4 Dec
2008
4 Dec
'08
1:41 p.m.
Hello all,
I recently add a classical Audiocodes Mediant 2000 with 2x 8E1, the purpose
is to have several interconnections with PSTN.
I configured it like this :
Audiocodes registers as a gateway to the Kamailio, using a dedicated port
(5062).
Registration seems to be OK, and the pstn gw uses OPTIONS method to ping the
proxy.
I can attack the Audiocodes with a SIP phone behind Kamailio, no pbm.
But the audiocodes returns some errors about SIP headers sent by Kamailio :
( sip_stack)(44732 ) AcSIPParser: Problem in SIP Message Headers [Time:
12:30:26]
( sip_stack)(44733 ) !! [ERROR] AcSIPParser: Parse Error. Unexpected symbol
'0' in scheme. ALPHA expected
Here you have the example of an INVITE from a SIP phone to the PSTN :
** audiocodes debug **
4d:12h:30m:26s ( lgr_flow)(44730 ) ---- Incoming SIP Message from
77.246.81.132:5060 ----
INVITE sip:0323719001@77.246.81.136:5062;transport=udp SIP/2.0
Record-Route: <sip:77.246.81.132;lr=on;ftag=71078b346a20fb3eo0;nat=yes>
Via: SIP/2.0/UDP 77.246.81.132;branch=z9hG4bKdace.1ab1d59.0
Via: SIP/2.0/UDP
192.168.0.113:5060;rport=15170;received=77.246.81.162;branch=z9hG4bK-b432f96
From: "Sam" <sip:0123451010@sip.720.fr
<sip%3A0123451010(a)sip.720.fr>>;tag=71078b346a20fb3eo0
To: <sip:0323719001@sip.720.fr <sip%3A0323719001(a)sip.720.fr>>
Call-ID: 944d8aec-27503ee6(a)192.168.0.113
CSeq: 102 INVITE
Max-Forwards: 49
Contact: "Sam" <sip:0123451010@77.246.81.162:15170>
Expires: 240
User-Agent: Linksys/SPA941-5.1.8
Content-Length: 281
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER
Supported: 100rel, replaces
Content-Type: application/sdp
P-Asserted-Identity: <0123451010>
Remote-Party-ID: <0123451010>;party=caller;privacy=none;screen=yes
v=0
o=- 26933860 26933860 IN IP4 192.168.0.113
s=-
c=IN IP4 77.246.81.133
t=0 0
m=audio 35038 RTP/AVP 18 0 8 101
a=rtpmap:18 G729a/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
a=nortpproxy:yes
( sip_stack)(44732 ) AcSIPParser: Problem in SIP Message Headers [Time:
12:30:26]
( sip_stack)(44733 ) !! [ERROR] AcSIPParser: Parse Error. Unexpected symbol
'0' in scheme. ALPHA expected
( sip_stack)(44734 ) !! [ERROR] Message type: INVITE [Time: 12:30:26]
( sip_stack)(44735 ) !! [ERROR] Source header: [Time: 12:30:26]
( sip_stack)(44736 ) !! [ERROR] Line: 17. Column: 23 [Time: 12:30:26]
The outgoing INVITE from Kamailio is exactly the same received by the
AudioCodes.
When I searched over Google, I just found 2 answers about Asterisk /
Audiocodes unsolved problem, but no more informations.
I supposed that the problem is as indicated : " s=- " where source is empty
in place of "NULL" / "0" or something like this ...
Someone can confirm or already met the problem ?
Many thanks all :)
.Sam.
4 Dec
4 Dec
2:28 p.m.
It seems that the P-Asserted-Identity header is not correctly
formatted in the INVITE. It must be a sip, sips, or tel URI. This
would be something that your proxy is adding to the INVITE. Here is a
quote from section RFC 3325.
9.1 The P-Asserted-Identity Header
The P-Asserted-Identity header field is used among trusted SIP
entities (typically intermediaries) to carry the identity of the user
sending a SIP message as it was verified by authentication.
PAssertedID = "P-Asserted-Identity" HCOLON PAssertedID-value
*(COMMA PAssertedID-value)
PAssertedID-value = name-addr / addr-spec
A P-Asserted-Identity header field value MUST consist of exactly one
name-addr or addr-spec. There may be one or two P-Asserted-Identity
values. If there is one value, it MUST be a sip, sips, or tel URI.
--
Raj Jain
On Thu, Dec 4, 2008 at 6:41 AM, Samuel Muller <sml(a)720.fr> wrote:
Hello all,
I recently add a classical Audiocodes Mediant 2000 with 2x 8E1, the purpose
is to have several interconnections with PSTN.
I configured it like this :
Audiocodes registers as a gateway to the Kamailio, using a dedicated port
(5062).
Registration seems to be OK, and the pstn gw uses OPTIONS method to ping the
proxy.
I can attack the Audiocodes with a SIP phone behind Kamailio, no pbm.
But the audiocodes returns some errors about SIP headers sent by Kamailio :
( sip_stack)(44732 ) AcSIPParser: Problem in SIP Message Headers [Time:
12:30:26]
( sip_stack)(44733 ) !! [ERROR] AcSIPParser: Parse Error. Unexpected symbol
'0' in scheme. ALPHA expected
Here you have the example of an INVITE from a SIP phone to the PSTN :
** audiocodes debug **
4d:12h:30m:26s ( lgr_flow)(44730 ) ---- Incoming SIP Message from
77.246.81.132:5060 ----
INVITE sip:0323719001@77.246.81.136:5062;transport=udp SIP/2.0
Record-Route: <sip:77.246.81.132;lr=on;ftag=71078b346a20fb3eo0;nat=yes>
Via: SIP/2.0/UDP 77.246.81.132;branch=z9hG4bKdace.1ab1d59.0
Via: SIP/2.0/UDP
192.168.0.113:5060;rport=15170;received=77.246.81.162;branch=z9hG4bK-b432f96
From: "Sam" <sip:0123451010@sip.720.fr>;tag=71078b346a20fb3eo0
To: <sip:0323719001@sip.720.fr>
Call-ID: 944d8aec-27503ee6(a)192.168.0.113
CSeq: 102 INVITE
Max-Forwards: 49
Contact: "Sam" <sip:0123451010@77.246.81.162:15170>
Expires: 240
User-Agent: Linksys/SPA941-5.1.8
Content-Length: 281
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER
Supported: 100rel, replaces
Content-Type: application/sdp
P-Asserted-Identity: <0123451010>
Remote-Party-ID: <0123451010>;party=caller;privacy=none;screen=yes
v=0
o=- 26933860 26933860 IN IP4 192.168.0.113
s=-
c=IN IP4 77.246.81.133
t=0 0
m=audio 35038 RTP/AVP 18 0 8 101
a=rtpmap:18 G729a/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
a=nortpproxy:yes
( sip_stack)(44732 ) AcSIPParser: Problem in SIP Message Headers [Time:
12:30:26]
( sip_stack)(44733 ) !! [ERROR] AcSIPParser: Parse Error. Unexpected symbol
'0' in scheme. ALPHA expected
( sip_stack)(44734 ) !! [ERROR] Message type: INVITE [Time: 12:30:26]
( sip_stack)(44735 ) !! [ERROR] Source header: [Time: 12:30:26]
( sip_stack)(44736 ) !! [ERROR] Line: 17. Column: 23 [Time: 12:30:26]
The outgoing INVITE from Kamailio is exactly the same received by the
AudioCodes.
When I searched over Google, I just found 2 answers about Asterisk /
Audiocodes unsolved problem, but no more informations.
I supposed that the problem is as indicated : " s=- " where source is empty
in place of "NULL" / "0" or something like this ...
Someone can confirm or already met the problem ?
Many thanks all :)
.Sam.
_______________________________________________
Users mailing list
Users(a)lists.kamailio.org
http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
2:35 p.m.
New subject: [Kamailio-Users] AudioCodes + Kamailio : Problem in SIP Message Headers
Further, the log message does not have an empty line between SIP headers
and the body. Either you have forgotten to add \r\n when adding the
header or this is just not diplays correctly in the logfile.
klaus
Raj Jain schrieb:
It seems that the P-Asserted-Identity header is not
correctly
formatted in the INVITE. It must be a sip, sips, or tel URI. This
would be something that your proxy is adding to the INVITE. Here is a
quote from section RFC 3325.
9.1 The P-Asserted-Identity Header
The P-Asserted-Identity header field is used among trusted SIP
entities (typically intermediaries) to carry the identity of the user
sending a SIP message as it was verified by authentication.
PAssertedID = "P-Asserted-Identity" HCOLON PAssertedID-value
*(COMMA PAssertedID-value)
PAssertedID-value = name-addr / addr-spec
A P-Asserted-Identity header field value MUST consist of exactly one
name-addr or addr-spec. There may be one or two P-Asserted-Identity
values. If there is one value, it MUST be a sip, sips, or tel URI.
--
Raj Jain
On Thu, Dec 4, 2008 at 6:41 AM, Samuel Muller <sml(a)720.fr> wrote:
Hello all,
I recently add a classical Audiocodes Mediant 2000 with 2x 8E1, the purpose
is to have several interconnections with PSTN.
I configured it like this :
Audiocodes registers as a gateway to the Kamailio, using a dedicated port
(5062).
Registration seems to be OK, and the pstn gw uses OPTIONS method to ping the
proxy.
I can attack the Audiocodes with a SIP phone behind Kamailio, no pbm.
But the audiocodes returns some errors about SIP headers sent by Kamailio :
( sip_stack)(44732 ) AcSIPParser: Problem in SIP Message Headers [Time:
12:30:26]
( sip_stack)(44733 ) !! [ERROR] AcSIPParser: Parse Error. Unexpected symbol
'0' in scheme. ALPHA expected
Here you have the example of an INVITE from a SIP phone to the PSTN :
** audiocodes debug **
4d:12h:30m:26s ( lgr_flow)(44730 ) ---- Incoming SIP Message from
77.246.81.132:5060 ----
INVITE sip:0323719001@77.246.81.136:5062;transport=udp SIP/2.0
Record-Route: <sip:77.246.81.132;lr=on;ftag=71078b346a20fb3eo0;nat=yes>
Via: SIP/2.0/UDP 77.246.81.132;branch=z9hG4bKdace.1ab1d59.0
Via: SIP/2.0/UDP
192.168.0.113:5060;rport=15170;received=77.246.81.162;branch=z9hG4bK-b432f96
From: "Sam" <sip:0123451010@sip.720.fr>;tag=71078b346a20fb3eo0
To: <sip:0323719001@sip.720.fr>
Call-ID: 944d8aec-27503ee6(a)192.168.0.113
CSeq: 102 INVITE
Max-Forwards: 49
Contact: "Sam" <sip:0123451010@77.246.81.162:15170>
Expires: 240
User-Agent: Linksys/SPA941-5.1.8
Content-Length: 281
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER
Supported: 100rel, replaces
Content-Type: application/sdp
P-Asserted-Identity: <0123451010>
Remote-Party-ID: <0123451010>;party=caller;privacy=none;screen=yes
v=0
o=- 26933860 26933860 IN IP4 192.168.0.113
s=-
c=IN IP4 77.246.81.133
t=0 0
m=audio 35038 RTP/AVP 18 0 8 101
a=rtpmap:18 G729a/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
a=nortpproxy:yes
( sip_stack)(44732 ) AcSIPParser: Problem in SIP Message Headers [Time:
12:30:26]
( sip_stack)(44733 ) !! [ERROR] AcSIPParser: Parse Error. Unexpected symbol
'0' in scheme. ALPHA expected
( sip_stack)(44734 ) !! [ERROR] Message type: INVITE [Time: 12:30:26]
( sip_stack)(44735 ) !! [ERROR] Source header: [Time: 12:30:26]
( sip_stack)(44736 ) !! [ERROR] Line: 17. Column: 23 [Time: 12:30:26]
The outgoing INVITE from Kamailio is exactly the same received by the
AudioCodes.
When I searched over Google, I just found 2 answers about Asterisk /
Audiocodes unsolved problem, but no more informations.
I supposed that the problem is as indicated : " s=- " where source is empty
in place of "NULL" / "0" or something like this ...
Someone can confirm or already met the problem ?
Many thanks all :)
.Sam.
_______________________________________________
Users mailing list
Users(a)lists.kamailio.org
http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
_______________________________________________
Users mailing list
Users(a)lists.kamailio.org
http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
5:21 p.m.
Hello guys,
many thanks, you were right :)
I changed the PAI and the RPID stuff and it works ...
-- KAMAILIO --
# flag 9 = clir
if (is_avp_set("$avp(s:caller_cli)/s") && !isflagset(9))
{
if (is_present_hf("P-Asserted-Identity"))
{ remove_hf("P-Asserted-Identity"); }
if (is_present_hf("Remote-Party-ID"))
{ remove_hf("Remote-Party-ID"); }
append_hf("P-Asserted-Identity: $avp(s:caller_cli)
<sip:$avp(s:caller_cli)@$fd>\r\n");
append_hf("Remote-Party-ID: $avp(s:caller_cli)
<sip:$avp(s:caller_cli)@$si>;party=caller;privacy=none;screen=yes\r\n");
}
Do you have a better solution to have the best rpid and pai coding way ?
And, is the P-Preferred-Identity really necessary for PSTN ?
log in the gateway :
-- AUDIOCODES --
4d:15h:33m:43s ( lgr_flow)(51994 ) ---- Incoming SIP Message from
77.246.81.132:5060 ----
INVITE sip:0663128505@77.246.81.136:5062;transport=udp SIP/2.0
Record-Route: <sip:77.246.81.132;lr=on;ftag=a4143abfbda0611ao0;nat=yes Via: SIP/2.0/UDP
77.246.81.132;branch=z9hG4bK89df.da4cd5e2.0
Via: SIP/2.0/UDP 192.168.0.113:5060;rport=15170;received=77.246.81.162
;branch=z9hG4bK-8a13206a
From: "Sam" <sip:0123451010@sip.720.fr <sip%3A0123451010(a)sip.720.fr
Call-ID: ced89363-47d540c6(a)192.168.0.113
CSeq: 102 INVITE
Max-Forwards: 49
Contact: "Sam" <sip:0123451010@77.246.81.162:15170 Expires: 240
User-Agent: Linksys/SPA941-5.1.8
Content-Length: 281
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER
Supported: 100rel, replaces
Content-Type: application/sdp
P-Asserted-Identity: 0123451010
<sip:0123451010@sip.720.fr<sip%3A0123451010@sip.720.fr
Remote-Party-ID: 0123451010
<sip:0123451010@77.246.81.162<sip%3A0123451010@77.246.81.162 > klaus
> Raj Jain schrieb:
> It seems that the P-Asserted-Identity
header is not correctly
>> formatted in the INVITE. It must be a sip, sips, or tel URI. This
>> would be something that your proxy is adding to the INVITE. Here is a
>> quote from section RFC 3325.
> >
>> 9.1 The P-Asserted-Identity Header
> >> The P-Asserted-Identity header field
is used among trusted SIP
>> entities (typically intermediaries) to carry the identity of the user
>> sending a SIP message as it was verified by authentication.
> >> PAssertedID =
"P-Asserted-Identity" HCOLON PAssertedID-value
>> *(COMMA PAssertedID-value)
>> PAssertedID-value = name-addr / addr-spec
> >> A P-Asserted-Identity header field
value MUST consist of exactly one
>> name-addr or addr-spec. There may be one or two P-Asserted-Identity
>> values. If there is one value, it MUST be a sip, sips, or tel URI.
> >> --
>> Raj Jain
> >> On Thu, Dec 4, 2008 at 6:41 AM, Samuel
Muller <sml(a)720.fr >>> Hello all,
>> >>> I
recently add a classical Audiocodes Mediant 2000 with 2x 8E1, the
>>> purpose
>>> is to have several interconnections with PSTN.
>> >>> I
configured it like this :
>> >>>
Audiocodes registers as a gateway to the Kamailio, using a dedicated port
>>> (5062).
>>> Registration seems to be OK, and the pstn gw uses OPTIONS method to ping
>>> the
>>> proxy.
>>> I can attack the Audiocodes with a SIP phone behind Kamailio, no pbm.
>> >>> But the
audiocodes returns some errors about SIP headers sent by Kamailio
>>> :
>> >>> (
sip_stack)(44732 ) AcSIPParser: Problem in SIP Message Headers [Time:
>>> 12:30:26]
>>> ( sip_stack)(44733 ) !! [ERROR] AcSIPParser: Parse Error. Unexpected
>>> symbol
>>> '0' in scheme. ALPHA expected
>> >>> Here you
have the example of an INVITE from a SIP phone to the PSTN :
>> >>> **
audiocodes debug **
>> >>>
4d:12h:30m:26s ( lgr_flow)(44730 ) ---- Incoming SIP Message from
>>> 77.246.81.132:5060 ----
>> >>> INVITE
sip:0323719001@77.246.81.136:5062;transport=udp SIP/2.0
>>> Record-Route: <sip:77.246.81.132;lr=on;ftag=71078b346a20fb3eo0;nat=yes >>> Via: SIP/2.0/UDP
77.246.81.132;branch=z9hG4bKdace.1ab1d59.0
>>> Via: SIP/2.0/UDP
>>> 192.168.0.113:5060;rport=15170;received=77.246.81.162
>>> ;branch=z9hG4bK-b432f96
>>> From: "Sam" <sip:0123451010@sip.720.fr
<sip%3A0123451010(a)sip.720.fr >>>
>;tag=71078b346a20fb3eo0
>>> To: <sip:0323719001@sip.720.fr <sip%3A0323719001(a)sip.720.fr> >>> Call-ID:
944d8aec-27503ee6(a)192.168.0.113
>>> CSeq: 102 INVITE
>>> Max-Forwards: 49
>>> Contact: "Sam" <sip:0123451010@77.246.81.162:15170 >>> Expires: 240
>>> User-Agent: Linksys/SPA941-5.1.8
>>> Content-Length: 281
>>> Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER
>>> Supported: 100rel, replaces
>>> Content-Type: application/sdp
>>> P-Asserted-Identity: <0123451010
>>> Remote-Party-ID: <0123451010 >>> (
sip_stack)(44732 ) AcSIPParser: Problem in SIP Message Headers [Time:
>>> 12:30:26]
>>> ( sip_stack)(44733 ) !! [ERROR] AcSIPParser: Parse Error. Unexpected
>>> symbol
>>> '0' in scheme. ALPHA expected
>>> ( sip_stack)(44734 ) !! [ERROR] Message type: INVITE [Time: 12:30:26]
>>> ( sip_stack)(44735 ) !! [ERROR] Source header: [Time: 12:30:26]
>>> ( sip_stack)(44736 ) !! [ERROR] Line: 17. Column: 23 [Time: 12:30:26]
>> >> >>> The outgoing INVITE from Kamailio
is exactly the same received by the
>>> AudioCodes.
>>> When I searched over Google, I just found 2 answers about Asterisk /
>>> Audiocodes unsolved problem, but no more informations.
>> >>> I
supposed that the problem is as indicated : " s=- " where source is
>>> empty
>>> in place of "NULL" / "0" or something like this ...
>>> Someone can confirm or already met the problem ?
>> >>> Many
thanks all :)
>> >>> .Sam.
>> >> >>
>> >> >>>
_______________________________________________
>>> Users mailing list
>>> Users(a)lists.kamailio.org
>>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
>> >> >>
>> _______________________________________________
>> Users mailing list
>> Users(a)lists.kamailio.org
>> http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
>
--
Samuel MULLER
Ingénieur Reseaux & Telecom
720 DEGRES
+33 (0)663 128 505
sml(a)720.fr
;tag=a4143abfbda0611ao0
To: <sip:0663128505@sip.720.fr
<sip%3A0663128505(a)sip.720.fr>
;party=caller;privacy=none;screen=yes
v=0 o=- 28033614 28033614 IN IP4 192.168.0.113
s=-
c=IN IP4 77.246.81.133
t=0 0
m=audio 35056 RTP/AVP 18 0 8 101
a=rtpmap:18 G729a/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
a=nortpproxy:yes [Time: 15:33:43]
( lgr_flow)(51996 ) | | new GetNewSIPCall created - #357 [Time: 15:33:43]
( sip_stack)(51997 ) new AcSIPCallAPI created - #285 [Time: 15:33:43]
( lgr_stk_mngr)(51998 ) Resource StackSession <#285> Allocated [Time:
15:33:43]
( lgr_flow)(51999 ) | |(SIPTU#357)INVITE State:Idle() [Time: 15:33:43]
( sip_stack)(52000 ) SIPCall(#357) changes state from Idle to Invited [Time:
15:33:43]
( lgr_flow)(52001 ) | | | #285:SIP_SETUP_EV(ced89363-47d540c6@192.168.0.113)
[Time: 15:33:43]
( lgr_callf)(52002 ) new Call created - #285 [Time: 15:33:43]
( lgr_stk_ses)(52003 ) SIPStackSession::HandleStackSetupEV - NEWCALL:
SrcPN=0 [Time: 15:33:43]
( lgr_stk_ses)(52004 ) <SESSION #285> SendToCall - event: NEW_CALL_EV m_Call
= 108260848 [Time: 15:33:43]
( lgr_flow)(52033 ) ---- Incoming SIP Message from 77.246.81.132:5060 ----
[Time: 15:33:43]
ACK sip:0663128505@77.246.81.136:5062;transport=udp SIP/2.0
Via: SIP/2.0/UDP 77.246.81.132;branch=z9hG4bK89df.da4cd5e2.0
From: "Sam" <sip:0123451010@sip.720.fr <sip%3A0123451010(a)sip.720.fr
;tag=a4143abfbda0611ao0
Call-ID: ced89363-47d540c6(a)192.168.0.113
To: <sip:0663128505@sip.720.fr <sip%3A0663128505(a)sip.720.fr
;tag=1c249703390
CSeq: 102 ACK
Max-Forwards: 70
User-Agent: kamailio 1.4.2 - 720 DEGRES
Content-Length: 0
( sip_stack)(52035 ) UdpRtxMngr::Remove 404 Response 102 INVITE [Time:
15:33:43]
( lgr_flow)(52036 ) | |(SIPTU#357)ACK State:Disconnected(
ced89363-47d540c6(a)192.168.0.113) [Time: 15:33:43]
Again, thanks guys :)
.Sam.
On Thu, Dec 4, 2008 at 1:35 PM, Klaus Darilion <klaus.mailinglists(a)pernau.at
wrote:
> Further, the log message does not have an empty line between SIP headers
> and the body. Either you have forgotten to add \r\n when adding the header
> or this is just not diplays correctly in the logfile.
wrote:
>
;party=caller;privacy=none;screen=yes
>>> v=0
>>> o=- 26933860 26933860 IN IP4 192.168.0.113
>>> s=-
>>> c=IN IP4 77.246.81.133
>>> t=0 0
>>> m=audio 35038 RTP/AVP 18 0 8 101
>>> a=rtpmap:18 G729a/8000
>>> a=rtpmap:0 PCMU/8000
>>> a=rtpmap:8 PCMA/8000
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-15
>>> a=ptime:30
>>> a=sendrecv
>>> a=nortpproxy:yes
>>
5:28 p.m.
2008/12/4 Samuel Muller <sml(a)720.fr>fr>:
append_hf("P-Asserted-Identity:
$avp(s:caller_cli)
<sip:$avp(s:caller_cli)@$fd>\r\n");
append_hf("Remote-Party-ID: $avp(s:caller_cli)
<sip:$avp(s:caller_cli)@$si>;party=caller;privacy=none;screen=yes\r\n");
}
Do you have a better solution to have the best rpid and pai coding way ?
It's ok now.
And, is the P-Preferred-Identity really necessary for
PSTN ?
It depends on the gateway configuration, but basically PPI shouldn't
be used when sending the request to a gateway (they usually expect
RPID or PAI).
--
Iñaki Baz Castillo
<ibc(a)aliax.net>
26 Sep
26 Sep
9:55 a.m.
New subject: AudioCodes + Kamailio : Problem in SIP Message Headers
Samuel Muller-2 wrote:
Hello all,
I recently add a classical Audiocodes Mediant 2000 with 2x 8E1, the
purpose
is to have several interconnections with PSTN.
I configured it like this :
Audiocodes registers as a gateway to the Kamailio, using a dedicated port
(5062).
Registration seems to be OK, and the pstn gw uses OPTIONS method to ping
the
proxy.
I can attack the Audiocodes with a SIP phone behind Kamailio, no pbm.
But the audiocodes returns some errors about SIP headers sent by Kamailio
:
( sip_stack)(44732 ) AcSIPParser: Problem in SIP Message Headers [Time:
12:30:26]
( sip_stack)(44733 ) !! [ERROR] AcSIPParser: Parse Error. Unexpected
symbol
'0' in scheme. ALPHA expected
Here you have the example of an INVITE from a SIP phone to the PSTN :
** audiocodes debug **
4d:12h:30m:26s ( lgr_flow)(44730 ) ---- Incoming SIP Message from
77.246.81.132:5060 ----
INVITE sip:0323719001@77.246.81.136:5062;transport=udp SIP/2.0
Record-Route: <sip:77.246.81.132;lr=on;ftag=71078b346a20fb3eo0;nat=yes>
Via: SIP/2.0/UDP 77.246.81.132;branch=z9hG4bKdace.1ab1d59.0
Via: SIP/2.0/UDP
192.168.0.113:5060;rport=15170;received=77.246.81.162;branch=z9hG4bK-b432f96
From: "Sam" <sip:0123451010@sip.720.fr
<sip%3A0123451010(a)sip.720.fr>>;tag=71078b346a20fb3eo0
To: <sip:0323719001@sip.720.fr <sip%3A0323719001(a)sip.720.fr>>
Call-ID: 944d8aec-27503ee6(a)192.168.0.113
CSeq: 102 INVITE
Max-Forwards: 49
Contact: "Sam" <sip:0123451010@77.246.81.162:15170>
Expires: 240
User-Agent: Linksys/SPA941-5.1.8
Content-Length: 281
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER
Supported: 100rel, replaces
Content-Type: application/sdp
P-Asserted-Identity: <0123451010>
Remote-Party-ID: <0123451010>;party=caller;privacy=none;screen=yes
v=0
o=- 26933860 26933860 IN IP4 192.168.0.113
s=-
c=IN IP4 77.246.81.133
t=0 0
m=audio 35038 RTP/AVP 18 0 8 101
a=rtpmap:18 G729a/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
a=nortpproxy:yes
( sip_stack)(44732 ) AcSIPParser: Problem in SIP Message Headers [Time:
12:30:26]
( sip_stack)(44733 ) !! [ERROR] AcSIPParser: Parse Error. Unexpected
symbol
'0' in scheme. ALPHA expected
( sip_stack)(44734 ) !! [ERROR] Message type: INVITE [Time: 12:30:26]
( sip_stack)(44735 ) !! [ERROR] Source header: [Time: 12:30:26]
( sip_stack)(44736 ) !! [ERROR] Line: 17. Column: 23 [Time: 12:30:26]
The outgoing INVITE from Kamailio is exactly the same received by the
AudioCodes.
When I searched over Google, I just found 2 answers about Asterisk /
Audiocodes unsolved problem, but no more informations.
I supposed that the problem is as indicated : " s=- " where source is
empty
in place of "NULL" / "0" or something like this ...
Someone can confirm or already met the problem ?
Many thanks all :)
.Sam.
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5 comments
5 participants
participants (5)
-
cong -
Iñaki Baz Castillo -
Klaus Darilion -
Raj Jain -
Samuel Muller