[asterisk-users] Help : problem in SLA (Shared Line Apperence
On 8/7/07, raviprakash sunkara [EMAIL PROTECTED] wrote:
Hello Russell,
Nice To meet U and Good Morning. I got u r mail-Id from
http://www.asterisk.org/node/48325
Recently i started the SLA configuration. But i didn't understand the
Flow of its Functionality
One of the My Client Ask to have do deploySLA feature
He Using the Aastra 55i, when users is busy , Aastra 55i will blink lamps
in SLA.conf
slatest]
type=trunk
device=SIP/1001
autocontext=slatest
[slatest1]
type=trunk
device=SIP/1003
autocontext=slatest1
[slateststation]
type=station
device=SIP/1002
autocontext=slateststation
trunk=slatest
trunk=slatest1
sip.conf
[1001]
type=friend
username=1001
secret=1001
host=dynamic
;context=slatest
context=slatest
dtmfmode=rfc2833
Language=en
qualify=yes
[EMAIL PROTECTED]
disallow=all
allow=all
[1002]
type=friend
username=1002
secret=1002
host=dynamic
;context=default1
context=slateststation
dtmfmode=rfc2833
Language=en
qualify=yes
[EMAIL PROTECTED]
disallow=all
allow=all
[1003]
type=friend
username=1003
secret=1003
host=dynamic
;context=default1
context=slatest1
dtmfmode=rfc2833
Language=en
qualify=yes
[EMAIL PROTECTED]
disallow=all
allow=all
Dialplan
[testing]
exten = _100X,1,Dial(SIP/${EXTEN}/${EXTEN})
exten = 101,1,Goto(slateststation|102|1)
exten = 102,1,Goto(slatest|1|1)
exten = 103,1,Goto(slatest1|1|1)
exten = h,1,Hangup()
[slatest]
exten = 1,1,SLATrunk(slatest)
exten = _X.,1,Dial(SIP/${EXTEN}/${EXTEN})
[slatest1]
exten = 1,1,SLATrunk(slatest1)
exten = _X.,1,Dial(SIP/${EXTEN}/${EXTEN})
[slateststation]
exten = 102,1,SLAStation(slateststation)
Thanks Regards
Ravi Prakash Sunkara
India
--
Thanks Regards
Ravi Prakash Sunkara
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[asterisk-users] No Audio when integrating with openSER and Asterisk in the SAME LAN ,
Hello Users ,
I Posted to mailing list, No one is replying My issues,
My Issue is No Audio when Openser and Asterik integrated in Same LAN ,
When UAC are Behind the NAT, With out the Asterisk integration Behind the
NAT is working Fine.
SIP port and RTP ports are forwarded into router to OpenSER System only.
openser.cfg
listen=192.168.2.11
alias=sip.hyperion.com
# Invite Section
if ( method== invite )
{
# proxy authentication
if( uri==sip:[EMAIL PROTECTED] )
{ # rewriting to Asterisk server for Voicemail messages
rewritehostport(192.168.2.75:5060);
exit;
}
}
The Asterisk is executing , But No Audio to the UAC in Behind the NAT ..
In Asterisk sever is I'm setting this chan_Sip.so:[2344] retrans_pkt:
Maximum exceeded on transmission
--
Thanks and Regards
Ravi Prakash Sunkara
[EMAIL PROTECTED]
M:+91 9985077535
www.hyperion-tech.com
Client and Parent company :- www.august-networks.com
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[asterisk-users] No Audio when integrating openSER and Asterisk , in NAT
Hello Users openSER is sip proxy and registrar , Asterisk is as PBX, Conference and Voicemail servers, openSER and Asterisk are in the Same N/w Where As the UAC are in Behind the NAT, When Astetrisk is not integrated , UAC are in Behind the NAT is working, openSER is 192.168.2.5 Asterisk is 192.168.2.6 I'm just use rewritehost to asterisk server, UAC openSER - - - - Asterisk (voicemail sserver ). Please help me -- Thanks and Regards Ravi Prakash Sunkara [EMAIL PROTECTED] M:+91 9985077535 www.hyperion-tech.com Client and Parent company :- www.august-networks.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OutBound Proxy calls failing
Hello Users, Good AfterNoon to all I'm Mainly focused on OpenSER and Asterisk Integration. I didn't Find any solution of My Question ? Till now I'm doing only communicating OpenSER and Asterisk through SIP Channel only. User in Asterisk can Call to OpenSER and also vice-versa . But My Question ? I have one VoIP Service line from Voyage ( SIP change ), I want, if one of the User in Asterisk has to call Voyage service, and Call from Voyage line to has to Asterisk server, that play the IVR. -- Thanks and Regards Ravi Prakash Sunkara [EMAIL PROTECTED] M:+91 9985077535 www.hyperion-tech.com Client and Parent company :- www.august-networks.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Out Proxy Call
Hello Users I have one VoIP service from Packet8 ( SIP protocal ) Packet8- Astetisk server - My SIP agents My Sip Agents are in Asterisk Server , I configured.. If any one user in My Asterisk has to Call the Packet8 service providers , How can I configure it. Till now I'm Doing on OpenSER and ASterisk (Voicemail and Confereing ..) But My Asterisk has to connect the Packet8 service providers please Help me... -- Thanks and Regards Ravi Prakash Sunkara [EMAIL PROTECTED] M:+91 9985077535 www.hyperion-tech.com Client and Parent company :- www.august-networks.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Load Balancing
Hello Users, How can I perform the load Balancing in My SIP server of Both OpenSER and Asterisk , Currently I have One OpenSER server and Asterisk Server, For OpenSER is to need use these modules, and is any 1) LCR and Dispatcher modules, 2) OSP Modules ( also need ) Please can anyone help me .. -- Thanks and Regards Ravi Prakash Sunkara [EMAIL PROTECTED] M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 [EMAIL PROTECTED] www.hyperion-tech.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] integrating with Asterisk and OpenSER for Voicemail
Hi Users,
I'm Setting UP the Voicemails by integrating with Asterisk and OpenSER,
After 32 sec or 6 ring, it has to go the Voicemail server of Asterisk,
In openser.cfg ... is not hiiting the Asterisk server
. ... any one help me
modparam(tm,fr_timer,6)
modparam(tm,fr_inv_timer,24)
modparam(tm,wt_timer,1)
#mrodparam(tm, ruri_matching, 0)
#modparam(tm, via1_matching, 0)
modparam(avpops,avp_url,mysql://root:[EMAIL PROTECTED]/openser)
modparam(avpops, avp_table, usr_preferences)
modparam(avpops,avp_aliases,inv=i:15)
...
route
{
..
if (loose_route()) {
t_relay();
exit;
};
if(is_method(INVITE) !has_totag())
{
xdbg(user [$ruri] has voicemail redirection
enabled\n);
# backup R-URI
avp_write($ruri,$avp(inv));
setflag(2);
};
..
route(1);
}
route[1] {
if(isflagset(2))
{
t_on_failure(1);
};
}
failure_route[1] {
log(- \n);
if (t_was_cancelled()) {
xdbg(transaction was cancelled by UAC\n);
return;
}
# restore initial uri
avp_pushto($ruri, $avp(inv));
prefix(9);
# route to Asterisk Media Server
rewritehostport(192.168.2.75:5060);
append_branch();
t_relay(192.168.2.75:5060);
resetflag(2);
}
.
--
Thanks and Regards
Ravi Prakash Sunkara
[EMAIL PROTECTED]
M:+91 9985077535
O:+91 40 23114549
F:+91 40 40208727
[EMAIL PROTECTED]
www.hyperion-tech.com
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[asterisk-users] Happy X-mas
Hello * * * * * * * * * * * * * Happy X-mas and Adv Happy New Year ... ** * -- Thanks and Regards Ravi Prakash Sunkara [EMAIL PROTECTED] M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 [EMAIL PROTECTED] www.hyperion-tech.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to communicated Both SIP and IAX2 each other ?
Hello Users.. Is it possible to do. one UA is SIP and other UA is IAX2, UA(sip)---OpenSER-- Asterisk-- UA(IAX2) . UA(IAX2) --- Asterisk --- OpenSER -- UA (SIP ). other wise we can like that.. UA(SIP ) --- Asterisk-UA(IAX2) But SIP message and IAX messages are different , Then How can we communicate the both SIP and IAX2 -- Thanks and Regards Ravi Prakash Sunkara [EMAIL PROTECTED] M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 [EMAIL PROTECTED] www.hyperion-tech.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How can i processed with Call Snooping,
Hello Users, Nice to meet You, How can i Processed the call Snooping, it my fifth Requesting and posting to Users, Nobody replies it,,, in Call snooping , How can i record the Voip users, We can record to users by using Monitor Application , Can any give clear layout of the Snooping Feature... -- Thanks and Regards Ravi Prakash Sunkara [EMAIL PROTECTED] M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 [EMAIL PROTECTED] www.hyperion-tech.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to do Call barging with SIP channel
Hello Users I'm planning to do Call Barging and Call snooping , I saw this Feature in asterisk.org. This Barging and Snooping are test for is Agents are replying the Answer or not that I'm guessing Can anybody help me... this Feature .. How to do Call Barging and snooping in SIP Channels , I;m not using any Zaptel Card -- Thanks and Regards Ravi Prakash Sunkara [EMAIL PROTECTED] M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 [EMAIL PROTECTED] www.hyperion-tech.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] help in Call parking......
Hello Users I'm Doing working on Both OpenSER and Asterisk ... 9001 and 9003 are registered in OpenSER in extension.conf [from-sip] exten=115,1,Park() exten =115,2.Hungup() in Feature.conf ( default park no 701) in sip.conf [9001] ... .. [9002] [9003] When 9003 dial the 115 ( Parking itself) , Asterisk Server says U parked on 701 extension After When 9001 dial 701 . it Say 483 too many parameters ... in X-lite , Actual it has to ring 9003, -- Executing Park(SIP/9003-085d9e10, ) in new stack == Parked SIP/9003-085d9e10 on 701. Will timeout back to extension [from-sip] s, 1 in 45 seconds -- Added extension '701' priority 1 to parkedcalls -- Playing 'digits/7' (language 'en') -- Playing 'digits/0' (language 'en') -- Playing 'digits/1' (language 'en') -- Started music on hold, class 'default', on channel 'SIP/9003-085d9e10' == Spawn extension (from-sip, s, 1) exited KEEPALIVE on 'SIP/9003-085d9e10' Nov 22 18:18:45 NOTICE[3289]: rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 192.168.2.5 -- Stopped music on hold on SIP/9003-085d9e10 -- Registered extension context 'park-dial' -- Added extension 'SIP/9003' priority 1 to park-dial == Timeout for SIP/9003-085d9e10 parked on 701. Returning to park-dial,SIP/9003,1 -- Executing Dial(SIP/9003-085d9e10, SIP/9003||t) in new stack -- Called 9003 -- Got SIP response 482 Loop Detected back from 192.168.2.76 -- Now forwarding SIP/9003-085d9e10 to 'Local/[EMAIL PROTECTED]' (thanks to SIP/9003-085df878) Nov 22 18:19:20 NOTICE[3949]: chan_local.c:498 local_alloc: No such extension/context [EMAIL PROTECTED] creating local channel Nov 22 18:19:20 NOTICE[3949]: app_dial.c:474 wait_for_answer: Unable to create local channel for call forward to 'Local/[EMAIL PROTECTED]' (cause = 0) == Everyone is busy/congested at this time (1:0/0/1) Nov 22 18:19:30 WARNING[3949]: pbx.c:2415 __ast_pbx_run: Timeout, but no rule 't' in context 'park-dial' -- Thanks and Regards Ravi Prakash Sunkara [EMAIL PROTECTED] M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 [EMAIL PROTECTED] www.hyperion-tech.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] in Asterisk Manger its Unauthentication User and Host ..........
Hello Users. I'm Now doing on Asterisk Manager for My knowledge Growth, Can anybody explan me on Asterisk Manager settings... in manager.conf [general] enabled =yes port = 5038 bindaddr = 192.168.2.75 displayconnects = yes [hyperion] secret = hyperion permit=192.168.2.76/255.255.255.0 deny=0.0.0.0/0.0.0.0 read = system,call,log,verbose,command,agent,user write = system,call,log,verbose,command,agent,user Type #usr/sbin ./astman 192.168.2.75 its says following == Parsing '/etc/asterisk/manager.conf': Found Nov 23 10:04:39 NOTICE[24118]: manager.c:529 authenticate: 192.168.2.75failed to pass IP ACL as 'hyperion' == Connect attempt from '192.168.2.75' unable to authenticate == Parsing '/etc/asterisk/manager.conf': Found -- Thanks and Regards Ravi Prakash Sunkara [EMAIL PROTECTED] M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 [EMAIL PROTECTED] www.hyperion-tech.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to do the Call Snooping
Hello Users,I googled Call Snooping, its shows only the it means, But i didn't find How to dialplan the Call Snooping,I seen that What is Trixbox in Asterisk I Use only some Feature in Asterisk (20), Is it need Asterisk to install the TrixBox in that same System where i installed the Asterisk ServerHelp me please :P-- Thanks and RegardsRavi Prakash Sunkara [EMAIL PROTECTED] M:+91 9985077535O:+91 40 23114549F:+91 40 40208727 [EMAIL PROTECTED] www.hyperion-tech.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help me on Call parking
Hello Users...I'm Strucked in Call parking...I'm Using the Asterisk-1.1.11 version in My FC5 box,In That there is feature.confI'm Using SIP channel By using Asterisk + OpenSER [general]parkext = 9006 ; What extension to dial to parkparkpos = 9007-9009 ; What extensions to park calls on. These needs to be ; numeric, as Asterisk starts from the start position ; and increments with one for the next parked call.context = parkedcalls ; Which context parked calls are inparkingtime = 45 ; Number of seconds a call can be parked for ; (default is 45 seconds)IIn Extension.conf .. I'm confused to give the Dial planning..Can Help -- Thanks and RegardsRavi Prakash Sunkara [EMAIL PROTECTED] M:+91 9985077535O:+91 40 23114549F:+91 40 40208727 [EMAIL PROTECTED] www.hyperion-tech.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Some Warning in Asterisk for Voicemail intgreting,
Hello Users,I doing on Voicemail in Asterisk For my RealTime, By using the ODBC connectivity For Voicemessages.in Made the Change in res_odbc.conf,odbc.ini, odbcinst.ini and voicemail.conf When I start My Asterisk server it give me Some Warning,When I googled , a proper Docummentation is not found, it found in some there languages,the First Warning is. Warning [30188] res_odbc.c 565 odbc_obj_connect: res_object:Error SqlConnect =-1 Error=0 [UnixODBC][Driver Manager]Data Source Name Not Found and No default Driver Specified And Second one is Warning [30202] app_application.c 2107 MessageCount : Failed to Obtained for ' Asterisk' ! help me this.. -- Thanks and RegardsRavi Prakash Sunkara [EMAIL PROTECTED] M:+91 9985077535O:+91 40 23114549F:+91 40 40208727 [EMAIL PROTECTED]www.hyperion-tech.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem in Voice Message Storing...............
Good Morning, I need a small help from U on Regarding the Asterisk , Currently I'm Doing Voice Mail in Asterisk which is forwarded By OpenSER. I can Leave the Voice message to the Caller , But Stores in this Directory /var/spool/asterisk/voicemail/ context. But For My Real Time and User interface developing , I want to Store in Database, As per My Knowledge and Googled , I found the ODBC with MySQl is there. When I configured it The Record are not store in Voicemessages Table ... Help Me -- Forwarded message --From: raviprakash sunkara [EMAIL PROTECTED] Date: Oct 14, 2006 10:57 AMSubject: Problem in Voice Message Storing...To: ram [EMAIL PROTECTED]Hello RamGood Morning, I need a small help from U on Regarding the Asterisk ,Currently I'm Doing Voice Mail in Asterisk which is forwarded By OpenSER.I can Leave the Voice message to the Caller , But Stores in this Directory /var/spool/asterisk/voicemail/ context. But For My Real Time and User interface developing , I want to Store in Database,As per My Knowledge and Googled , I found the ODBC with MySQl is there. When I configured it The Record are not store in Voicemessages Table ...Help MeAnd, If U don't Mind can U give Ur Contact Number... Please. -- Thanks and RegardsRavi Prakash Sunkara [EMAIL PROTECTED] M:+91 9985077535O:+91 40 23114549F:+91 40 40208727 [EMAIL PROTECTED] www.hyperion-tech.com -- Thanks and RegardsRavi Prakash Sunkara [EMAIL PROTECTED] M:+91 9985077535O:+91 40 23114549F:+91 40 40208727 [EMAIL PROTECTED]www.hyperion-tech.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Some file aren't loaded its No file in that Directory.
Hello Users,I Installed the Asterisk-1.2.11, For My Real time Use I'm Use MySql For Asterisk Database, By Using the Asterisk-addons -1.2.4 in My Linux.For My Voice messages Storage , I want To Use the MySql. In Googled it shows me the ODBC integration..Is it need for that ODBC integration with MySql for my Voice Message storing in MySql.When I'm trying to integrate with ODBC + MySql. and Reinstall the Asterisk .. As per the Below Url..http://www.voip-info.org/wiki/view/Realtime+Integration+Of+Asterisk+With+OpenSERWhen i followed the Step by Step. And While reinstall the Asterisk Server ...it Shows me errors...sql.h and sqltest.h is not found in /usr/src/asterisk-1.2.11/includes/asterisk/ Please Help me in this Issue or..Help in How to Store the Voice Messages without integrating the ODBCStorage. -- Thanks and RegardsRavi Prakash Sunkara [EMAIL PROTECTED] M:+91 9985077535O:+91 40 23114549F:+91 40 40208727 [EMAIL PROTECTED]www.hyperion-tech.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] help this....
Hello UsersHelp me ... the below error [codec_lpc10.so] = (LPC10 2.4kbps (signed linear) Voice Coder) == Parsing '/etc/asterisk/codecs.conf': Found -- Message count requested for mailbox [EMAIL PROTECTED] but voicemail not loaded. -- codec_lpc10: using generic PLC == Registered translator 'lpc10tolin' from format lpc10 to slin, cost 41 == Registered translator 'lintolpc10' from format slin to lpc10, cost 2 Bolded one... is occuring in difference configuring files...Help what this means... :P-- Thanks and RegardsRavi Prakash Sunkara [EMAIL PROTECTED] M:+91 9985077535O:+91 40 23114549F:+91 40 40208727 [EMAIL PROTECTED] www.hyperion-tech.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP vz IAX...
Hello Users.I'm in Dilemma with the performance on SIP and IAXCan any one help ... 1) Difference between the SIP and IAX... which one is Best... in VOIP service I'm using only SIP protocol for my VOIP in OpenSER...And Also I using Asterisk in SIPwe can Communicate the SIP and IAX by below scenarioSIP (UA) OPENSER - ASTERISK IAX (UA)... this I can do... IAX --- OPENSER - ASTERISK - SIP/IAX.But main problem is ...SupposeIAX -- ASTERISK--- openSER SIP / IAX ... How ? Help me this forgive me in English :P-- Thanks and RegardsRavi Prakash Sunkara [EMAIL PROTECTED] M:+91 9985077535O:+91 40 23114549F:+91 40 40208727 [EMAIL PROTECTED]www.hyperion-tech.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Message count requested for mailbox [EMAIL PROTECTED] but voicemail not loaded.
Hello Can any help this messages .. What it meanMessage count requested for mailbox [EMAIL PROTECTED] but voicemail not loaded. -- Thanks and RegardsRavi Prakash Sunkara [EMAIL PROTECTED] M:+91 9985077535 O:+91 40 23114549F:+91 40 40208727 [EMAIL PROTECTED]www.hyperion-tech.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help in MySQL + Asterisk.
Hello Users...Can any one help on Asterisk with MySqLI don't want to use ODBC+MySqL. for RealTime...Just need the MySql and Asterisk integration..On That i need extension.conf ,sip.conf ,and voicemail.conf,meetme.conf,musiconhold.conf are in MySql Databases accesingIn Flaf files its working fine... with OpenSERHelp me..-- Thanks and RegardsRavi Prakash Sunkara [EMAIL PROTECTED] M:+91 9985077535O:+91 40 23114549F:+91 40 40208727 [EMAIL PROTECTED] www.hyperion-tech.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How can the User Know he has voicemail in the Databases.
Hi Users, I'm developing the Voicemail, By flat files I made it, But now I need to do in MySql Databases,In res_mysql.conf and cdr_mysql.conf I given the Database entitesWhile I'm reloading the asterisk server I have arrrived below one message,Can any one tell what this messages means,[cdr_addon_mysql.so] = (MySQL CDR Backend) == Parsing '/etc/asterisk/cdr_mysql.conf': Found -- Message count requested for mailbox [EMAIL PROTECTED] but voicemail not loaded. How to count the mailbox in the CDR or Voicemail _user.Help me And How to know users has a voicemail box in Database..,,-- Thanks and Regards Ravi Prakash Sunkara [EMAIL PROTECTED] M:+91 9985077535O:+91 40 23114549F:+91 40 40208727 [EMAIL PROTECTED]www.hyperion-tech.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help in Reloading of Asterisk...
Hi Users,I need help or clues from U, please help me...I'm new Asterisk, I want to do the Asterisk in RealTime ConfiguringMy problem is below one . After every change to the database, the asterisk will need to be reloaded.How to Reload the Asterisk server.Now its simple , stupid doubt to put in mail-list...I don't know it. -- Thanks and RegardsRavi Prakash Sunkara [EMAIL PROTECTED] M:+91 9985077535O:+91 40 23114549F:+91 40 40208727 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Integrating the Openser for VoiceMail and PBX with Asterisk, For Account
Hi Users,I'm new to Asterisk programming , I'm in working the Voip Technologies by using the OpenSER for my call routing process and Radius For AAA.But in Asterisk i need it for only PBX and VoiceMail,For Account I'm using the Openser + Radius . Main My doubt is that, For Call Routing my using the OpenSER. every thing is fine and Good .. But i need the Voicemail , it forword to the Asterisk Server, that . Is the Openser takes the Accounting part or Asterisk it Take place Who did the Database know the User had a voicemail in his voice mail Box... That Databases i need How ?Please Help me And Mainly Excuse me in English -- Thanks and Regards Ravi Prakash Sunkara M:+91 9985077535 O:+91 40 23114549 F:+91 40 40208727 [EMAIL PROTECTED] www.hyperion-tech.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help in dailplan in asterisk
Hi Users,I'm new to Asterisk, and I'm working with openSER , For Call Routing I'm using the OpenSER and for PBX, Voicemail and Conferencing Im using the Asterisk.Now i'm planing for Voicemail, openser listen on 192.168.2.75:5060Asterisk listen on 192.168.2.76:5060 .in Extension.conf[from-sip[ exten = 9001,1,Ringingexten = 9001,2,Voicemail(9001) Just I wnat to leave the voicemail in voice mail box,Can anybody help me. below message What it is ?Executing Ringing(SIP/9001-0984c1e8, ) in new stack -- Executing VoiceMail(SIP/9001-0984c1e8, 9001) in new stackAug 31 19:31:10 ERROR[7869]: res_config_mysql.c:651 mysql_reconnect: MySQL RealTime: Failed to connect database server asterisk on (err 2002). Check debug for more info. Aug 31 19:31:10 WARNING[7869]: app_voicemail.c:2412 leave_voicemail: No entry in voicemail config file for '9001'Aug 31 19:31:10 WARNING[7869]: pbx.c:1700 pbx_extension_helper: No application 'Hungup' for extension (from-sip, 9001, 3) == Spawn extension (from-sip, 9001, 3) exited non-zero on 'SIP/9001-0984c1e8' -- Incoming call: Got SIP response 479 Regretfully, we were not able to process the URI (479/SL) back from 192.168.2.75-- Thanks and Regards with cheers Sunkara Ravi Prakash (Voip Developer)Hyperion TechnologyKondapur, Hi-tech city,Hyderabad.www.hyperion-tech.com+91-9985077535 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] New to Asterisk...
Hi UsersI'm new to Asterisk PBX.Mainly i'm using the openser for call routing and Asterisk as PBX and Voicemail generating.let see my secnario ---UAC -- ser Asterisk(for voice mail only and extension and PBX Purposes SER system ip is 192.168.2.75:5060Asterisk is in 192.168.2.76:5060 When i start the asterisk server by typing asterisk -c [chan_sip.so] = (Session Initiation Protocol (SIP)) == Parsing '/etc/asterisk/sip.conf': FoundAug 30 19:41:05 WARNING[6694]: acl.c:244 ast_get_ip_or_srv: Unable to lookup ' 192.168.2.75:5060'Aug 30 19:41:11 WARNING[6694]: acl.c:244 ast_get_ip_or_srv: Unable to lookup ' 192.168.2.75:5060' == SIP Listening on 0.0.0.0:5060 == Using TOS bits 0please help in this. -- Thanks and Regards with cheersSunkara Ravi Prakash (Voip Developer) Hyperion TechnologyKondapur, Hi-tech city,Hyderabad.www.hyperion-tech.com+91-9985077535 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
