h192.168.100.32
<--- Reliably Transmitting (no NAT) to
192.168.100.50:5070 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
192.168.100.50:5070;branch=z9hG4bK-eE6hF;rport=;received=192
.168.100.50
From: <sip:2001192.168.100.32>
To: <sip:2000192.168.100.32>;tag=as3887b859
Call-ID: eE6h192.168.100.32
CSeq: 3 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5,
realm="asterisk", nonce="4e5f93f7"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'eE6h192.168.100.32' in 32000 ms (Method: INVITE)
Found user '2001'
<--- SIP read from 192.168.100.50:5070 --->
INVITE sip:2000192.168.100.32 SIP/2.0
Via: SIP/2.0/UDP
192.168.100.50:5070;branch=z9hG4bK-eE6hF;rport=
From: <sip:2001192.168.100.32>
To: <sip:2000192.168.100.32>;tag=as3887b859
Call-ID: eE6h192.168.100.32
CSeq: 3 INVITE
Contact: <sip:2001192.168.100.50>
User-Agent: TwinSIP Communicator
Content-Type: application/sdp
Content-Length: 99
Proxy-Authorization: Digest username="2001",
realm="asterisk", nonce="4e5f93f7",
uri="192.168.100.32",
response="39abd8d49d07c1cbb489b0fc467d2856",
algorithm=MD5
<------------->
--- (11 headers 0 lines) ---
Ignoring this INVITE request
Retransmitting #1 (no NAT) to 192.168.100.50:5070:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
192.168.100.50:5070;branch=z9hG4bK-eE6hF;rport=;received=192
.168.100.50
From: <sip:2001192.168.100.32>
To: <sip:2000192.168.100.32>;tag=as3887b859
Call-ID: eE6h192.168.100.32
CSeq: 3 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5,
realm="asterisk", nonce="4e5f93f7"
Content-Length: 0
---
<--- SIP read from 192.168.100.50:5070 --->
INVITE sip:2000192.168.100.32 SIP/2.0
Via: SIP/2.0/UDP
192.168.100.50:5070;branch=z9hG4bK-eE6hF;rport=
From: <sip:2001192.168.100.32>
To: <sip:2000192.168.100.32>;tag=as3887b859
Call-ID: eE6h192.168.100.32
CSeq: 3 INVITE
Contact: <sip:2001192.168.100.50>
User-Agent: TwinSIP Communicator
Content-Type: application/sdp
Content-Length: 99
Proxy-Authorization: Digest username="2001",
realm="asterisk", nonce="4e5f93f7",
uri="192.168.100.32",
response="39abd8d49d07c1cbb489b0fc467d2856",
algorithm=MD5
<------------->
--- (11 headers 0 lines) ---
Ignoring this INVITE request
Retransmitting #2 (no NAT) to 192.168.100.50:5070:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
192.168.100.50:5070;branch=z9hG4bK-eE6hF;rport=;received=192
.168.100.50
From: <sip:2001192.168.100.32>
To: <sip:2000192.168.100.32>;tag=as3887b859
Call-ID: eE6h192.168.100.32
CSeq: 3 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5,
realm="asterisk", nonce="4e5f93f7"
Content-Length: 0
---
<--- SIP read from 192.168.100.50:5070 --->
INVITE sip:2000192.168.100.32 SIP/2.0
Via: SIP/2.0/UDP
192.168.100.50:5070;branch=z9hG4bK-eE6hF;rport=
From: <sip:2001192.168.100.32>
To: <sip:2000192.168.100.32>;tag=as3887b859
Call-ID: eE6h192.168.100.32
CSeq: 3 INVITE
Contact: <sip:2001192.168.100.50>
User-Agent: TwinSIP Communicator
Content-Type: application/sdp
Content-Length: 99
Proxy-Authorization: Digest username="2001",
realm="asterisk", nonce="4e5f93f7",
uri="192.168.100.32",
response="39abd8d49d07c1cbb489b0fc467d2856",
algorithm=MD5
<------------->
--- (11 headers 0 lines) ---
Ignoring this INVITE request
Retransmitting #3 (no NAT) to 192.168.100.50:5070:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
192.168.100.50:5070;branch=z9hG4bK-eE6hF;rport=;received=192
.168.100.50
From: <sip:2001192.168.100.32>
To: <sip:2000192.168.100.32>;tag=as3887b859
Call-ID: eE6h192.168.100.32
CSeq: 3 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5,
realm="asterisk", nonce="4e5f93f7"
Content-Length: 0
---
<--- SIP read from 192.168.100.50:5070 --->
INVITE sip:2000192.168.100.32 SIP/2.0
Via: SIP/2.0/UDP
192.168.100.50:5070;branch=z9hG4bK-eE6hF;rport=
From: <sip:2001192.168.100.32>
To: <sip:2000192.168.100.32>;tag=as3887b859
Call-ID: eE6h192.168.100.32
CSeq: 3 INVITE
Contact: <sip:2001192.168.100.50>
User-Agent: TwinSIP Communicator
Content-Type: application/sdp
Content-Length: 99
Proxy-Authorization: Digest username="2001",
realm="asterisk", nonce="4e5f93f7",
uri="192.168.100.32",
response="39abd8d49d07c1cbb489b0fc467d2856",
algorithm=MD5
<------------->
--- (11 headers 0 lines) ---
Ignoring this INVITE request
Retransmitting #4 (no NAT) to 192.168.100.50:5070:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
192.168.100.50:5070;branch=z9hG4bK-eE6hF;rport=;received=192
.168.100.50
From: <sip:2001192.168.100.32>
To: <sip:2000192.168.100.32>;tag=as3887b859
Call-ID: eE6h192.168.100.32
CSeq: 3 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5,
realm="asterisk", nonce="4e5f93f7"
Content-Length: 0
---
<--- SIP read from 192.168.100.50:5070 --->
INVITE sip:2000192.168.100.32 SIP/2.0
Via: SIP/2.0/UDP
192.168.100.50:5070;branch=z9hG4bK-eE6hF;rport=
From: <sip:2001192.168.100.32>
To: <sip:2000192.168.100.32>;tag=as3887b859
Call-ID: eE6h192.168.100.32
CSeq: 3 INVITE
Contact: <sip:2001192.168.100.50>
User-Agent: TwinSIP Communicator
Content-Type: application/sdp
Content-Length: 99
Proxy-Authorization: Digest username="2001",
realm="asterisk", nonce="4e5f93f7",
uri="192.168.100.32",
response="39abd8d49d07c1cbb489b0fc467d2856",
algorithm=MD5
<------------->
--- (11 headers 0 lines) ---
Ignoring this INVITE request
Retransmitting #5 (no NAT) to 192.168.100.50:5070:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
192.168.100.50:5070;branch=z9hG4bK-eE6hF;rport=;received=192
.168.100.50
From: <sip:2001192.168.100.32>
To: <sip:2000192.168.100.32>;tag=as3887b859
Call-ID: eE6h192.168.100.32
CSeq: 3 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5,
realm="asterisk", nonce="4e5f93f7"
Content-Length: 0
---
<--- SIP read from 192.168.100.50:5070 --->
INVITE sip:2000192.168.100.32 SIP/2.0
Via: SIP/2.0/UDP
192.168.100.50:5070;branch=z9hG4bK-eE6hF;rport=
From: <sip:2001192.168.100.32>
To: <sip:2000192.168.100.32>;tag=as3887b859
Call-ID: eE6h192.168.100.32
CSeq: 3 INVITE
Contact: <sip:2001192.168.100.50>
User-Agent: TwinSIP Communicator
Content-Type: application/sdp
Content-Length: 99
Proxy-Authorization: Digest username="2001",
realm="asterisk", nonce="4e5f93f7",
uri="192.168.100.32",
response="39abd8d49d07c1cbb489b0fc467d2856",
algorithm=MD5
<------------->
--- (11 headers 0 lines) ---
Ignoring this INVITE request
Retransmitting #6 (no NAT) to 192.168.100.50:5070:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
192.168.100.50:5070;branch=z9hG4bK-eE6hF;rport=;received=192
.168.100.50
From: <sip:2001192.168.100.32>
To: <sip:2000192.168.100.32>;tag=as3887b859
Call-ID: eE6h192.168.100.32
CSeq: 3 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5,
realm="asterisk", nonce="4e5f93f7"
Content-Length: 0
---
<--- SIP read from 192.168.100.50:5070 --->
INVITE sip:2000192.168.100.32 SIP/2.0
Via: SIP/2.0/UDP
192.168.100.50:5070;branch=z9hG4bK-eE6hF;rport=
From: <sip:2001192.168.100.32>
To: <sip:2000192.168.100.32>;tag=as3887b859
Call-ID: eE6h192.168.100.32
CSeq: 3 INVITE
Contact: <sip:2001192.168.100.50>
User-Agent: TwinSIP Communicator
Content-Type: application/sdp
Content-Length: 99
Proxy-Authorization: Digest username="2001",
realm="asterisk", nonce="4e5f93f7",
uri="192.168.100.32",
response="39abd8d49d07c1cbb489b0fc467d2856",
algorithm=MD5
<------------->
--- (11 headers 0 lines) ---
Ignoring this INVITE request
[Apr 24 17:31:57] WARNING[5391]: chan_sip.c:1939
retrans_pkt: Maximum retries exceeded on transmission
eE6h192.168.100.32 for seqno 3 (Non-critical Response)
Really destroying SIP dialog 'eE6h192.168.100.32' Method:
INVITE
End of INVITE-Part.
Is anything wrong with the SIP-Messages? Or what can we do
to solve this problem?
Thanks in advance,
Alex
------------------------------
Message: 2
Date: Fri, 25 Apr 2008 16:04:10 +0530
From: "Malay Si" <malaysiTechMahindra.com>
Subject: Re: [SIPForum-discussion] 185 ringing / 199 trying
To: "Karthik Ramiya" <Karthik_Ramiyainfosys.com>, "gaurav katiyar"
<gaurav_katiyaryahoo.com>, "mwilliam
prusty"
<callwilliam90653rediffmail.com>, <discussionsipforum.org>
Message-ID:
<95AC8977BA07AD4DA139EAE39D9C1564034A1EEBSINBNGEX001.TechMahindra.com>
Content-Type: text/plain; charset=us-ascii
The reason phrase is for human readability and the reason
code is use
for the server.
Thanks & Regards,
| Malay Si |Bangalore| INDIA
* Office: +91 80 4041 7000 | Extn: 7890 |
Email: malaysitechmahindra.com
www.techmahindra.com
-----Original Message-----
From: Karthik Ramiya [mailto:Karthik_Ramiyainfosys.com]
Sent: Friday, April 25, 2008 3:51 PM
To: Malay Si; gaurav katiyar; mwilliam prusty;
discussionsipforum.org
Subject: RE: [SIPForum-discussion] 185 ringing / 199 trying
Hi Malaysi,
I agree with you. But here it is clearly defined like
the reason
phase is 185 ringing / 199 trying, So you mean like
irrespective of the
reason phases, it wll jus look into the 1XX response and
treat it as
183?? In that case, it means that there wont be any
recognition for the
reason phase rite.
Please clarify,
Thanks and regards,
Karthik
________________________________
From: discussion-bouncessipforum.org
[discussion-bouncessipforum.org]
On Behalf Of Malay Si [malaysitechmahindra.com]
Sent: Friday, April 25, 2008 2:20 PM
To: gaurav katiyar; mwilliam prusty; discussionsipforum.org
Subject: Re: [SIPForum-discussion] 185 ringing / 199 trying
If a SIP element gets a response which is having unknown
"reason-
phrase" (not define in RFC 3261) then the element
treated the response
as follows.
Wrong 1xx series---183(session progress)
Wrong 2xx series---200
Wrong 3xx series---300
Wrong 4xx series---400
Wrong 5xx series---500
Wrong 6xx series---600
In your case A will treat the response as a 183 (session
progress) and
will wait for the final response.
Thanks & Regards,
| Malay Si | Tech Mahindra| Bangalore| INDIA
* Office: +91 80 4041 7000 | Extn: 7890 |
Email: malaysitechmahindra.com<mailto:malaysitechmahindra.com>
www.techmahindra.com<http://www.techmahin
dra.com/>
________________________________
From: discussion-bouncessipforum.org
[mailto:discussion-bouncessipforum.org] On Behalf Of
gaurav katiyar
Sent: Friday, April 25, 2008 1:47 PM
To: mwilliam prusty; discussionsipforum.org
Subject: Re: [SIPForum-discussion] 185 ringing / 199 trying
Sip does not allow to send response for a response. wait for
final
response or u can send cancel to terminate call as per your
application
requirements.
Gaurav
mwilliam prusty <callwilliam90653rediffmail.com> wrote:
Hi
Can any one tell me suppose A is sending Invite request to
B. B in
steadof giving 180 ringing , it is giving 185 ringing( same
message
format as 180 ringing ) or 199 trying ( which has same
message format as
180 ringing) In this case whether A will give any error
response
Regs
William
_______________________________________________
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Post to the list at discussionsipforum.org
Gaurav Katiyar
Induslogic india pvt. ltd
B-34/1 sector 59 NOIDA
Phone:9818381368
________________________________
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------------------------------
Message: 3
Date: 25 Apr 2008 10:40:50 -0000
From: "mwilliam prusty" <callwilliam90653rediffmail.com>
Subject: Re: [SIPForum-discussion] 185 ringing / 199 trying
To: "Malay Si" <malaysiTechMahindra.com>
Cc: Karthik Ramiya <Karthik_Ramiyainfosys.com>,
discussionsipforum.org
Message-ID: <20080425104050.28923.qmailf4mail-235-239.rediffmail.com>
Content-Type: text/plain; charset="iso-8859-1"
?
Hi Malaya
So you are telling that If A is sending Invite to B.
B is sending 185 ringing( msg conect same as 180 ringing )
, the call will sucessful.?? CLARIFY
REGARDS
wILLIAM
On Fri, 25 Apr 2008 Malay Si wrote :
>
>The reason phrase is for human readability and the
reason code is use
>for the server.
>
>Thanks & Regards,
>
>| Malay Si |Bangalore| INDIA
>* Office: +91 80 4041 7000 | Extn: 7890 |
>Email: malaysitechmahindra.com
>www.techmahindra.com
>
>
>-----Original Message-----
> From: Karthik Ramiya [mailto:Karthik_Ramiyainfosys.com]
>Sent: Friday, April 25, 2008 3:51 PM
>To: Malay Si; gaurav katiyar; mwilliam prusty;
discussionsipforum.org
>Subject: RE: [SIPForum-discussion] 185 ringing / 199
trying
>
>
>Hi Malaysi,
>
> I agree with you. But here it is clearly defined
like the reason
>phase is 185 ringing / 199 trying, So you mean like
irrespective of the
>reason phases, it wll jus look into the 1XX response and
treat it as
>183?? In that case, it means that there wont be any
recognition for the
>reason phase rite.
>
>Please clarify,
>
>Thanks and regards,
>Karthik
>
>
>
>
>________________________________
> From: discussion-bouncessipforum.org
[discussion-bouncessipforum.org]
>On Behalf Of Malay Si [malaysitechmahindra.com]
>Sent: Friday, April 25, 2008 2:20 PM
>To: gaurav katiyar; mwilliam prusty; discussionsipforum.org
>Subject: Re: [SIPForum-discussion] 185 ringing / 199
trying
>
>If a SIP element gets a response which is having unknown
"reason-
>phrase" (not define in RFC 3261) then the element
treated the response
>as follows.
>
>Wrong 1xx series---183(session progress)
>Wrong 2xx series---200
>Wrong 3xx series---300
>Wrong 4xx series---400
>Wrong 5xx series---500
>Wrong 6xx series---600
>
>In your case A will treat the response as a 183 (session
progress) and
>will wait for the final response.
>
>
>
>
>Thanks & Regards,
>
>| Malay Si | Tech Mahindra| Bangalore| INDIA
>* Office: +91 80 4041 7000 | Extn: 7890 |
>Email: malaysitechmahindra.com<mailto:malaysitechmahindra.com>
>www.techmahindra.com<http://www.techmahin
dra.com/>
>
>
>
>________________________________
> From: discussion-bouncessipforum.org
>[mailto:discussion-bouncessipforum.org] On Behalf Of
gaurav katiyar
>Sent: Friday, April 25, 2008 1:47 PM
>To: mwilliam prusty; discussionsipforum.org
>Subject: Re: [SIPForum-discussion] 185 ringing / 199
trying
>
>Sip does not allow to send response for a response. wait
for final
>response or u can send cancel to terminate call as per
your application
>requirements.
>
>Gaurav
>
>mwilliam prusty <callwilliam90653rediffmail.com> wrote:
>
>Hi
>
>Can any one tell me suppose A is sending Invite request
to B. B in
>steadof giving 180 ringing , it is giving 185 ringing(
same message
>format as 180 ringing ) or 199 trying ( which has same
message format as
>180 ringing) In this case whether A will give any
error response
>
>
>
>
>Regs
>William
>
>
>_______________________________________________
>This is the SIP Forum discussion mailing list
>TO UNSUBSCRIBE, or edit your delivery options, please
visit
>http:
//sipforum.org/mailman/listinfo/discussion
>Post to the list at discussionsipforum.org
>
>
>Gaurav Katiyar
>Induslogic india pvt. ltd
>B-34/1 sector 59 NOIDA
>Phone:9818381368
>
>
>
>
>________________________________
>Be a better friend, newshound, and know-it-all with
Yahoo! Mobile. Try
>it
>now.<http://us.rd.yahoo.com/evt=51733/*http:/mobil
e.yahoo.com/;_ylt=Ahu0
>6i62sR8HDtDypao8Wcj9tAcJ%20>
>========================================================
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