[OpenSIPS-Users] max branches count error
HI,all:
We try to make a call to a server and try as many as possible, but we find we
can try around 10 times then errors happen (we expect 20 times):
ERROR: tm:add_uac: maximum number of branches exceeded
ERROR: tm:t_forward_nonack: failure to add branches
ERROR: tm:w_t_relay: t_forward_nonack failed
route[1]
{
$avp(i)=0;
t_on_failure("1");
t_relay("1.2.3.4");
exit;
}
failure_route[1]
{
if ( $avp(i) < 20 ) {# max try 20 times
xlog( " i=$avp(i) \n") ;
$avp(i)=$avp(i)+1;
remove_branch(1);
t_on_failure("1");
t_relay("1.2.3.4");
}
exit;
}
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Re: [OpenSIPS-Users] stop accounting
Bogdan, A call with a BYE but no 200 ok , I take the internet away so UA never receive BYE to generate 200 ok. br, kaiser Bogdan-Andrei Iancu 提到: > Hi, > > Just to clarify - you simulated a call without no BYE at all (no > request), or a call with a BYE but no 200 OK for the BYE ? > > Regards, > Bogdan > > kaiser wrote: >> Dear sir, >> >> The opensips script coded with radius accouting, >> if the call is terminated with BYE and 200 OK, radius stop accouinging >> will be sent out. >> >> I try a call without 200 OK for BYE, I made an UA discoonected with LAN. >> >> acc module seems fail in radius accouting, no radius packert send out. >> How do I fix it? >> >> >> kk >> >> >> ___ >> Users mailing list >> Users@lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] stop accounting
Thanks, I have used mediaproxy at the same time, why dom't I get benefit of it? (of course with dialog module) kk Norman Brandinger 提到: > One solution is to take a look at mediaproxy from AG Projects: > http://mediaproxy.ag-projects.com > > Regards, > Norm > > kaiser wrote: >> Dear sir, >> >> The opensips script coded with radius accouting, >> if the call is terminated with BYE and 200 OK, radius stop accouinging >> will be sent out. >> >> I try a call without 200 OK for BYE, I made an UA discoonected with LAN. >> >> acc module seems fail in radius accouting, no radius packert send out. >> How do I fix it? >> >> >> kk >> >> >> ___ >> Users mailing list >> Users@lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] stop accounting
Dear sir, The opensips script coded with radius accouting, if the call is terminated with BYE and 200 OK, radius stop accouinging will be sent out. I try a call without 200 OK for BYE, I made an UA discoonected with LAN. acc module seems fail in radius accouting, no radius packert send out. How do I fix it? kk ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] replace_from restore redundant from
Bogdan,
For test, I do a simple script, get the same result ( 2 from hdr)
There are several UA register in proxy, 40009,40010,40011,40012
When on-net call fail, serial call start ...
Don't answer first shoot, I answer the 2nd shoot and get 2 from hdr
modparam("uac","from_restore_mode","auto")
modparam("tm", "fr_inv_timer", 6)
route {
$var(c)=0;
if (method=="INVITE") {
lookup("location");
t_on_failure("3");
t_relay();
exit;
}
}
failure_route[3]
{
log(1,"*** I'm failure_route[3] ***\n");
if ($var(c)<5)
{
# append_branch();
switch($var(c))
{
case 0:
rewriteuser("40009");
break;
case 1:
rewriteuser("40011");
break;
case 2:
rewriteuser("40012");
break;
case 3:
rewriteuser("40013");
break;
default:
log("process other requests here\n");
}
$var(c)=$var(c)+1;
lookup("location");
append_branch();
t_on_failure("3");
route(31);
t_relay();
#exit;
}
}
route[31]
{
xlog("change from sip:$...@$fd\n");
uac_replace_from("sip:9...@$fd"); # A call B, than forking to C,
C will got B callerid.
}
the incoming 200 ok looks ok
To: "40010";tag=411c5202
SIP Display info: "40010"
SIP to address: sip:40...@63.26.2.2
SIP tag: 411c5202
From: "kaiser";tag=4894c61e
SIP Display info: "kaiser"
SIP from address: sip:9...@63.26.2.2
SIP tag: 4894c61e
Call-ID: MWJmOTM0MTM4ZjY4NzlkNDEyM2U2NzE0YzIyZjlhNGI.
CSeq: 1 INVITE
Sequence Number: 1
reply after restore from:
To: "40010";tag=411c5202
SIP Display info: "40010"
SIP to address: sip:40...@63.26.2.2
SIP tag: 411c5202
From: "kaiser";tag=4894c61e
SIP Display info: "kaiser"
SIP from address: sip:40...@63.26.2.2
SIP tag: 4894c61e
From: "kaiser";tag=4894c61e
SIP Display info: "kaiser"
SIP from address: sip:40...@63.26.2.2
SIP tag: 4894c61e
Call-ID: MWJmOTM0MTM4ZjY4NzlkNDEyM2U2NzE0YzIyZjlhNGI.
CSeq: 1 INVITE
Sequence Number: 1
kaiser
Bogdan-Andrei Iancu 提到:
> Hi Kaiser,
>
> kaiser wrote:
>> Bogdan,
>>
>> We call the uac_replace_from() in failure_route(none in request route),
>> and this failure_route will call itself again if the t_relay() in
>> failure_route fail again.
> Ok, correct
>>
>> If line 10 t_relay() fail, opensips will raise a failure_route(3)
>> again to get next serial call target.
>> the new branch will be added for next call (line 6), so the
>> uac_replace_from will be called again but in different branch.
>> It means line 8 will be run many time, but in different branch
> right - logically speaking is ok - let me try this scenario to see
> what I get.
>
> Regards,
> Bogdan
>>
>>
>> 1 failure_route[3]
>> 2 {
>> 3 log(1,"*** I'm failure_route[3] ***\n");
>>
>> 4 if (avp_pushto("$ru", "$avp(forward)"))
>> 5 {
>> 6 append_branch();
>> 7 avp_delete("$avp(forward)");
>> 8 uac_replace_from("sip:1...@$fd"); # change from header
>> 9 t_on_failure("3"); # try next , recursive
>>
>> 10 t_relay();
>>
>> 11 }
>>
>> 12 }
>>
>>
>> call flow summary,
>> caller make a call to proxy, proxy try called party 40012,
>> 40010,10011 in serial if no answer.
>> called party 3 answer the call, and opensips restore from hdr in 200
>> OK with 2 from header
>>
>>
>> caller to proxy Request: INVITE sip:40...@63.26.2.3, with session
>> description
>> proxy to caller Status: 100 Trying..
>> proxy to caller Status: 180 Ringing
>> *** called party timer out , no answer, proxy cancel the call
>> proxy to called1 Request: CANCEL
>> sip:40...@59.10.208.208:9805;rinstance=c2b3a16f215cf07e
>>
>> *** proxy run to failure route send a new call to a new number with
>> uac_replac
Re: [OpenSIPS-Users] replace_from restore redundant from
Bogdan,
We call the uac_replace_from() in failure_route(none in request route),
and this failure_route will call itself again if the t_relay() in
failure_route fail again.
If line 10 t_relay() fail, opensips will raise a failure_route(3) again
to get next serial call target.
the new branch will be added for next call (line 6), so the
uac_replace_from will be called again but in different branch.
It means line 8 will be run many time, but in different branch
1failure_route[3]
2{
3log(1,"*** I'm failure_route[3] ***\n");
4if (avp_pushto("$ru", "$avp(forward)"))
5{
6append_branch();
7avp_delete("$avp(forward)");
8uac_replace_from("sip:1...@$fd"); # change from header
9t_on_failure("3"); # try next , recursive
10 t_relay();
11 }
12 }
call flow summary,
caller make a call to proxy, proxy try called party 40012, 40010,10011
in serial if no answer.
called party 3 answer the call, and opensips restore from hdr in 200 OK
with 2 from header
caller to proxy Request: INVITE sip:40...@63.26.2.3, with session
description
proxy to caller Status: 100 Trying..
proxy to caller Status: 180 Ringing
*** called party timer out , no answer, proxy cancel the call
proxy to called1 Request: CANCEL
sip:40...@59.10.208.208:9805;rinstance=c2b3a16f215cf07e
*** proxy run to failure route send a new call to a new number with
uac_replace_from
proxy to called2 SIP/SDP Request: INVITE sip:40...@63.26.2.3, with
session description
called2 to proxy SIP Status: 100 Giving a try
called2 to proxy SIP Status: 180 Ringing
*** called party timer out , no answer, proxy cancel the call
proxy to caller Request: CANCEL sip:40...@63.26.2.3
run failure_route again, send the other call to called3 , change
from again
proxy to called3 Request: INVITE sip:40...@63.26.2.3, with session
description
called3 to proxy Status: 180 Ringing
proxy to callerSIP Status: 180 Ringing
*** called party answer the call , 200 OK return from called party
called3 to proxy SIP/SDP Status: 200 OK, with session description
proxy to callerSIP/SDP Status: 200 OK, with session description
the 200 OK looks like this
Status-Line: SIP/2.0 200 OK
Via: SIP/2.0/UDP
63.28.2.7:5060;branch=z9hG4bK58da20a7bfb4a3161fe339136a58343d
Record-Route:
From: "k100" ;tag=3449301004-70991
From: "k100" ;tag=3449301004-70991
To: ;tag=as7b56bdb1
Call-ID: 214351-3449301004-70...@msx1.mydomain.com
CSeq: 1 INVITE
User-Agent: IPPBX
Bogdan-Andrei Iancu 提到:
> Hi Kaiser,
>
> you mean you call twice the uac_replace_from() in failure_route for
> the same request? or for different ones?
>
> because for a request/branch, you must call only once the
> uac_replace_from() function.
>
> Regards,
> Bogdan
>
> kaiser wrote:
>> Bogdan,
>>
>> After calling twice uac_replace_from() in failure route
>>
>>
>> the 200 OK reply restores the from header as
>>
>> Status-Line: SIP/2.0 200 OK
>>
>> Via: SIP/2.0/UDP
>> 63.28.2.7:5060;branch=z9hG4bK58da20a7bfb4a3161fe339136a58343d
>> Record-Route:
>>
>> From: "k100" ;tag=3449301004-70991
>> From: "k100" ;tag=3449301004-70991
>> To: ;tag=as7b56bdb1
>> Call-ID: 214351-3449301004-70...@msx1.mydomain.com
>> CSeq: 1 INVITE
>> User-Agent: IPPBX
>>
>>
>> There are 2 FROM hd in this message which from hdr restored.
>>
>> I try to find a way to output a trace in a compact format, it's long...
>>
>> best regards
>> kaiser
>>
>>
>> Bogdan-Andrei Iancu 提到:
>>> Hi Kaiser,
>>>
>>> kaiser wrote:
>>>> Bogdan,
>>>>
>>>> We have tested several ways:
>>>>
>>>> 1. use uac_replace_from in failure route only:
>>>> this make redundant "from" header in reply 200 OK.
>>> so you use it only in failure_route, this should be ok - what do you
>>> mean by redundant FROM hdr? can you post a trace ?
>>>>
>>>> 2. use uac_replace_from in request route and failure route:
>>>> It makes outgoping INVITE with strange from header.
>>> yes, this is is not correct. When doing uac_replace_from from
>>> request route, you do the change for all future branches.
>>>
>>> Regards,
>>> Bogdan
>>>>
>>>> br
>>>> kk
>>>>
>>>>
>>>> Bogdan-Andrei Iancu 提到:
>>>>> Hi KK,
>>>>>
>>>>> do you do uac_replace_from() in other routes than failure_route()
>>>>> ? like in request route when you process the request for the first
>&g
Re: [OpenSIPS-Users] replace_from restore redundant from
Bogdan, If I use uac_replace_from in a route already, how do I change from in further failure_route call handling? Or I have use a B2BUA for this purpose? best regards kaiser >> >> 2. use uac_replace_from in request route and failure route: >> It makes outgoping INVITE with strange from header. > yes, this is is not correct. When doing uac_replace_from from request > route, you do the change for all future branches. > > Regards, > Bogdan >> > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] replace_from restore redundant from
Bogdan,
After calling twice uac_replace_from() in failure route
the 200 OK reply restores the from header as
Status-Line: SIP/2.0 200 OK
Via: SIP/2.0/UDP
63.28.2.7:5060;branch=z9hG4bK58da20a7bfb4a3161fe339136a58343d
Record-Route:
From: "k100" ;tag=3449301004-70991
From: "k100" ;tag=3449301004-70991
To: ;tag=as7b56bdb1
Call-ID: 214351-3449301004-70...@msx1.mydomain.com
CSeq: 1 INVITE
User-Agent: IPPBX
There are 2 FROM hd in this message which from hdr restored.
I try to find a way to output a trace in a compact format, it's long...
best regards
kaiser
Bogdan-Andrei Iancu 提到:
> Hi Kaiser,
>
> kaiser wrote:
>> Bogdan,
>>
>> We have tested several ways:
>>
>> 1. use uac_replace_from in failure route only:
>> this make redundant "from" header in reply 200 OK.
> so you use it only in failure_route, this should be ok - what do you
> mean by redundant FROM hdr? can you post a trace ?
>>
>> 2. use uac_replace_from in request route and failure route:
>> It makes outgoping INVITE with strange from header.
> yes, this is is not correct. When doing uac_replace_from from request
> route, you do the change for all future branches.
>
> Regards,
> Bogdan
>>
>> br
>> kk
>>
>>
>> Bogdan-Andrei Iancu 提到:
>>> Hi KK,
>>>
>>> do you do uac_replace_from() in other routes than failure_route() ?
>>> like in request route when you process the request for the first time?
>>>
>>> Regards,
>>> Bogdan
>>>
>>> Gentrice's kaiser wrote:
>>>> Dear Sir:
>>>>
>>>>
>>>> I wrote a failure route for serial call mechanism, it help for no
>>>> answer call ...
>>>> but all these serial calls need to have a new CallerID,
>>>> so I use replace_from for this purpose.
>>>>
>>>>
>>>>
>>>> modparam("uac","from_restore_mode","auto")
>>>>
>>>>
>>>> failure_route[3]
>>>> {
>>>> log(1,"*** I'm failure_route[3] ***\n");
>>>>
>>>> if (avp_pushto("$ru", "$avp(forward)"))
>>>> {
>>>> append_branch();
>>>> avp_delete("$avp(forward)");
>>>> uac_replace_from("sip:1...@$fd"); # change from header
>>>> t_on_failure("3"); # try next , recursive
>>>>
>>>> t_relay();
>>>>
>>>> }
>>>>
>>>> }
>>>>
>>>>
>>>>
>>>> If only one forward number in AVP array , this code working fine,
>>>> the issue will happen if multiple forward number trying, ( forward
>>>> 2nd,3rd ...number)
>>>> the UAC module will restore from header in 200 OK of its following
>>>> message when a UA answered,
>>>> we will have a duplicated "from header" in 200 OK (sending out
>>>> opensips).
>>>> It means we will have 2 "from" headers (in 200 OK) to caller, if
>>>> 2nd forwarded callee answer.
>>>>
>>>> Caller ---> Invite ---> opensips > UA 0, no answer
>>>> > UA 1, no answer
>>>> >UA 2, answer
>>>> <200 OK
>>>> <---200OK<---
>>>>
>>>> this 200ok will have 2 "from" header.
>>>>
>>>>
>>>>
>>>> Most of UA just ignore redundant from header, but some can not.
>>>> I think it should be UAC modules issue, anyone can help me?
>>>>
>>>> best regards
>>>> KK
>>>>
>>>>
>>>>
>>>>
>>>> ___
>>>> Users mailing list
>>>> Users@lists.opensips.org
>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>
>>>
>>
>
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Re: [OpenSIPS-Users] replace_from restore redundant from
Bogdan,
We have tested several ways:
1. use uac_replace_from in failure route only:
this make redundant "from" header in reply 200 OK.
2. use uac_replace_from in request route and failure route:
It makes outgoping INVITE with strange from header.
br
kk
Bogdan-Andrei Iancu 提到:
> Hi KK,
>
> do you do uac_replace_from() in other routes than failure_route() ?
> like in request route when you process the request for the first time?
>
> Regards,
> Bogdan
>
> Gentrice's kaiser wrote:
>> Dear Sir:
>>
>>
>> I wrote a failure route for serial call mechanism, it help for no
>> answer call ...
>> but all these serial calls need to have a new CallerID,
>> so I use replace_from for this purpose.
>>
>>
>>
>> modparam("uac","from_restore_mode","auto")
>>
>>
>> failure_route[3]
>> {
>> log(1,"*** I'm failure_route[3] ***\n");
>>
>> if (avp_pushto("$ru", "$avp(forward)"))
>> {
>> append_branch();
>> avp_delete("$avp(forward)");
>> uac_replace_from("sip:1...@$fd"); # change from header
>> t_on_failure("3"); # try next , recursive
>>
>> t_relay();
>>
>> }
>>
>> }
>>
>>
>>
>> If only one forward number in AVP array , this code working fine,
>> the issue will happen if multiple forward number trying, ( forward
>> 2nd,3rd ...number)
>> the UAC module will restore from header in 200 OK of its following
>> message when a UA answered,
>> we will have a duplicated "from header" in 200 OK (sending out
>> opensips).
>> It means we will have 2 "from" headers (in 200 OK) to caller, if 2nd
>> forwarded callee answer.
>>
>> Caller ---> Invite ---> opensips > UA 0, no answer
>> > UA 1, no answer
>> >UA 2, answer
>> <200 OK
>> <---200OK<---
>>
>> this 200ok will have 2 "from" header.
>>
>>
>>
>> Most of UA just ignore redundant from header, but some can not.
>> I think it should be UAC modules issue, anyone can help me?
>>
>> best regards
>> KK
>>
>>
>>
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>
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[OpenSIPS-Users] replace_from restore redundant from
Dear Sir:
I wrote a failure route for serial call mechanism, it help for no
answer call ...
but all these serial calls need to have a new CallerID,
so I use replace_from for this purpose.
modparam("uac","from_restore_mode","auto")
failure_route[3]
{
log(1,"*** I'm failure_route[3] ***\n");
if (avp_pushto("$ru", "$avp(forward)"))
{
append_branch();
avp_delete("$avp(forward)");
uac_replace_from("sip:1...@$fd"); # change from
header
t_on_failure("3");# try next , recursive
t_relay();
}
}
If only one forward number in AVP array , this code working fine,
the issue will happen if multiple forward number trying, ( forward
2nd,3rd ...number)
the UAC module will restore from header in 200 OK of its following
message when a UA answered,
we will have a duplicated "from header" in 200 OK (sending out
opensips).
It means we will have 2 "from" headers (in 200 OK) to caller, if 2nd
forwarded callee answer.
Caller ---> Invite ---> opensips > UA 0, no answer
> UA 1, no answer
>UA 2, answer
<200 OK
<---200OK<---
this 200ok will have 2 "from" header.
Most of UA just ignore redundant from header, but some can not.
I think it should be UAC modules issue, anyone can help me?
best regards
KK
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