* You are viewing the archive for February 2nd, 2011

Regarding bob-invite-alice xml scenario

Hi all,
I have written two xml files one for bob-invites-client.xml and
other is bob-invites-alice-server.xml. when i run these two files with
asterisk server i got successful message till bye towards client and from
client bye message is not going to server and from server 200 ok message is
not coming to client……..

can any one suggest me some helpp please i am in deep trouble……………

I am attaching my files and wire shark trace plz go throw with it and inform
me where i did wrong..

clientxml:



INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: bob [local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
To: alice
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: [len]

v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [media_port] RTP/AVP 0
a=rtpmap:0 PCMU/8000

]]>





ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: bob [local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
To: alice [peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0
]]>


BYE sip:[service]@[remote_ip]:[remote_port] SIP/2.
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: bob [local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
To: alice [peer_tag_param]
Call-ID: [call_id]
CSeq: 2 BYE
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0

]]>




SERVER XML:




SIP/2.0 180 Ringing
[last_Via:]
[last_From:]
[last_To:];tag=[pid]SIPpTag01[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact:
Content-Length: 0
]]>


SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:];tag=[pid]SIPpTag01[call_number]
[last_Call-ID:]
[last_CSeq:]
Contact:
Content-Type: application/sdp
Content-Length: [len]

v=0
o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[media_ip_type] [media_ip]
t=0 0
m=audio [media_port] RTP/AVP 0
a=rtpmap:0 PCMU/8000
]]>




SIP/2.0 200 OK
[last_Via:]
[last_From:]
[last_To:]
[last_Call-ID:]
[last_CSeq:]
Contact:
Content-Length: 0

]]>



Problems using Background within a macro on V 1.4

Hi List

I have had a look at the various posts on this and seem to be more confused
than ever – but then again that’s not hard ;-)

I am using Version 1.4.33.1 build from the Debian “lenny” distros

I am trying to implement a simple screening

[macro-screen]
exten => s,1,Background(press1)
exten => s,n,WaitExten(5)
exten => 1,1,NoOp(accepted) ; Dont set a reply so dial connects
exten => i,1,Set(MACRO_RESULT=CONTINUE)
exten => t,1,Set(MACRO_RESULT=CONTINUE)

[test]
exten => _1234,1,Dial(SIP/2001,10,M(screen))
exten => _1234,n,playback(sorry,noanswer)

This config plays to the caller “sorry” if I don’t answer SIP/2001 – good
and if I answer SIP/2001 but don’t press ANY key (so timeout) the caller
gets the sorry message – again good.

The problem I have is that the call gets connected whatever key I press -
not just the “1″ key.

I have seen various posts about using background within macros e.g.
http://www.voip-info.org/wiki/view/Asterisk+cmd+BackGround

but changing the macro to

[macro-screen]
exten => s,1,Background(press1,,screen)
….

or – I am not clear what the name should be

[macro-screen]
exten => s,1,Background(press1,,macro-screen)
….

just produces the same result – If I can get this working it will be part of
a bigger dialplan for which the followme app is not suitable hence the back
to basics

can anybody point me in the right direction – pleeese :-)

Paddy

Regarding asterisk

Hi every one,
I am using asterisk version 1.6.2…… i did not
install mysql data base and when i tried to register a client from SIPp xml
file….. it is registered….

My questions are 1. where can i find that registered client?

2. when i type the command “core stop now” it exists and the registered
users are not shown why this is happening?

3. Is it compulsary to have a mysql database link up with the asterisk
server in order to register 2 or 3 users and invite them to the session.?

Host dnsmgr Username Refresh
State Reg.Time
myserver.com:5060 N bob 105
Registered Wed, 02 Feb 2011 21:41:15
1 SIP registrations.

can anyone tell me what the result is? what is 105? etc….

4. After registering the 2 clients i started placing calls to one of them
client from other one… but i am getting error as discarding message and
404 not found…..

can anyone give me correct and exact information and invite xml fileee for
creating session.

BR
viswavardhan.

asterisk18 rpm issues

Hi there,

Per the instruction from http://www.asterisk.org/downloads/yum , I
setup the yum repository on my Centos 5 x86_64 machine and did a

yum install asterisk18 asterisk18-configs

then I startup the asterisk (with no changes to config) just to see if
it runs, but see below errors in the /var/log/asterisk/messages:

[Jan 31 11:41:40] WARNING[2899] loader.c: Error loading module
‘res_pktccops’: /usr/lib/asterisk/modules/res_pktccops.so: cannot open
shared object file: No such file or directory
[Jan 31 11:41:40] WARNING[2899] loader.c: Error loading module
‘chan_mgcp.so’: /usr/lib/asterisk/modules/chan_mgcp.so: undefined
symbol: ast_pktccops_gate_alloc

I checked the system and can’t find the file
/usr/lib/asterisk/modules/res_pktccops.so at all. I double checked the
rpm file downloaded by yum and “res_pktccops.so” is not in any rpms.

Since yum repo only has asterisk 1.8.2.2, but it seems 1.8.2.3 is
already out. So I tried to rebuild from 1.8.2.3 source and see if that
will include “res_pktccops.so” using the same specs file from the
1.8.2.2 source rpm but replacing the source tar file. First I noticed
is the new 1.8.2.3 tar file I downloaded is much bigger in size than
the 1.8.2.2 source tar file within the original source rpm. Are there
scripts or steps to “shrink” the regular source tar.gz to be used in
the src rpm?

Anyway, it seems the rpm I rebuilt using the latest 1.8.2.3 also
doesn’t have the “res_pktccops.so” either.

Any thoughts?

Thanks!
Frank

Outgoing agent

Hi, is there any way to manage outgoing calls from agents?

Mi agents are answering in pstn lines. I can send agents outgoing calls to
my Asterisk but I don’t know wich agent is making the call…because, may be
he is unregister…
Is there any solution?

Thanks

SIP Originate on 1.8.X

I am having a problem trying to use originate from the CLI on Asterisk
1.8.2.3. The SIP peer is defined correctly and it works if I dial using
my IP phone. When I try to dial from the CLI I get this message:

pbxoficina*CLI> originate SIP/protel-out/0445540881644 application playback tt-monkeys
[Jan 18 12:00:09] WARNING[3336]: chan_sip.c:19048
handle_response_invite: Received response: “Forbidden” from ‘”Anonymous”
;tag=as67d9024d’

This same peer works fine in Asterisk 1.6.2.X so I guess something need
to be modified for 1.8? So far it has failed in all 1.8.X versions. Here is the sip.conf definition:

[protel-out]
fromdomain=i2next.com.mx
defaultuser=123456789
secret=secret
fromuser=123456789
type=peer
host=i2next.com.mx
qualify=no
insecure=port,invite
directmedia=no
disallow=all
allow=g729
allow=ulaw

This same peer works when used from a dial command in the dial plan. I guess the problem has something to do that originate is not
setting the proper domain and it is trying to send the call as anonymous. Should I open a bug for this?