Regarding bob-invite-alice xml scenario

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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:

Asterisk Users 4.7 years ago 0 Answers

Problems using Background within a macro on V 1.4

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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 -…

Asterisk Users 4.7 years ago 1 Answer

Regarding asterisk

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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…

Asterisk Users 4.7 years ago 1 Answer

asterisk18 rpm issues

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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

Asterisk Users 4.7 years ago 3 Answers

Outgoing agent

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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

Asterisk Users 4.7 years ago 0 Answers

SIP Originate on 1.8.X

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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]

Asterisk Users 4.7 years ago 0 Answers

AGI script exits non-zero when running system command

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Hey guys I was hoping I could get a few pointers on a problem I have been
trying to debug for the last couple of months regarding asterisk AGI scripts
and unexpected termination.
I have this agi script that accepts incoming faxes using RxFax on the latest
asterisk 1.4 branch. Its written with perl and it works fine except for one
line that causes the entire script to terminate unexpectedly. The script always terminates at the point where I use the 'system' command
or backticks to run a system command.
Example:
system(…

Asterisk Users 4.7 years ago 2 Answers

regarding sip.conf and extensions.conf

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Hi all,
My experiment scenario is like this: SIPp Uac -----------------------------> ASTERISK
SERVER---------------------------------->SIPp uas
1. when i had registered bob with this command ./sipp -sf
register_client.xml -inf register1.csv -i 192.168.1.6:5060 192.168.1.6 -p
5061 -m 10000 it has registered....
If i want to register another client alice with same command but with few
changes in port number ...etc..... The first registered client is erased and
showing second one when i have checked with sip show peers and sip show
registry................... Why is this happening? 2. if i want to register…

Asterisk Users 4.7 years ago 0 Answers