Re: [asterisk-users] Getting 603 Declined after AGI execution

2011-05-21 Thread virendra bhati
Hi Alejandro,

As per my a2billing knowledge it's termination issue. Might be your call
balance is finished please check and then test again.

On Thu, May 19, 2011 at 8:40 PM, Alejandro Mejia Evertsz ame...@gua.netwrote:

 Hello everyone.

 I'm using Asterisk 1.4.31 and A2Billing 1.7.0 to manage a small wholesale
 operation, so I configured A2Billing for not to answer the call nor play any
 greetings or balance notifications to the caller.
 I'm authenticating each customer by it's IP address, and each customer has
 it's own context, in which I set the following:

 ;=in extensions.conf==
 [from-customerX]
 exten = _X.,1,Set(CDR(accountcode)=)  ;Here I change the accountcode
 depending on each customer
 exten = _X.,2,Set(CALLERID(dnid)=${EXTEN})
 exten = _X.,3,Goto(a2billing|${EXTEN}|1)

 [from-customerY]
 exten = _X.,1,Set(CDR(accountcode)=)
 exten = _X.,2,Set(CALLERID(dnid)=${EXTEN})
 exten = _X.,3,Goto(a2billing|${EXTEN}|1)

 [a2billing]
 exten = _X.,1,DeadAGI(a2billing.php|1)
 exten = _X.,2,Hangup(34)
 ;=

 A2Billing authenticates and routes the call properly, but when the
 termination gateway for the destination dialed by the customer rejects the
 call, my Asterisk box sends 603 Declined to the customer.
 It also happens when A2Billing doesn't find any route for that destination,
 in which it should return 404 Not Found, but returns 603 Declined
 instead.
 I tried to force every rejected attempt with 503 Service Unavailable
 putting the Hangup(34) you see on my config, but it never seems to get
 there.
 The last thing I see on CLI running in verbose is: -- AGI Script
 a2billing.php completed, returning 0

 Is there anything I could do to return a different cause than 603
 Declined?
 I posted the same question on A2Billing's forum, but had no luck.

 Thanks in advance,

 Alejandro Mejia

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Asterisk Engineer
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[asterisk-users] difference between SIP peer and SIP user ?

2011-05-21 Thread virendra bhati
Hi list,

I am confuse about these CLI commands
*sip show users
sip show peers*

Can someone clear my doubt . what are the difference between them?

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 Virendra Bhati
+91-9172341457
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[asterisk-users] how to user SIP realtime option

2011-05-21 Thread virendra bhati
Hi List,

After read the link
http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip. I changes
all information in below conf files

*res_mysql.conf*

[mpathsala]
dbhost = localhost
dbname = mpathsala
dbuser = root
dbpass =
dbport = 3306

*extconfig.conf*

sipusers = mysql,mpathsala,sip_buddies
sippeers = mysql,mpathsala,sip_buddies

*mysql*

make same table sip_buddies in mpathsala database.
insert 1 row into database

[image: Full 
Texts]http://192.168.193.69/phpmyadmin/sql.php?db=mpathsalatable=sip_buddiessql_query=SELECT+%2A+FROM+%60sip_buddies%60goto=sql.php%3Fdb%3Dmpathsala%26amp%3Btable%3Dsip_buddies%26amp%3Btoken%3D0ba5913eba31b6657e85f0a1ffec66de%26amp%3Bsql_query%3DSELECT%2B%252A%2BFROM%2B%2560sip_buddies%2560dontlimitchars=1token=0ba5913eba31b6657e85f0a1ffec66de
id
name host nat type accountcode amaflags call-limit callgroup callerid
cancallforward
canreinvite context defaultip dtmfmode fromuser fromdomain insecure language
mailbox md5secret deny permit mask musiconhold pickupgroup qualify
regexten restrictcid
rtptimeout rtpholdtimeout secret setvar disallow allow fullcontact ipaddr port
regserver regseconds lastms username defaultuser subscribecontext
useragent [image:
Edit]http://192.168.193.69/phpmyadmin/tbl_change.php?db=mpathsalatable=sip_buddiestoken=0ba5913eba31b6657e85f0a1ffec66deprimary_key=+%60sip_buddies%60.%60id%60+%3D+1sql_query=SELECT+%2A+FROM+%60sip_buddies%60goto=sql.php
 [image:
Delete]http://192.168.193.69/phpmyadmin/sql.php?db=mpathsalatable=sip_buddiestoken=0ba5913eba31b6657e85f0a1ffec66desql_query=DELETE+FROM+%60sip_buddies%60+WHERE+%60sip_buddies%60.%60id%60+%3D+1+LIMIT+1zero_rows=The+row+has+been+deletedgoto=sql.php%3Fdb%3Dmpathsala%26table%3Dsip_buddies%26token%3D0ba5913eba31b6657e85f0a1ffec66de%26sql_query%3DSELECT%2B%252A%2BFROM%2B%2560sip_buddies%2560%26zero_rows%3DThe%2Brow%2Bhas%2Bbeen%2Bdeleted%26goto%3Dsql.php%3Fdb%3Dmpathsala%26amp%3Btable%3Dsip_buddies%26amp%3Btoken%3D0ba5913eba31b6657e85f0a1ffec66de%26amp%3Bsql_query%3DSELECT%2B%252A%2BFROM%2B%2560sip_buddies%2560
1 300 dynamic no friend *NULL* *NULL* *NULL* *NULL* 100 yes yes *NULL* *NULL
* *NULL* *NULL* *NULL* *NULL* *NULL* *NULL* *NULL* *NULL* *NULL* *NULL* *
NULL* *NULL* *NULL* *NULL* *NULL* *NULL* *NULL* *NULL* *NULL* all
g729;ilbc;gsm;ulaw;alaw 0 *NULL* 0 0 *NULL* *NULL*

*sip.conf *

[general]

rtcachefriends=yes
rtsavesysname=yes
rtupdate=yes
rtautoclear=yes

After all when I check on CLI then I will get

*cent70*CLI sip show peers*
Name/username  HostDyn Nat ACL Port Status
Realtime
0 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 0
offline]

*cent70*CLI sip show users*
Username   Secret   Accountcode
Def.Context  ACL  NAT

Why SIP/300 is not display here ??

Please help me I want to learn asterisk real-time concept to make my server
real-time.
-
Thanks and regards

 Virendra Bhati
+91-9172341457
Asterisk Engineer
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Re: [asterisk-users] how to know how many calls are on hold

2011-05-21 Thread virendra bhati
Hi
Yes it show none until we accept the call but after receive the call and
then we check the status is too complected.
1) before answer the call

**CLI core show channels concise*
SIP/200-0007!office!200!1!Ringing!AppDial!(Outgoing
Line)!200!!3!11!(None)!1305982734.7
SIP/300-0006!office!200!2!Up!Dial!SIP/200,45,tTrkK!300!!3!12!(None)!1305982734.6
SIP/200-0005!office!!1!Up!AppDial!(Outgoing
Line)!200!!3!88!SIP/100-0004!1305982658.5
SIP/100-0004!office!200!2!Up!Dial!SIP/200,45,tTrkK!100!!3!88!SIP/200-0005!1305982658.4

2) After answer the call (info is different due to different log info)

SIP/200-0003!office!!1!Up!
AppDial!(Outgoing Line)!200!!3!35!SIP/300-0002!1305981842.3
SIP/300-0002!office!200!2!Up!Dial!SIP/200,45,tTrkK!300!!3!36!SIP/200-0003!1305981841.2
SIP/200-0001!office!!1!Up!AppDial!(Outgoing
Line)!200!!3!77!SIP/100-!1305981800.1
SIP/100-!office!200!2!Up!Dial!SIP/200,45,tTrkK!100!!3!77!SIP/200-0001!1305981800.0

Please help me on that.



On Wed, May 18, 2011 at 1:47 PM, Tiago Geada tiago.ge...@gmail.com wrote:

 core show channels concise

 Those with '(None)' haven't been briged yet.

 On 17 May 2011 15:16, virendra bhati virbh...@gmail.com wrote:

 hi list,

 please help me how to know how many calls are on hold.

 --



 -
 Thanks and regards

  Virendra Bhati
 +91-9172341457
 Asterisk Engineer


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+91-9172341457
Asterisk Engineer
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Re: [asterisk-users] how to user SIP realtime option

2011-05-21 Thread Rafael dos Santos Saraiva
Hi
Trying exclude rtcachefriends from your sip.conf and include the field
rtchachefriends in table sip_buddies. And exclude the field qualify from
sip_buddies. Set YES in field rtcachefriends.

Att,
Rafael Saraiva

2011/5/21 virendra bhati virbh...@gmail.com

 Hi List,

 After read the link
 http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip . I changes all
 information in below conf files

 *res_mysql.conf*

 [mpathsala]
 dbhost = localhost
 dbname = mpathsala
 dbuser = root
 dbpass =
 dbport = 3306

 *extconfig.conf*

 sipusers = mysql,mpathsala,sip_buddies
 sippeers = mysql,mpathsala,sip_buddies

 *mysql*

 make same table sip_buddies in mpathsala database.
 insert 1 row into database

 [image: Full 
 Texts]http://192.168.193.69/phpmyadmin/sql.php?db=mpathsalatable=sip_buddiessql_query=SELECT+%2A+FROM+%60sip_buddies%60goto=sql.php%3Fdb%3Dmpathsala%26amp%3Btable%3Dsip_buddies%26amp%3Btoken%3D0ba5913eba31b6657e85f0a1ffec66de%26amp%3Bsql_query%3DSELECT%2B%252A%2BFROM%2B%2560sip_buddies%2560dontlimitchars=1token=0ba5913eba31b6657e85f0a1ffec66de
  id
 name host nat type accountcode amaflags call-limit callgroup callerid 
 cancallforward
 canreinvite context defaultip dtmfmode fromuser fromdomain insecure language
 mailbox md5secret deny permit mask musiconhold pickupgroup qualify regexten
 restrictcid rtptimeout rtpholdtimeout secret setvar disallow allow fullcontact
 ipaddr port regserver regseconds lastms username defaultuser subscribecontext
 useragent [image: 
 Edit]http://192.168.193.69/phpmyadmin/tbl_change.php?db=mpathsalatable=sip_buddiestoken=0ba5913eba31b6657e85f0a1ffec66deprimary_key=+%60sip_buddies%60.%60id%60+%3D+1sql_query=SELECT+%2A+FROM+%60sip_buddies%60goto=sql.php
   [image:
 Delete]http://192.168.193.69/phpmyadmin/sql.php?db=mpathsalatable=sip_buddiestoken=0ba5913eba31b6657e85f0a1ffec66desql_query=DELETE+FROM+%60sip_buddies%60+WHERE+%60sip_buddies%60.%60id%60+%3D+1+LIMIT+1zero_rows=The+row+has+been+deletedgoto=sql.php%3Fdb%3Dmpathsala%26table%3Dsip_buddies%26token%3D0ba5913eba31b6657e85f0a1ffec66de%26sql_query%3DSELECT%2B%252A%2BFROM%2B%2560sip_buddies%2560%26zero_rows%3DThe%2Brow%2Bhas%2Bbeen%2Bdeleted%26goto%3Dsql.php%3Fdb%3Dmpathsala%26amp%3Btable%3Dsip_buddies%26amp%3Btoken%3D0ba5913eba31b6657e85f0a1ffec66de%26amp%3Bsql_query%3DSELECT%2B%252A%2BFROM%2B%2560sip_buddies%2560
 1 300 dynamic no friend *NULL* *NULL* *NULL* *NULL* 100 yes yes *NULL* *
 NULL* *NULL* *NULL* *NULL* *NULL* *NULL* *NULL* *NULL* *NULL* *NULL* *NULL
 * *NULL* *NULL* *NULL* *NULL* *NULL* *NULL* *NULL* *NULL* *NULL* all
 g729;ilbc;gsm;ulaw;alaw 0 *NULL* 0 0 *NULL* *NULL*

 *sip.conf *

 [general]

 rtcachefriends=yes
 rtsavesysname=yes
 rtupdate=yes
 rtautoclear=yes

 After all when I check on CLI then I will get

 *cent70*CLI sip show peers*
 Name/username  HostDyn Nat ACL Port Status
 Realtime
 0 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 0
 offline]

 *cent70*CLI sip show users*
 Username   Secret   Accountcode
 Def.Context  ACL  NAT

 Why SIP/300 is not display here ??

 Please help me I want to learn asterisk real-time concept to make my server
 real-time.
 -
 Thanks and regards

  Virendra Bhati
 +91-9172341457
 Asterisk Engineer


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Att,
Rafael Saraiva
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Re: [asterisk-users] how to know how many calls are on hold

2011-05-21 Thread Ryan Wagoner
On Tue, May 17, 2011 at 10:16 AM, virendra bhati virbh...@gmail.com wrote:
 hi list,

 please help me how to know how many calls are on hold.


If they are SIP channels you can use: sip show inuse The last column
are calls on hold.

Ryan

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Re: [asterisk-users] Polycom IP335 3.3.1 Call Waiting

2011-05-21 Thread Ryan Wagoner
On Thu, May 19, 2011 at 1:24 PM, Ryan Wagoner rswago...@gmail.com wrote:
 I updated my phones to the UCS 3.3.1 firmware a few months back. The
 scenario is I place a call and receive an incoming call. With 3.3.1
 the screen will show call 1/2 and I have to press the down arrow to
 see the caller name / number. Has anybody else noticed this with
 3.3.1? I had thought with 3.2.4 it would automatically show call
 waiting name and number without pressing any keys. It could be
 possible I missed a setting, but I didn't see anything in the admin
 guide.

 Ryan

For those wondering it appears to be a bug in 3.2.5 and later
versions. I downgraded to 3.2.4 and the caller id for the incoming
call waiting call will show for 10 seconds as described in the Polycom
user guide. This only effects those with IP33x model phones.

Ryan

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