[Asterisk-Users] Strange Problem

2005-12-23 Thread Gulzar Hussain 
Hi All

I am having a strange problem when I call from 1 RTC
Client to another without Asterisk in between
everything use to be fine but when asterisk is there
as a Registrar a problem use to occur in many calls,
Caller can hear the voice of the receiving side
but the receiver cant be able to hear the caller for
exact 8 seconds, conversation will become two
way after 8 seconds but this problem is a big
hurdle in proper establishment of a call

Does anybody ever had this problem ?
Any suggestions will be higly apreciated

i have tried capturing packets but dont find anything
abnormal

Thanx in Advance






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RE: [Asterisk-Users] Recording Volume on Zap Channel

2005-12-18 Thread Gulzar Hussain 

i have tried 
rsgain=100
txgain=100

recording volume improved but still not good

--- Steve Totaro [EMAIL PROTECTED]
wrote:

  
  Hi All
  
  I have a call center working on 8 FXO Channels,
  everything working fine except one little problem,
 I
  am using asterisk queues with
  monitor-format = wav49
  and
  monitot-join = yes
  asterisk is recording all conversations but the
  problem is that the volume of Zap Channel is too
 low
  in most of the calls i am unable to understand
 what
  other person was saying (ZAP Channel) although
 Agent's
  (SIP Channel) vocie use to get recorded pretty
 good.
  
  Any suggession will be higly appreciated
  Thanks in Advance
  
 
 Did you try adjusting the gain in Zapata.conf?
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[Asterisk-Users] Recording Volume on Zap Channel

2005-12-07 Thread Gulzar Hussain 
Hi All

I have a call center working on 8 FXO Channels,
everything working fine except one little problem, I
am using asterisk queues with 
monitor-format = wav49 
and 
monitot-join = yes 
asterisk is recording all conversations but the
problem is that the volume of Zap Channel is too low
in most of the calls i am unable to understand what
other person was saying (ZAP Channel) although Agent's
(SIP Channel) vocie use to get recorded pretty good.

Any suggession will be higly appreciated
Thanks in Advance


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Re: [Asterisk-Users] asterisk with EWSD v16

2005-12-07 Thread Gulzar Hussain 

I am using EWSD's PRIs and I am not having this
problem my configs are

Zaptel.conf

span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
loadzone = us
defaultzone=us

Zapata.conf
[channels]
language=en
context=ext-acd
switchtype=euroisdn
signalling=pri_cpe
echocancel=yes
echocancelwhenbridged=yes
group=1
channel = 1-15
channel = 17-31
pridialplan=private
prilocaldialplan=private
overlapdial=yes
usecallerid=yes
hidecallerid=no
immediate=no
usecallingpres=no



--- Atif Rasheed [EMAIL PROTECTED] wrote:

 if any EWSD guru out there..please help ???
 
  Hello all,
 
  I am running Asterisk with Digium E1 card with
 zaptel, libpri, 
  asterisk cvs v1-2. My server is interfaced with
 EWSD v16 using a PRI 
  on E1. I am running into a problem that at my
 telco's end alot of 
  trunks are getting BPRM (Block permanant) status.
 I am not sure why 
  EWSD is blocking trunks.
 
  config at my end:::
  coding = hdb3
  format = ccs,crc4
  signalling = euroisdn, pri_cpe
 
  config at my telco's end
  coding = hdb3
  format = crc4mf
  signalling = euroisdn, pri_net
 
  Is there any EWSD guru around who can explain why
 trunks are getting 
  BPRM status in EWSD switch. I will really
 appriciate your help
 
  Thank you
  -- 
  Atif Rasheed
 
 
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Re: [Asterisk-Users] Wierd Problem

2005-09-06 Thread Gulzar Hussain 

theres nothing between Lucent Max TNT and Asterisk box

both are using same IP Class network and both are
connected on same switch


--- [EMAIL PROTECTED] wrote:

 On 8/30/2005, Gulzar Hussain [EMAIL PROTECTED]
 wrote:
  Hi All
 
  I have posted this problem many times on the list
 but
  no reply, trying one more time may be someone will
  response this time
 
  When I call from 1 RTC Client to another without
  Asterisk everything use to be fine but when
 asterisk
  is there as a Registrar a problem use to occur in
 more
  than 90% calls, Caller can hear the voice of the
  receiving side but the receiver cant be able to
 hear the
  caller for exactly 12 seconds, conversation will
 become
  two way after 12 seconds.
 
  My Scenario
 
  Lucent Max TNT - Asterisk - RTC Client API
 
  Does anybody ever had this problem ?
  Any suggestions will be higly apreciated
  Thanx in Advance
 
 EXACTLY 12 seconds... Sounds like a timeout. Either
 the TNT or
 the Asterisk box is 'looking' for something - no
 idea which or what.
 DNS? CID? Something.
 
 Brett
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[Asterisk-Users] Wierd Problem

2005-08-30 Thread Gulzar Hussain 
Hi All

I have posted this problem many times on the list but
no reply, trying one more time may be someone will
response this time


When I call from 1 RTC Client to another without
Asterisk everything use to be fine but when asterisk
is there as a Registrar a problem use to occur in more
than 90% calls, Caller can hear the voice of the
receiving side
but the receiver cant be able to hear the caller for
exactly 12 seconds, conversation will become two
way after 12 seconds.

My Scenario

Lucent Max TNT - Asterisk - RTC Client API


Does anybody ever had this problem ?
Any suggestions will be higly apreciated
Thanx in Advance



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Re: [Asterisk-Users] OT: Are you using a Lucent?

2005-08-29 Thread Gulzar Hussain 

Hi 

I am using a Lucent MAX TNT to terminate 11 PRIs and
using a single Asterisk box to handle all calls


--- Andrew Thrift [EMAIL PROTECTED] wrote:

 We have the ability to do this on a large scale, but
 want to do it on a 
 smaller scale for 1 to maybe a maximum of 5 TNT's.
 
 
 Andrew Thrift wrote:
 
  Hi Mathew,
 
  We are interested in doing this too, is it
 possible you can share the 
  information with us?
 
  We are looking at using a TNT MAX to terminate 8
 E1's from the Telco, 
  but we need a way of receiving the SS7 signalling
 and passing it to 
  the TNT's via IPDC or whatever.
 
  Regards,
 
 
 
  Andy
 
  Matthew Boehm wrote:
 
  Is anyone out there using Lucent brand equipment
 to handle an 
  incomming DS3, converting all 672 calls to SIP
 (as G729) and sending 
  those to Asterisk/SER over ethernet?
 
  If you are and are willing to speak to my boss
 about your experiences 
  (over the phone) with it, please contact me off
 list.
 
  We have a possible contract with a local CLEC to
 handle their long 
  distance, and they want to send to us using DS3
 and SS7.
 
  I'm trying to convince my boss to use a $9K
 Lucent, but he wants to 
  spend much more by breaking out the DS3 into
 DS1's and stack up 6 
  asterisk boxes with 1 4-port card in each.
 
  Again, if you are using Lucent and are willing to
 speak to my boss 
  about your experiences, please contact me off
 list so I can setup a 
  call.
 
  Thanks,
  Matthew
 
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Re: [Asterisk-Users] Custom Application For Asterisk

2005-08-29 Thread Gulzar Hussain 

Hi

no i write this application for my custom needs, but
anybody of you can use it or customized it according
to your needs 

cheers


--- Matt Riddell [EMAIL PROTECTED] wrote:

 Gulzar Hussain wrote:
  Hi All
  
  I just completed a custom application for Asterisk
 (i
  m not a C guru so i just copy codes from other
  application and alter according to my needs) 
  
  attached files is the source file
  
  this application is working fine but still i need
 you
  people to give suggestion to improve it
  
  Primary task of this application is to get a
 parameter
  from extensions.conf, query sql server and play a
  files according to the result
 
 Is this GPL?
 
 Is there a site where people can read about it and
 download it?
 
 -- 
 Cheers,
 
 Matt Riddell
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[Asterisk-Users] Which Card to choose

2005-08-25 Thread Gulzar Hussain 
Hi All

I want to terminate as much POTS lines as possible to
my Asterisk Server, please advice me which Card to
choose with accessories

Thanks




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[Asterisk-Users] Custom Application For Asterisk

2005-08-25 Thread Gulzar Hussain 
Hi All

I just completed a custom application for Asterisk (i
m not a C guru so i just copy codes from other
application and alter according to my needs) 

attached files is the source file

this application is working fine but still i need you
people to give suggestion to improve it

Primary task of this application is to get a parameter
from extensions.conf, query sql server and play a
files according to the result

Thanks


(I have changed some code to make my code secure ;) )


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http://mail.yahoo.com #include asterisk/lock.h
#include asterisk/file.h
#include asterisk/pbx.h
#include asterisk/app.h
#include asterisk/say.h
#include stdlib.h
#include unistd.h
#include string.h
#include errno.h
#include stdlib.h
#include stdio.h
#include pthread.h

#include libpq-fe.h
#include asterisk/translate.h
#include asterisk/musiconhold.h
#include asterisk/callerid.h
#include sys/time.h
#include sys/signal.h
#include netinet/in.h

#include sys/types.h
#include asterisk/config.h
#include asterisk/options.h
#include asterisk/channel.h
#include asterisk/module.h
#include asterisk/logger.h
#include ../asterisk.h

#include unistd.h
#include time.h
#include math.h

#include tds.h
#include tdsconvert.h
#include ctype.h

#if !defined(TDS_INT_EXIT) 
#define TDS_PRE_0_62
#warning You have older TDS, you should upgrade!
#endif

#define DATE_FORMAT %Y/%m/%d %T

static char *config = abcd.conf;
static char *tdesc = abcd Application;
static char *app = abcd;
static char *synopsis = abcd New;
static char *descrip =
   abcd-IVR: Requires a user to enter a
   number. and announce number according to query result\n;
  

static char abcdhostname[30] = ;
static char abcddbname[30] = ;
static char abcddbuser[30] = ;
static char abcdpassword[30] = ;
static char abcdcharset[30] = ;
static char abcdlanguage[30] = ;

#define DEFAULTCHARSET iso_1
#define DEFAULTLANGUAGE us_english
static int connected = 0;

static int mssql_connect(void);
static int mssql_disconnect(void);
static int play_file(struct ast_channel *chan, char *filename);

AST_MUTEX_DEFINE_STATIC(tdslock);
static TDSSOCKET *tds;
static TDSLOGIN *login;
static TDSCONTEXT *context;

STANDARD_LOCAL_USER;
LOCAL_USER_DECL;
struct abcd_user {
char moh[80];   
char announce[80];  
char context[80];   
int handled;
time_t start;   
int queuetimeout;   
struct ast_channel *chan;   
struct queue_ent *next; 
};

static int abcd_exec(struct ast_channel *chan, void *data)
{
int retried = 0;
int res = 0;
int res_type;
int tdsret;
int rowtype;
int computeid;
int i;
int sucs = 0;
struct localuser *u;
char mysqlcmd[1024];
char myretnumber[6];
const void *value;
char resulttype[4];
if (!data) {
ast_log(LOG_WARNING, abcd requires an argument (number)\n);
return -1;
}
LOCAL_USER_ADD(u);
ast_mutex_lock(tdslock);
memset(mysqlcmd, 0, sizeof(mysqlcmd));
sprintf(mysqlcmd, Select MyFunction(\'%s\') As result,((char *) 
data));
do {
if (!connected) {
if (mssql_connect())
ast_log(LOG_ERROR, Failed to reconnect to SQL 
database.\n);
else
ast_log(LOG_WARNING, Reconnected to SQL 
database.\n);
retried = 1;
}
if (!connected || (tds_submit_query(tds, mysqlcmd) != 
TDS_SUCCEED))
{
ast_verbose(VERBOSE_PREFIX_3 Failed to query 
database.\n);
mssql_disconnect();
}
} while (!connected  !retried);
if (!connected) {
res = -1;
ast_mutex_unlock(tdslock);
LOCAL_USER_REMOVE(u);
return res;
}
tdsret = tds_process_result_tokens(tds, res_type, NULL);
switch (tdsret) {
case TDS_SUCCEED:
switch(res_type){
case TDS_DONE_RESULT:
break;
case TDS_DONEPROC_RESULT:
break;
case TDS_DONEINPROC_RESULT:
break;
case TDS_ROWFMT_RESULT:
while ((res=tds_process_row_tokens(tds, 
rowtype, computeid))==TDS_SUCCEED) {
for (i=0; 
itds-res_info-num_cols; i++) {
value  =

Re: [Asterisk-Users] Asterisk Zaptel Leading Zero Problem With TE110P

2005-08-24 Thread Gulzar Hussain 
yeah i am using chan_zap and i have tried all
combinations of pridialplan and nationalprefix etc.


--- Peter Svensson [EMAIL PROTECTED] wrote:

 On Sat, 20 Aug 2005, Gulzar Hussain wrote:
 
  I am having another strnage problem :)
  
  When I dialout on any number from asterisk, it use
 to
  add a leading zero in dialed number
  for e.g
  I dial a number 5832876
  and when I check the tracer's result of PSTN
 switch
  that shows me call request for 05832876
  
  thats why I can dial NWD and ISD calls but unable
 to
  dial local numbers
 
 What channel do you use? For chan_zap you may want
 to look at the 
 pridialplan, especially pridialplan=dynamic and the
 nationalprefix etc.
 
 Peter
 
 
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[Asterisk-Users] Asterisk Zaptel Leading Zero Problem With TE110P

2005-08-20 Thread Gulzar Hussain 
Hi All

I am having another strnage problem :)

When I dialout on any number from asterisk, it use to
add a leading zero in dialed number
for e.g
I dial a number 5832876
and when I check the tracer's result of PSTN switch
that shows me call request for 05832876

thats why I can dial NWD and ISD calls but unable to
dial local numbers

Thanks 




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[Asterisk-Users] Disable Call Waiting On SIP User Agents

2005-08-13 Thread Gulzar Hussain 
Hi 

how to disable call waiting on SIP User agents 

(incominglimit=1 is Deprecated , End of life already
announced

no idea how to use setgroup to achieve same
functionality)

Thanks





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[Asterisk-Users] Asterisk

2005-08-10 Thread Gulzar Hussain 
 
 

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http://mail.yahoo.com ---BeginMessage---
Hi 
When I call from 1 RTC Client to another without
Asterisk everything use to be fine but when asterisk
is there as a Registrar a problem use to occur in more
than 90% calls, Caller can hear the voice of the
receiving side
but the receiver cant be able to hear the caller for
exactly 12 seconds, conversation will become two
way after 12 seconds.

My Scenario

Lucent Max TNT - Asterisk - RTC Client API


Does anybody ever had this problem ?
Any suggestions will be higly apreciated
Thanx in Advance






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[Asterisk-Users] Asterisk RTC Client API

2005-08-10 Thread Gulzar Hussain 
 
 

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http://mail.yahoo.com ---BeginMessage---
Hi 
When I call from 1 RTC Client to another without
Asterisk everything use to be fine but when asterisk
is there as a Registrar a problem use to occur in more
than 90% calls, Caller can hear the voice of the
receiving side
but the receiver cant be able to hear the caller for
exactly 12 seconds, conversation will become two
way after 12 seconds.

My Scenario

Lucent Max TNT - Asterisk - RTC Client API


Does anybody ever had this problem ?
Any suggestions will be higly apreciated
Thanx in Advance






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[Asterisk-Users] Asterisk Call Queue Application

2005-08-10 Thread Gulzar Hussain 
Hi

I want Queue Application not to call those agents who
are busy talking 

is it possible ?


Thanks




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[Asterisk-Users] RTC Client API Asterisk

2005-08-08 Thread Gulzar Hussain 
Hi 
When I call from 1 RTC Client to another without
Asterisk everything use to be fine but when asterisk
is there as a Registrar a problem use to occur in more
than 90% calls, Caller can hear the voice of the
receiving side
but the receiver cant be able to hear the caller for
exactly 12 seconds, conversation will become two
way after 12 seconds.

My Scenario

Lucent Max TNT - Asterisk - RTC Client API


Does anybody ever had this problem ?
Any suggestions will be higly apreciated
Thanx in Advance






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[Asterisk-Users] CDR TDS

2005-08-08 Thread Gulzar Hussain 
Hi 

I am using asterisk and logging CDR to my SQL Server,
it works fine but if the Connection breaks between
Asterisk and SQL Server it doesnt reconnect itself

does somebody has any patch for doing it

Thanks





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[Asterisk-Users] Asterisk Windows Messenger

2004-11-22 Thread Gulzar Hussain 
Hi All

When I call from 1 Windows Messenger to  another
without Asterisk everything use to be fine but when
asterisk is there as a Registrar a problem use to
occur in many calls (With Canreinvite = Yes in
SIP.CONF), Caller can hear the voice of the receiving
side but the receiver cant be able hear the caller for
about 5 to 10 seconds, conversation will become two
way after 5 - 10 seconds but this problem is a big
hurdle in proper establishment of a call.

When i use sjphone instead of windows messenger i dont
get this problem but i have to use RTC Client APIs.

Thanks In Advance



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[Asterisk-Users] Problem In RTC Client With Asterisk

2004-11-04 Thread Gulzar Hussain 
Hi 
When I call from 1 RTC Client to another without
Asterisk everything use to be fine but when asterisk
is there as a Registrar a problem use to occur in many
calls, Caller can hear the voice of the receiving side
but the receiver doesnt get any voice for
about 5 to 10 seconds, conversation will become two
way after 5 - 10 seconds.

Everything use to be fine if Asterisk isnt there.
Everything use to be fine if I use SJPhone or XLite as
SIP UA.

Does anybody ever had this problem ?
Any suggestions will be higly apreciated
Thanx in Advance





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