I have also had problems with getting the ACK back.
I don't completely understand your configuration, you
must allow for packets going both directions, right?
Here is my config :
route
{
# check to see if the message has been around too long
# probably means that it is looping
#
if (!mf_process_maxfwd_header("10"))
{
log("LOG: Too many hops\n");
sl_send_reply("483","Too Many Hops");
break;
};
#
# make sure the length of the message isn't too long!
#
if (len_gt( max_len ))
{
sl_send_reply("513", "Wow -- Message too large");
break;
};
#
# do the loose-routing thing, this is important!
#
if(loose_route())
{
log(1,"doing top loose route");
t_relay();
break;
};
# this is where I was dropping the ACKS.
# I was simply dropping these, but they must be relayed
# because they can be ACKs
if(!(uri==myself))
{
if(!t_relay())
{
sl_reply_error();
break;
};
break;
};
This gets the ACKs through for me.
By the way, I have this configured with Cisco ATAs, version 2.16.
---greg
>
>I have the same problem and posed it to the group yesterday ([Serusers]
>Ignored 200 OK message.) So far the only workaround that I have found is to
>use the rules in my gateway to rewrite the dialed digits before sending them
>to the PSTN PRI, thus leaving the origianl URI intact for SIP
>communications.
>
>One person told me that this is a bug in the Cisco ATA, but it happens on my
>IPDialog phones also. It seems to me that the INVITE is being processed by
>the SER dial rules and is rewritten, but the ACK is not.
>
>Sean
>_______________________________________________
>
>Sean Robertson
>
>NETXUSA
>p. 800-289-6389
>f. 864-233-4344 "Ask me about Voice over IP."
>http://www.netxusa.com/
>
>----- Original Message -----
>From: "Alexander Mayrhofer" <axelm(a)nic.at>
>To: <serusers(a)lists.iptel.org>
>Sent: Friday, June 27, 2003 12:15 PM
>Subject: [Serusers] rewrite & ACK forwarding problem
>
>
>>
>> Hi,
>>
>> we're running SER together with a PSTN Gateway. Before a call get's
>> forwarded to the gateway, we are rewriting the request URI to make
>> rewriting on the GW as simple as possible:
>>
>> route {
>> ...
>> strip(3); # +43xxx -> xxx
>> prefix("0"); # xxx -> 0xxx
>> rewritehostport(xxx.xxx.xxx.xxx, 5060); # request to gateway
>> route(1);
>> break;
>> ...
>>
>> SIP call flow looks like (record route enabled):
>>
>> (1) phone -> SER
>> INVITE sip:*43699xxxxxxxx@nic.at43.at SIP/2.0
>>
>> (2) SER -> phone
>> SIP/2.0 100 trying -- your call is important to us
>>
>> (3) SER -> GW
>> INVITE sip:0699xxxxxxxx@xx.xx.xx.xx:5060 SIP/2.0
>>
>> (4) GW -> SER
>> SIP/2.0 100 Trying
>>
>> (5) GW -> SER
>> SIP/2.0 183 Session Progress
>>
>> (6) SER -> phone
>> SIP/2.0 183 Session Progress
>>
>> (7) GW -> SER
>> SIP/2.0 180 Ringing
>>
>> (8) SER -> phone
>> SIP/2.0 180 Ringing
>>
>> (9) GW -> SER
>> SIP/2.0 200 OK
>> Contact: <sip:0699xxxxxxxx@xx.xx.xx.xx:5060>
>>
>> (10) SER -> phone
>> SIP/2.0 200 OK
>> Contact: <sip:0699xxxxxxx@xx.xx.xx.xx:5060>
>>
>> [ call established, we can talk, but ... ]
>>
>> (11) phone -> SER
>> ACK sip:0699xxxxxxxx@xx.xx.xx.xx:5060 SIP/2.0
>>
>> --> Here starts the problem. That ACK (11) never gets forwarded to the
>> Gateway, so after a few seconds, the GW starts over at (9). Those three
>> packets (9-11) repeat a few times until GW runs into a timeout and drops
>> the call.
>>
>> I have the impression that SER can't match the packet to the previous
>> requests because of the rewritten URI. Is that correct?
>>
>> The only output at debug level 3 is:
>>
>> Warning: sl_send_reply: I won't send a reply for ACK!!
>>
>> Is that a routing goof somewhere in our scripts or is that a more
>> generic problem? Is the problem that the warning indicates somehow
>> related to the fact that the ACK is not being forwarded?
>>
>> Help appreciated.
>>
>> cheers
>>
>> axelm
>>
>> _______________________________________________
>> Serusers mailing list
>> serusers(a)lists.iptel.org
>> http://lists.iptel.org/mailman/listinfo/serusers
>>
>
>
>_______________________________________________
>Serusers mailing list
>serusers(a)lists.iptel.org
>http://lists.iptel.org/mailman/listinfo/serusers
>
Hi List,
i tried on asterisk list as well, but any of you can give any idea abt SIP authenticatin problem with Asterisk and SIP UA (i tried SIPPS and X-Lite)
I could not properly get authenticated with my SIP UA to asterisk. i m using a username for UA "12321" and following are SIP.conf file user params
[12321]
type=friend
username=12321
host=dynamic
secret=ccarta
context=default
mailbox=1234,2345 ; Mailbox for message waiting indicator
[77777]
type=friend
username=77777
host=dynamic
secret=atracc
context=default
mailbox=1234,2345
m trying with SIPPS UA, that gets status "Acquired", not "Registered", Can anyone give any idea about it? I tried same with X-Lite, didnt work.
Sip debug messages are pasted below.
Best Regards,
JF
Sip read:
REGISTER sip:192.168.100.71 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.66:5062;branch=z9hG4bKnp992800961-40de3e5f192.168.100.66
From: <sip:12321@192.168.100.71>;tag=3b2cf0ba
To: <sip:12321@192.168.100.71>
Call-ID: 990209125-415b5d61@990209122-415b5d5e
Contact: ccarta <sip:12321@192.168.100.66:5062>;expires=600;q=0.500
Expires: 600
CSeq: 1 REGISTER
Content-Length: 0
User-Agent: Ahead SIPPS IP Phone Version 2.0.42.13
10 headers, 0 lines
Using latest request as basis request
Sending to 192.168.100.66 : 5062 (non-NAT)
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.100.66:5062;branch=z9hG4bKnp992800961-40de3e5f192.168.100.66
From: <sip:12321@192.168.100.71>;tag=3b2cf0ba
To: <sip:12321@192.168.100.71>;tag=as648287fa
Call-ID: 990209125-415b5d61@990209122-415b5d5e
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:12321@192.168.100.71>
Content-Length: 0
to 192.168.100.66:5062
Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.100.66:5062;branch=z9hG4bKnp992800961-40de3e5f192.168.100.66
From: <sip:12321@192.168.100.71>;tag=3b2cf0ba
To: <sip:12321@192.168.100.71>;tag=as648287fa
Call-ID: 990209125-415b5d61@990209122-415b5d5e
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:12321@192.168.100.71>
Proxy-Authenticate: Digest realm="asterisk", nonce="7c7fba4d"
Content-Length: 0
to 192.168.100.66:5062
Sip read:
REGISTER sip:192.168.100.71 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.66:5062;branch=z9hG4bKnp992804895-411b471d192.168.100.66
From: <sip:12321@192.168.100.71>;tag=3b2d0018
To: <sip:12321@192.168.100.71>
Call-ID: 990209125-415b5d61@990209122-415b5d5e
Contact: ccarta <sip:12321@192.168.100.66:5062>;expires=600;q=0.500
Expires: 600
CSeq: 2 REGISTER
Content-Length: 0
Proxy-Authorization: Digest username="12321",realm="asterisk",uri="sip:192.168.100.66",nonce="7c7fba4d",nc="00000001",response="b8b1d7fc53eff354dfc31dfa3f800749"
User-Agent: Ahead SIPPS IP Phone Version 2.0.42.13
11 headers, 0 lines
Using latest request as basis request
Sending to 192.168.100.66 : 5062 (non-NAT)
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.100.66:5062;branch=z9hG4bKnp992804895-411b471d192.168.100.66
From: <sip:12321@192.168.100.71>;tag=3b2d0018
To: <sip:12321@192.168.100.71>;tag=as648287fa
Call-ID: 990209125-415b5d61@990209122-415b5d5e
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:12321@192.168.100.71>
Content-Length: 0
to 192.168.100.66:5062
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.66:5062;branch=z9hG4bKnp992804895-411b471d192.168.100.66
From: <sip:12321@192.168.100.71>;tag=3b2d0018
To: <sip:12321@192.168.100.71>;tag=as648287fa
Call-ID: 990209125-415b5d61@990209122-415b5d5e
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Expires: 600
Contact: <sip:12321@192.168.100.71>;expires=600
Date: Tue, 14 Oct 2003 13:46:14 GMT
Content-Length: 0
to 192.168.100.66:5062
11 headers, 2 lines
Reliably Transmitting:
NOTIFY sip:12321@192.168.100.66:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.71:5060;branch=z9hG4bK3ecb7a3b
From: "asterisk" <sip:asterisk@192.168.100.71>;tag=as3f6e8c0e
To: <sip:12321@192.168.100.66:5062>
Contact: <sip:asterisk@192.168.100.71>
Call-ID: 074dfadf24b95dd75e17b56c67ffcaf1(a)192.168.100.71
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 36
Messages-Waiting: no
Voicemail: 0/0
(no NAT) to 192.168.100.66:5062
Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.100.71;branch=z9hG4bK3ecb7a3b
From: "asterisk" <sip:asterisk@192.168.100.71>;tag=as3f6e8c0e
To: <sip:12321@192.168.100.66:5062>;tag=3b302259
Call-ID: 074dfadf24b95dd75e17b56c67ffcaf1(a)192.168.100.71
CSeq: 102 NOTIFY
User-Agent: Ahead SIPPS IP Phone Version 2.0.42.13
Content-Length: 0
8 headers, 0 lines
---------------------------------
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At 07:39 PM 8/14/2003, Chad Brown wrote:
>Perfect,
>
>Let me ask 2 quick follow-on questions...
>
>1. Can I go back to http://www.iptel.org/ser/tarball/ser_8_11_stable.tgz
>to get the latest patched and STABLE builds?
That's the latest, most stable source, you need to compile it myself.
There will be a new complete distribution by end of this month -- we are now
waiting to run SER through the upcoming SIPIT to release it.
>2. What location / version of the modules should I use when running
>builds for this location? (Serweb, mysql, jabber, etc)
All SER modules are included there.
uptodate SERWEB is now in the http://www.iptel.org/ser/tarball/ directory too.
-jiri
Hi,
I have radius authentication working thanks to Jan's help but I am still not receiving any accounting messages to my radius server. I found a mention of setting radius_log_flag but ser 0.9.11 tells me it isn't found in the "acc" module. How can I set invite and goodbye messages (or any others for that matter) to send information to my radius server for accounting purposes. I
have read at leat 90% of the old messages but have yet to find an answer.
Thanks in advance,
Steve
I have install ser-0.8.11-0.i386.rpm, mysql and serweb. My sip phone are
all window messenger. When someone contact a user who is not on-line,
you get this message:-
phone(a)192.168.1.3 has declined your request to have a voice
conversation.
When I login 'phone' to serwweb as a subscriber, under missed calls is
always empty & message store too.
How to make it ring first and after a period go to voicemail if
phone(a)192.168.1.3 is not online?
I went through the document but got confuse. Do you have a example
ser.cfg script that works.
2. Through the web documents there are 2 vm softwares
sems-0.1.0-0.i386.rpm & ortp-0.6.2.tar.gz. which is the right one?
3. if vm is required must you reinstall ser? If yes, what to change in
the makefile?
Help! Thanks!
Unfortunately, there is now no standard for use of RADIUS along
with SIP. SER users leveraging the combination of these two
technologies are left with implementation of expired internet
drafts. There are some chances that the IETF community revitalizes
the document.
Thus, I would appreciate hearing if any of the active RADIUS/SIP/SER
users have had any issues with the RADIUS authentication in SER,
which is based on draft-sterman-aaa-sip.
Thanks,
-jiri
--
Jiri Kuthan http://iptel.org/~jiri/
Hai,
We Xten released a newer version (seradmin v .03e) of SERAdmin.
Xten is building this for the ser community and we would
very much like your feedback so that we can build you a better product
The Source code is available at
http://developer.berlios.de/projects/seradmin
SERAdmin is a GUI interface between SIP Express Router (SER) and a SER
administrator.SERAdmin has an intuitive look and feel.
SERAdmin provides control over many SER tasks such as:
Start, Stop, Pause, Re-start, Monitor SER, Add User, Change Password
and EmailId Delete User, Add Alias , Edit Alias etc.
With this we can also use FIFO commands,Access controls,User Location.
The objective of the SERAdminv03e is to support the ser-0.8.11 version also. This latest SER Administration Application handles both ser-0.8.10 and ser-0.8.11.
What is new?.
The following commands available in this release
------------------------------------------------
Ping(uri) - Pinging a URI
Cisco_restart(uri) - Restart a Cisco Phone
arg - Arguments of SER
pwd - Present Working Directory
t_uac_dlg - Initiate a Transaction
you can download the components from
http://developer.berlios.de/projects/seradmin
For any feedback & help pls. contact xten_india(a)yahoo.com
Regards,
Team,
Xten_india
---------------------------------
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hello,
this is maybe a bit silly question, but.
Outbound proxy is needed, when UA is behind FW.
What is the function of this proxy (how can this help the UA to make
calls) and what kind of software can do this.
Can SER be also outbound proxy, or I have to use something else ?
hudecof
--
mail: [phudec(a)postel.sk] www: [http://www.postel.sk]
cellular: [+421 02 50203166] icq: [99518783]
gpg: [http://hudecof.net/data/hudecof.gpg]
Is this latest version of Ser already intergrated on release of latest version......in Gentoo.
Danny
---------------------------------
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