19 Aug
2005
19 Aug
'05
6:29 p.m.
Hi
I know this isn't the best solution but what if you did
!search("User-Agent: Asterisk PBX.*")
that way you could see the request coming from asterisk and process
differently. As for why the 0 does not match, once you sort out the loop
problem you could look at the URI, and setup the debug
Iqbal
Kamran Ahmad wrote:
Hello
i m getting loop detected
UA-------->SER
SER----->asterisk
Asterisk------(back TO SER after adding 0)----->SER
there is if condition that ll check if 0 is added then
dont send back to asterisk but it is not checking that
condition properly and sending request back to
asterisk.
--- Iqbal <iqbal(a)gigo.co.uk> wrote:
loop detected, when the call gets to asterisk
what
are you telling
asterisk to do, send it back to ser, or out
somewhere else.
Calls are not workinf after the change or before.
iqbal
Kamran Ahmad wrote:
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.
i m only sending invite to asterisk one when i
try
bindaddr=0.0.0.0
calls are not working. now i changed sip.conf
bindaddr=0.0.0.0
and in ser.cfg
port=5060
with these changes now register messages are stoped
but still getting 482 "Loop Detected".
--- Iqbal <iqbal(a)gigo.co.uk> wrote:
>Hi
>
>The register messages what username are they for,
>and from what IP
>address, do a sip debug in asterisk for this.
>Also why are you sending register messages to
>asterisk, just send your
>INVITES there.
>
>As for debug run ngrep, to pick up your messages ,
>and see what is going
>on, and to look at the uri problem, remove 2 out
>
>
of
>3 of the routes, and
>see what happens, if call fails then matching is
>
>
not
>correctly being
>done, or the uri is not correct.
>
>iqbal
>
>Kamran Ahmad wrote:
>
>
>
>
>
>>i tried to do debug (put messages in
>>
>>
/tmp/call.log)
>>
>>
>>but there is no Invite having only one zero(like
>>06999786) all call were 0097937223 or 46364736.
>>
>>
>>
>>
>this
>
>
>
>
>>means that only third else part is always active
>>
>>
>>
>>
>but
>
>
>
>
>>if third part is active then there must be some
>>
>>
>>
>>
>invite
>
>
>
>
>>starting with only one zero. it means second time
>>invite call is not comming here .
>>
>>My from main invites are only comming here
>>
>>
>>
>>
>route(3). i
>
>
>
>
>>think all messages are going to asterisk because
>>
>>
>>
>>
>there
>
>
>
>
>>is only one statement in ser.cfg having port 5970
>>(this is for asterisk) and all my register
>>
>>
messages
>>are also going there to asterisk.
>>
>>there are too many register messages on my
>>
>>
>>
>>
>asterisk. i
>
>
>
>
>>dont know why they are comming to asterisk as
>>
>>
this
>>port is not available for any user nobody
uses
>>
>>
this
>>port only ser is routing calls to
asterisk
>>
>>route {
>> if (uri==myself) {
>> if (method=="INVITE") {
>> route(3);
>> break;
>> }
>> }
>>}
>>route[3] {
>> exec_msg("cat >> /tmp/call.log");
>> if(uri=="^sip:00[1-9].*@.*") {
>> route(4);
>> break;
>> } else if (uri=="^sip:0[1-9].*@.*") {
>> strip(1);
>> route(5);
>> break;
>> } else {
>> route(4);
>> break;
>> }
>>
>>}
>>--- Iqbal <iqbal(a)gigo.co.uk> wrote:
>>
>>
>>
>>
>>
>>
>>
>>>dump a call trace showing your uri
>>>
>>>Iqbal
>>>
>>>Kamran Ahmad wrote:
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>>hello
>>>>
>>>>i m using ser-0.9.0 on 5060 and asterisk-1.0.9
>>>>
>>>>
on
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>5970
>>>
>>>
>>>
>>>
>>>
>>>
>>>>but problem is that i cannot send my ser back
>>>>
>>>>
to
>>>>asterisk.
>>>>the problem is that it always goto route(4)
>>>>
>>>>
>>>>
>>>>
>second
>
>
>
>
>>>>else if is not properly checked how to check
>>>>
>>>>
this
>>>>condition. i m adding 0 when
asterisk is
>>>>
>>>>
sending
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>call
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>>back.
>>>>>
>>>>>
>>>>> if(uri=="^sip:00[1-9]+@.*") {
>>>>> route(4);
>>>>> break;
>>>>> } else if (uri=="^sip:0[1-9]+@.*") {
>>>>> strip(1);
>>>>> route(5);
>>>>> break;
>>>>> } else {
>>>>> route(4);
>>>>> break;
>>>>> }
>>>>>
>>>>>
>>>>>
>>>>>
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>>>>>
>>>>>