4 Dec
2003
4 Dec
'03
11:12 p.m.
It would appear that SER v0.8.12 is performing URI matching against the
INVITE line rather than the To: header. This appears to cause problems
with Snom phones. Below is the initial INVITE request sent by a Snom
200:
INVITE sip:172.21.30.53 SIP/2.0..Via: SIP/2.0/UDP 172.21.20.89:5060;branch=
z9hG4bK-d4iq2xxwsyff..Route: <sip:5803932@172.21.30.53;user=phone>..From: "
xxxxxx" <sip:IC89@172.21.30.53>;tag=tf5tyz77r1..To:
<sip:5803932@172.21.30.
53;user=phone>..Call-ID: 3c267820f20d-hj2ilvrq6eoj@172-21-20-89..CSeq: 1 IN
VITE..Max-Forwards: 70..Contact: <sip:IC89@172.21.20.89:5060;line=1>..User-
Agent: snom200-2.02t..Accept-Language: en..Accept: application/sdp..Allow:
INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE
, INFO..Supported: timer, 100rel, replaces..Session-Expires: 7200..Content-
Type: application/sdp..Content-Length: 270....v=0..o=root 9679 9679 IN IP4
172.21.20.89..s=SIP Call..c=IN IP4 172.21.20.89..t=0 0..m=audio 10010 RTP/A
VP 0 8 3 18 97..a=rtpmap:0 pcmu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/
8000..a=rtpmap:18 g729/8000..a=rtpmap:97 telephone-event/8000..a=fmtp:97 0-
15..a=sendrecv..
The SER configuration has the following directive:
if (uri=~"^sip:580[0-9][0-9][0-9][0-9]@") {
log(1, "Local call sent to gateway");
rewritehost("172.21.30.51");
t_relay();
break;
};
This fails to match the above request even though the To: header in the
above request contains the necessary pattern. This same configuration
works fine with a Cisco phone, but it's INVITE request contains the a
complete URI:
INVITE sip:5803932@172.21.30.51 SIP/2.0..Max-Forwards: 10..Record-Route: <s
ip:5803932@172.21.30.53;ftag=f2be0b0014cdf23a7f153-5b3b8b0b;lr=on>..Via: SI
P/2.0/UDP 172.21.30.53;branch=z9hG4bK72cc.024d7ee4.0..Via: SIP/2.0/UDP 172.
21.20.225:5060..From: "IC225"
<sip:IC225@172.21.30.53>;tag=f2be0b0014cdf23a
7f153-5b3b8b0b..To: <sip:5803932@172.21.30.53>..Call-ID: 000bbef2-df4c0003-
6004eb37-6333d578@172.21.20.225..Date: Thu, 04 Dec 2003 20:42:51 GMT..CSeq:
101 INVITE..Expires: 180..User-Agent: Cisco-SIP-IP-Phone/2..Accept: applic
ation/sdp..Contact: sip:IC225@172.21.20.225:5060..Content-Type: application
/sdp..Content-Length: 222....v=0..o=CiscoSystemsSIP-IPPhone-UserAgent 8030
16590 IN IP4 172.21.20.225..s=SIP Call..c=IN IP4 172.21.20.225..t=0 0..m=au
dio 26274 RTP/AVP 0 8 18 96..a=rtpmap:0 pcmu/8000..a=rtpmap:96 telephone-ev
ent/8000..a=fmtp:96 0-11..
Is there anything I can change in the ser.cfg to instruct SER to match
against the To: header?
--
Jamin W. Collins
Remember, root always has a loaded gun. Don't run around with it unless
you absolutely need it. -- Vineet Kumar
5 Dec
5 Dec
2:11 a.m.
On Thu, Dec 04, 2003 at 02:12:36PM -0700, Jamin W. Collins wrote:
It would appear that SER v0.8.12 is performing URI
matching against the
INVITE line rather than the To: header.
I've just confirmed that this behavior is indeed different than 0.8.10.
I've rolled the servers back to 0.8.10, for now.
--
Jamin W. Collins
Linux is not The Answer. Yes is the answer. Linux is The Question. - Neo
11:29 a.m.
On Dec 04, 2003 at 17:11, Jamin W. Collins <jcollins(a)asgardsrealm.net> wrote:
On Thu, Dec 04, 2003 at 02:12:36PM -0700, Jamin W.
Collins wrote:
As Jiri said, ser always matched URI against request URIs and not To.
This behaviour never changed (I wrote the uri matching iand it is the same
from ser 0.0.1).
What's happening in your case, is snom uses a preloaded route ser, with
the proper uri in it. Cisco sends it directly as the request uri.
Do you use route processing before uri matching in both scripts?
(0.8.10 & 0.8.12)
I think you use it 0.8.10, but I'm not sure you do so in 0.8.12.
Andrei
It would appear that SER v0.8.12 is performing
URI matching against the
INVITE line rather than the To: header.
I've just confirmed that this behavior is indeed different than 0.8.10.
I've rolled the servers back to 0.8.10, for now.
5:31 p.m.
On Fri, Dec 05, 2003 at 10:29:06AM +0100, Andrei Pelinescu-Onciul wrote:
On Dec 04, 2003 at 17:11, Jamin W. Collins wrote:
I've forwared both configurations to Jiri. I tried to keep the logic
essentially the same between versions. It's quite possible that I've
missed something between the two, but I certainly don't see what. I'm
going to gather ngrep dumps of calls from the different versions.
--
Jamin W. Collins
Linux is not The Answer. Yes is the answer. Linux is The Question. - Neo
I've just confirmed that this behavior is indeed different than
0.8.10. I've rolled the servers back to 0.8.10, for now.
As Jiri said, ser always matched URI against request URIs and not To.
This behaviour never changed (I wrote the uri matching iand it is the
same from ser 0.0.1).
What's happening in your case, is snom uses a preloaded route ser,
with the proper uri in it. Cisco sends it directly as the request uri.
Do you use route processing before uri matching in both scripts?
(0.8.10 & 0.8.12)
I think you use it 0.8.10, but I'm not sure you do so in 0.8.12. 
2:48 a.m.
At 10:12 PM 12/4/2003, Jamin W. Collins wrote:
It would appear that SER v0.8.12 is performing URI
matching against the
INVITE line rather than the To: header. This appears to cause problems
with Snom phones. Below is the initial INVITE request sent by a Snom
200:
INVITE sip:172.21.30.53 SIP/2.0..Via: SIP/2.0/UDP 172.21.20.89:5060;branch=
z9hG4bK-d4iq2xxwsyff..Route: <sip:5803932@172.21.30.53;user=phone>..From: "
xxxxxx" <sip:IC89@172.21.30.53>;tag=tf5tyz77r1..To:
<sip:5803932@172.21.30.
53;user=phone>..Call-ID: 3c267820f20d-hj2ilvrq6eoj@172-21-20-89..CSeq: 1 IN
VITE..Max-Forwards: 70..Contact: <sip:IC89@172.21.20.89:5060;line=1>..User-
Agent: snom200-2.02t..Accept-Language: en..Accept: application/sdp..Allow:
INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE
, INFO..Supported: timer, 100rel, replaces..Session-Expires: 7200..Content-
Type: application/sdp..Content-Length: 270....v=0..o=root 9679 9679 IN IP4
172.21.20.89..s=SIP Call..c=IN IP4 172.21.20.89..t=0 0..m=audio 10010 RTP/A
VP 0 8 3 18 97..a=rtpmap:0 pcmu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/
8000..a=rtpmap:18 g729/8000..a=rtpmap:97 telephone-event/8000..a=fmtp:97 0-
15..a=sendrecv..
The SER configuration has the following directive:
if (uri=~"^sip:580[0-9][0-9][0-9][0-9]@") {
log(1, "Local call sent to gateway");
rewritehost("172.21.30.51");
t_relay();
break;
};
This fails to match the above request even though the To: header in the
above request contains the necessary pattern. This same configuration
works fine with a Cisco phone, but it's INVITE request contains the a
complete URI:
INVITE sip:5803932@172.21.30.51 SIP/2.0..Max-Forwards: 10..Record-Route: <s
ip:5803932@172.21.30.53;ftag=f2be0b0014cdf23a7f153-5b3b8b0b;lr=on>..Via: SI
P/2.0/UDP 172.21.30.53;branch=z9hG4bK72cc.024d7ee4.0..Via: SIP/2.0/UDP 172.
21.20.225:5060..From: "IC225"
<sip:IC225@172.21.30.53>;tag=f2be0b0014cdf23a
7f153-5b3b8b0b..To: <sip:5803932@172.21.30.53>..Call-ID: 000bbef2-df4c0003-
6004eb37-6333d578@172.21.20.225..Date: Thu, 04 Dec 2003 20:42:51 GMT..CSeq:
101 INVITE..Expires: 180..User-Agent: Cisco-SIP-IP-Phone/2..Accept: applic
ation/sdp..Contact: sip:IC225@172.21.20.225:5060..Content-Type: application
/sdp..Content-Length: 222....v=0..o=CiscoSystemsSIP-IPPhone-UserAgent 8030
16590 IN IP4 172.21.20.225..s=SIP Call..c=IN IP4 172.21.20.225..t=0 0..m=au
dio 26274 RTP/AVP 0 8 18 96..a=rtpmap:0 pcmu/8000..a=rtpmap:96 telephone-ev
ent/8000..a=fmtp:96 0-11..
Is there anything I can change in the ser.cfg to instruct SER to match
against the To: header?
SER has always matched against request URI and that's the proper thing to do.
The INVITE above should be processed using Route processing, as for example
in default script.
-jiri
5:23 a.m.
On Fri, Dec 05, 2003 at 01:48:15AM +0100, Jiri Kuthan wrote:
SER has always matched against request URI and that's the proper thing
to do. The INVITE above should be processed using Route processing,
as for example in default script.
Testing the same configuration (adjusted for syntax changes) with 0.8.10
and 0.8.12 reveals that the two versions handle the request differently.
With 0.8.10, the request is rewritten and forwarded to 172.21.30.51 with
0.8.12 it is not.
Section 2.2.2.2 example 2-7 of the Admin guide[1] shows almost exactly
the syntax and matching I'm trying to use.
Additionally, your statement doesn't explain why the same config would
work for the Cisco but not the Snom. They both have the same To:
request, but different INVITE lines. The Snom request is forwarded
based on the configuration's default rather than the above special
condition, while the Cisco's is handled by the special condition as I
would have expected both to be. I can provide full configuration files
and ngrep's for test's from both phones with both 0.8.10 and 0.8.12, if
it would help.
[1] - http://iptel.org/ser/doc/seruser/seruser.html
--
Jamin W. Collins
Linux is not The Answer. Yes is the answer. Linux is The Question. - Neo
5:32 a.m.
At 04:23 AM 12/5/2003, Jamin W. Collins wrote:
On Fri, Dec 05, 2003 at 01:48:15AM +0100, Jiri Kuthan
wrote:
config file similarity does not imply that either of them is correct.
As I told previously, your script apparently ignored Route processing.
Why it worked for one telephone and not for some other one may or may
not be related to their use of preloaded Routes, but that's just a quick
guess.
-jiri
SER has always matched against request URI and that's the proper thing
to do. The INVITE above should be processed using Route processing,
as for example in default script.
Testing the same configuration (adjusted for syntax changes) with 0.8.10
and 0.8.12 reveals that the two versions handle the request differently.
With 0.8.10, the request is rewritten and forwarded to 172.21.30.51 with
0.8.12 it is not. 
5:39 a.m.
Audiocodes?
I am playing with a MP-108.
It is an Audiocodes box with 8 FXS ports.
I have it registering, and placing calls (sort of).
However, when I try to call it, it acts like it isn't
seeing the invite....when I call out, it sets up the
call, then acts like it doesn't see the OK:
Here is the OK:
#
U 2003/12/04 21:27:32.334206 216.87.144.203:5060 -> 66.228.53.207:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 66.228.53.207;branch=z9hG4bKacNzOTHDl.
From: Gateway <sip:9723937976@addaline.com>;tag=1c1574720137.
To: <sip:2143357976@addaline.com>;tag=4ADC0734-1EA3.
Date: Fri, 05 Dec 2003 03:27:24 GMT.
Call-ID: 157472013720137YmWO-9723937976--2143357976(a)66.228.53.207.
Server: Cisco-SIPGateway/IOS-12.x.
CSeq: 34108 INVITE.
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER,
SUBSCRIBE, NOTIFY, INFO.
Allow-Events: telephone-event.
Contact: <sip:92143357976@216.87.144.196:5060>.
Record-Route: <sip:2143357976@216.87.144.203;ftag=1c1574720137;lr>.
Content-Type: application/sdp.
Content-Length: 221.
.
v=0.
o=CiscoSystemsSIP-GW-UserAgent 1796 7474 IN IP4 216.87.144.196.
s=SIP Call.
c=IN IP4 216.87.144.196.
t=0 0.
m=audio 18602 RTP/AVP 0 19.
c=IN IP4 216.87.144.196.
a=rtpmap:0 PCMU/8000.
a=rtpmap:19 CN/8000.
a=ptime:20.
The audio codes is completely ignoring it. it gets sent several times
then the gateway gives up. I know the SER server and MP-108 are
communicating, because it REGISTERs fine. It is almost like the MP-108
doesn't like something about the packet so it chooses to ignore it, but
I cannot find a status error message anywhere....
Does anyone have the MP-10X working?
Any ideas?
---gre
7:27 a.m.
New subject: [Serusers] Whoops, audiocodes is the subject....
A little more information I gathered,
the Cisco gateway is sending a packet, the
syslog output for audiocodes is saying:
StackMngr: Message was not Parsed correctly ! Last parsed line was
:AllowHeader :
1070601384 1624 godzillaa-206.august.net ( lgr_stk_mngr)(437 ) !!
[ERROR] The error was = Unexpected input
1070601384 1624 godzillaa-206.august.net ( lgr_stk_mngr)(438 ) !!
[ERROR] The Line of the error was = 9
1070601384 1624 godzillaa-206.august.net ( lgr_stk_mngr)(439 ) !!
[ERROR] The Column of the error was = 10
Doesn't tell me much, except that the error might be around the
Allow: stuff. So, I did notice the allow line is:
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER,
SUBSCRIBE, NOTIFY, INFO.
Kinda long, the audio codes might be having problems, so I did:
replace("Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER,
SUBSCRIBE, NOTIFY, INFO.", "Allow: INVITE, ACK, CANCEL, BYE, OPTIONS.");
as the first line in my script....but I'm still sending the
entire string to the audiocodes. Am I filtering wrong? The replace
statement is almost the first line of my route{} script.
---greg
Greg Fausak wrote:
Audiocodes?
I am playing with a MP-108.
It is an Audiocodes box with 8 FXS ports.
I have it registering, and placing calls (sort of).
However, when I try to call it, it acts like it isn't
seeing the invite....when I call out, it sets up the
call, then acts like it doesn't see the OK:
Here is the OK:
#
U 2003/12/04 21:27:32.334206 216.87.144.203:5060 -> 66.228.53.207:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 66.228.53.207;branch=z9hG4bKacNzOTHDl.
From: Gateway <sip:9723937976@addaline.com>;tag=1c1574720137.
To: <sip:2143357976@addaline.com>;tag=4ADC0734-1EA3.
Date: Fri, 05 Dec 2003 03:27:24 GMT.
Call-ID: 157472013720137YmWO-9723937976--2143357976(a)66.228.53.207.
Server: Cisco-SIPGateway/IOS-12.x.
CSeq: 34108 INVITE.
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER,
SUBSCRIBE, NOTIFY, INFO.
Allow-Events: telephone-event.
Contact: <sip:92143357976@216.87.144.196:5060>.
Record-Route: <sip:2143357976@216.87.144.203;ftag=1c1574720137;lr>.
Content-Type: application/sdp.
Content-Length: 221.
.
v=0.
o=CiscoSystemsSIP-GW-UserAgent 1796 7474 IN IP4 216.87.144.196.
s=SIP Call.
c=IN IP4 216.87.144.196.
t=0 0.
m=audio 18602 RTP/AVP 0 19.
c=IN IP4 216.87.144.196.
a=rtpmap:0 PCMU/8000.
a=rtpmap:19 CN/8000.
a=ptime:20.
The audio codes is completely ignoring it. it gets sent several times
then the gateway gives up. I know the SER server and MP-108 are
communicating, because it REGISTERs fine. It is almost like the MP-108
doesn't like something about the packet so it chooses to ignore it, but
I cannot find a status error message anywhere....
Does anyone have the MP-10X working?
Any ideas?
---gre
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
11:31 a.m.
On Thu, 4 Dec 2003, Greg Fausak wrote:
Audiocodes?
I am playing with a MP-108.
It is an Audiocodes box with 8 FXS ports.
[...]
The audio codes is completely ignoring it. it gets
sent several times
then the gateway gives up. I know the SER server and MP-108 are
communicating, because it REGISTERs fine. It is almost like the MP-108
doesn't like something about the packet so it chooses to ignore it, but
I cannot find a status error message anywhere....
Does anyone have the MP-10X working?
I've been testing an MP-104 (4 fxo ports) and they don't really work very
well... Audiocodes has sent me 4 different software versions after fixing
some reported bugs and with the last version you can register and place calls
but still not very stable.
The software version is:
Boot file name = ramMP108_SIP_873.cmp
Saludos
JesusR.
-------------------------------
Jesus Rodriguez
VozTelecom Sistemas, S.L.
jesusr(a)voztele.com
http://www.voztele.com
Tel. 902360305
-------------------------------
4:39 p.m.
New subject: [Serusers] Audiocodes (was mis-titled)
Jesus,
Thanks for the input!
I worked with the AudioCodes for several hours
yesterday.
It has some problems.
I can get it to register, but I cannot get it to originate
or terminate a phone call to my PSTN gateway. It does
originate calls to other UA devices.
My debugging session included turning on the syslog
feature of the MP-108. Here is an example of the
error I see in the log:
2003.12.04 22:35:15 godzillaa-206 ( lgr_stk_mngr)(234 ) !! [ERROR] StackMngr:
Message was not Parsed correctly ! Last parsed line was :AllowHeader :
2003.12.04 22:35:15 godzillaa-206 ( lgr_stk_mngr)(235 ) !! [ERROR] The error was =
Unexpected input
2003.12.04 22:35:15 godzillaa-206 ( lgr_stk_mngr)(236 ) !! [ERROR] The Line of the
error was = 9
2003.12.04 22:35:15 godzillaa-206 ( lgr_stk_mngr)(237 ) !! [ERROR] The Column of
the error was = 10
I also captured the packets as they went by, here is
the packet that triggered the message above:
#
U 2003/12/04 22:35:15.205420 216.87.144.196:5060 -> 216.87.144.203:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 216.87.144.203;branch=z9hG4bKf819.798ade57.0,SIP/2.0/UDP
66.228.53.207;branch=z9hG4bKacRuFEzdB.
From: Gateway <sip:9723937976@66.228.53.207>;tag=1c-1376639640.
To: <sip:9723236598@216.87.144.203>;tag=4B1A0D04-1586.
Date: Fri, 05 Dec 2003 04:35:09 GMT.
Call-ID: 291832765627656NWXe-9723937976--9723236598(a)66.228.53.207.
Server: Cisco-SIPGateway/IOS-12.x.
CSeq: 15138 INVITE.
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO.
Allow-Events: telephone-event.
Contact: <sip:99723236598@216.87.144.196:5060>.
Record-Route: <sip:9723236598@216.87.144.203;ftag=1c-1376639640;lr>.
Content-Type: application/sdp.
Content-Length: 219.
.
v=0.
o=CiscoSystemsSIP-GW-UserAgent 676 309 IN IP4 216.87.144.196.
s=SIP Call.
c=IN IP4 216.87.144.196.
t=0 0.
m=audio 18862 RTP/AVP 0 19.
c=IN IP4 216.87.144.196.
a=rtpmap:0 PCMU/8000.
a=rtpmap:19 CN/8000.
a=ptime:20.
I do not know how AudioCodes counts, from 0 or 1, but, if the counting
is from zero then line ten is:
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO.
This is produced by my Cisco 7206 VXR with 12.3.1(a) IOS.
This looks to be valid. This packet is the result of an internal
INVITE by the AudioCodes (can't make an outbound call). The same
thing happens with an incoming call from the PSTN.
---greg
Jesus Rodriguez wrote:
On Thu, 4 Dec 2003, Greg Fausak wrote:
Audiocodes?
I am playing with a MP-108.
It is an Audiocodes box with 8 FXS ports.
[...]
The audio codes is completely ignoring it. it
gets sent several times
then the gateway gives up. I know the SER server and MP-108 are
communicating, because it REGISTERs fine. It is almost like the MP-108
doesn't like something about the packet so it chooses to ignore it, but
I cannot find a status error message anywhere....
Does anyone have the MP-10X working?
I've been testing an MP-104 (4 fxo ports) and they don't really work very
well... Audiocodes has sent me 4 different software versions after fixing
some reported bugs and with the last version you can register and place calls
but still not very stable.
The software version is:
Boot file name = ramMP108_SIP_873.cmp
Saludos
JesusR.
-------------------------------
Jesus Rodriguez
VozTelecom Sistemas, S.L.
jesusr(a)voztele.com
http://www.voztele.com
Tel. 902360305
-------------------------------
7658
days inactive
7659
days old
10 comments
5 participants
participants (5)
-
Andrei Pelinescu-Onciul -
Greg Fausak -
Jamin W. Collins -
Jesus Rodriguez -
Jiri Kuthan