[asterisk-users] asterisk realtime SIP configuration
Hi All,
I am trying to configure asterisk realtime. But i am unable to get the
extensions listed successfully when i type sip show peers in the asterisk
CLI . i am unable to see any failure logs when i do a reload
i can able to connect to the data source through odbc show in the
CLI, Any hep in this regard is highly appreciated. Following is the
configuration and specification.
*Server Specification:*
1) asterisk-1.6.2.6
2) CentOS- 5.2 (64-bit)
3) Postgresql- 8.1
*Configuration:*
* odbc.ini*
[PostgreSQL]
Description = Test to Postgres
Driver = PostgreSQL
Trace = Yes
TraceFile = /tmp/sql.log
Database= bedrock
Servername = localhost
UserName=
Password=
Port= 5432
Protocol= 6.4
ReadOnly= No
RowVersioning = No
ShowSystemTables= No
ShowOidColumn = No
FakeOidIndex= No
ConnSettings=
*odbcinst.ini*
[PostgreSQL]
Description = ODBC for PostgreSQL
Driver = /usr/lib64/libodbcpsql.so
Setup = /usr/lib64/libodbcpsqlS.so
FileUsage = 1
* res_odbc.conf*
[postgres]
enabled = yes
dsn = PostgreSQL
username =postgres
password =postgres
pre-connect = yes
*Database table in postgres sip :*
Column | Type |Modifiers
++--
id | integer| not null default
nextval('sip_id_seq'::regclass)
name | character varying(80) | not null
accountcode| character varying(20) |
amaflags | character varying(7) |
callgroup | character varying(10) |
callerid | character varying(80) |
directmedia| character varying(3) | default 'yes'::character varying
context| character varying(80) | default 'default'::character
varying
defaultip | character varying(15) |
dtmfmode | character varying(7) |
fromuser | character varying(80) |
fromdomain | character varying(80) |
host | character varying(31) | not null default
'dynamic'::character varying
insecure | character varying(4) |
language | character varying(2) |
mailbox| character varying(50) |
md5secret | character varying(80) |
nat| character varying(5) | not null default 'no'::character
varying
permit | character varying(95) |
deny | character varying(95) |
mask | character varying(95) |
pickupgroup| character varying(10) |
port | character varying(5) |
qualify| character varying(3) |
restrictcid| character varying(1) |
rtptimeout | character varying(3) |
rtpholdtimeout | character varying(3) |
secret | character varying(80) |
type | character varying | not null default
'friend'::character varying
username | character varying(80) |
disallow | character varying(100) | default 'all'::character varying
allow | character varying(100) | default 'alaw,ulaw'::character
varying
musiconhold| character varying(100) |
regseconds | integer| not null default 0
ipaddr | character varying(15) |
regexten | character varying(80) |
cancallforward | character varying(3) | default 'yes'::character varying
lastms | character varying(80) |
useragent | character varying(100) |
defaultuser| character varying(100) |
fullcontact| character varying(100) |
regserver | character varying(100) |
Indexes:
sip_conf_pkey PRIMARY KEY, btree (id)
name UNIQUE, btree (name)
*extconfig.conf*
sipusers = odbc,postgres,sip
sippeers = odbc,postgres,sip
Thanks Regards
Murali Vasu
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[asterisk-users] call features affected by native bridging between sip phones
Hi Geeks, I am a beginner in asterisk, I read about native bridging option in asterisk which allows the RTP streaming through the SIP media terminals after initiating the call . I identified the following features are getting affected by this feature in my testing. 1) Call transfer. 2) Music On Hold 3) Conferencing with meetme. I wonder if there are any other features will get affected due to native bridging. Thanks in advance. Regards Murali Vasu -- Smile is the only priceless gift you can give without a price. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk PBX causes mysql to take more CPU time
Hi All, I am using asterisk-1.4.22 with mysql as my storage medium for voice messages.Right now i am running 700+ extensions with this setup . *System Configuration:* asterisk-1.4.22.1 (using odbc storage for voice messages) unixODBC-2.2.11-7.1 mysql-connector-odbc-3.51.12-2.2 mysql-server-5.0.45-7.el5 CentOS-5.2 (64 bit) *The following are the issues i am facing:* 1) CPU time always shows 30%-90% for mysql when asterisk is running . The CPU time goes normal for mysql when asterisk is stopped. 2) calls are getting dropped suddenly. 3) Some answered calls does not hear anything at the receiver/initiater end. 4) asterisk itself gets crashed at times 5) Core dumps are getting created. Is there any optimizations or configurations needs to be done on my server in order to drill down this issue? Please help me out in troubleshooting this issue, This problems are getting worsen day by day. Thanks in advance. -- always withsmile vimurli ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Core dump gets created while accessing voicemail
Hi ALL,
When i was accessing the voice message it suddenly goes dead and after that
i couldn't able to retrieve the voicemessage again from my mailbox . This
happens once in a while for any configured mailboxes
I am using the following system configuration.
asterisk 1.4.22.1
odbc storage of voicemail messages
centos 5.2 64bit
unixODBC-2.2.11-7.1
mysql-connector-odbc-3.51.12-2.2
mysql-server-5.0.45-7.el5
Following are the traces i found while troubleshooting the issue.
1) I found a .lock file created in the INBOX folder of my mailbox
2) The following core dump gets created during the first time the
voicemessage access got failed.
* (gdb) bt
*#0 0x00322b417649 in SQLFreeEnv () from /usr/lib64/libodbc.so.1
#1 0x00322b417b5c in SQLFreeHandle () from /usr/lib64/libodbc.so.1
#2 0x2aaac132ccdd in message_exists (dir=value optimized out,
msgnum=value optimized out)
from /usr/lib/asterisk/modules/app_voicemail.so
#3 0x2aaac132dab2 in save_to_folder (vmu=0x41751e80, vms=value
optimized out, msg=0, box=value optimized out)
from /usr/lib/asterisk/modules/app_voicemail.so
#4 0x2aaac132dc77 in close_mailbox (vms=0x4174bd60, vmu=0x41751e80)
from /usr/lib/asterisk/modules/app_voicemail.so
#5 0x2aaac1341b43 in vm_execmain (chan=0x2aaab42abe30, data=value
optimized out)
from /usr/lib/asterisk/modules/app_voicemail.so
#6 0x00481e2d in pbx_extension_helper (c=0x2aaab42abe30, con=value
optimized out,
context=0x2aaab42ac080 staff-international, exten=0x2aaab42ac0d0
*97, priority=106, label=value optimized out,
callerid=0xbff51d0 1369, action=E_SPAWN) at pbx.c:537
#7 0x00483b66 in __ast_pbx_run (c=0x2aaab42abe30) at pbx.c:2317
#8 0x00484849 in pbx_thread (data=0x322b663b40) at pbx.c:2621
#9 0x004aef5c in dummy_start (data=value optimized out) at
utils.c:912
#10 0x00322b006307 in start_thread () from /lib64/libpthread.so.0
#11 0x00322a4d1ded in clone () from /lib64/libc.so.6
---
*(gdb)bt full
*
#0 0x00322b417649 in SQLFreeEnv () from /usr/lib64/libodbc.so.1
#1 0x00322b417b5c in SQLFreeHandle () from /usr/lib64/libodbc.so.1
#2 0x2aaac132ccdd in message_exists (dir=value optimized out,
msgnum=value optimized out)
from /usr/lib/asterisk/modules/app_voicemail.so
#3 0x2aaac132dab2 in save_to_folder (vmu=0x41751e80, vms=value
optimized out, msg=0, box=value optimized out)
from /usr/lib/asterisk/modules/app_voicemail.so
#4 0x2aaac132dc77 in close_mailbox (vms=0x4174bd60, vmu=0x41751e80)
from /usr/lib/asterisk/modules/app_voicemail.so
#5 0x2aaac1341b43 in vm_execmain (chan=0x2aaab42abe30, data=value
optimized out)
from /usr/lib/asterisk/modules/app_voicemail.so
#6 0x00481e2d in pbx_extension_helper (c=0x2aaab42abe30, con=value
optimized out,
context=0x2aaab42ac080 staff-international, exten=0x2aaab42ac0d0
*97, priority=106, label=value optimized out,
callerid=0xbff51d0 1369, action=E_SPAWN) at pbx.c:537
#7 0x00483b66 in __ast_pbx_run (c=0x2aaab42abe30) at pbx.c:2317
#8 0x00484849 in pbx_thread (data=0x322b663b40) at pbx.c:2621
#9 0x004aef5c in dummy_start (data=value optimized out) at
utils.c:912
#10 0x00322b006307 in start_thread () from /lib64/libpthread.so.0
#11 0x00322a4d1ded in clone () from /lib64/libc.so.6
(gdb) bt full
#0 0x00322b417649 in SQLFreeEnv () from /usr/lib64/libodbc.so.1
No symbol table info available.
#1 0x00322b417b5c in SQLFreeHandle () from /usr/lib64/libodbc.so.1
No symbol table info available.
#2 0x2aaac132ccdd in message_exists (dir=value optimized out,
msgnum=value optimized out)
from /usr/lib/asterisk/modules/app_voicemail.so
x = 1
stmt = (SQLHSTMT) 0xbf45620
sql = SELECT COUNT(*) FROM voicemessages WHERE dir=? AND
msgnum=?\000\000\000\000\000`}tA\000\000\000\000\225-N\000\000\000\000\000(\000\000\\000\000\000ТtA\000\000\000\000à¡tA\000\000\000\000\032\224F*2\000\000\000\001\200û\000\000\000\000Ø\226#\f\000\000\000\000Ø\226#\f\000\000\000\000Ø\226#\f\000\000\000\000Ø\226#\f\000\000\000\000\003\227#\f\000\000\000\000\212\227#\f\000\000\000\000Ø\226#\f\000\000\000\000\212\227#\f,
'\0' repeats 52 times, ¨þh*2\000\000\...@Þ...
rowdata = 1\000k/voicemail/admin/
msgnums = 31\000\000\000\000\000\000¦-N\000\000\000\000\000/var
argv = {0x41748c10
/var/spool/asterisk/voicemail/admin/1369/Delete, 0x41748b70 31}
gps = {sql = 0x41747b50 SELECT COUNT(*) FROM voicemessages WHERE
dir=? AND msgnum=?, argc = 2, argv = 0x41748bb0}
obj = (struct odbc_obj *) 0xbc0a860
__PRETTY_FUNCTION__ = message_exists
#3 0x2aaac132dab2 in save_to_folder (vmu=0x41751e80, vms=value
optimized out, msg=0, box=value optimized out)
from
[asterisk-users] Record Application
Hi, I am new user of Asterisk. I am using Asterisk version 1.4.15, and I am having the following problem: I am using the Record application to record my SIP channel. With the timeout option, if I don't record (don't speak anything) anything, after the timeout period it comes out sucessfully. Is there way to find out that nothing (no voice activity in the media stream) has been recorded? -KMurali ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] One Question
Hi friends, Redhat9.0 installed in my machine.Asterisk1.0 installed in that machine. Thereis no mpg123 player. So, I download the mpg123 player and installed it. My sound card is configured correctly. When I tried to check asterisk feature SetMusicOnHold its not working im not able to hear any sounds. But the same config (extensions.conf and musiconhold.conf) is worked for Mepis2003 in another machine. Any one can suggest me Thanks in advance Regards Murali___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] One Question:CLI dial cmd
Hi friends, I tried to dial 111 from CLI without any hard/soft phones. I used the following config when i called 111 from CLI by CLI dial 111 I got these errors -- Executing Dial(OSS/dsp, CONSOLE/dsp) in new stack Sep 15 11:57:26 NOTICE[1217602880]: chan_oss.c:753 oss_request: Already have a call on the OSS channel Sep 15 11:57:26 NOTICE[1217602880]: app_dial.c:696 dial_exec: Unable to create channel of type 'CONSOLE' == Everyone is busy/congested at this time -- Executing Hangup(OSS/dsp, ) in new stack == Spawn extension (local, 111, 2) exited non-zero on 'OSS/dsp' Hangup on console ; oss.conf? ; ; Open Sound System Console Driver Configuration File ; [general] ; ; Automatically answer incoming calls on the console? Choose yes if ; for example you want to use this as an intercom. ; autoanswer=yes ; ; Default context (is overridden with @context syntax) ; context=local ; ; Default extension to call ; extension=s ; ; Default language ; ;language=en ; ; Silence supression can be enabled when sound is over a certain threshold. ; The value for the threshold should probably be between 500 and 2000 or so, ; but your mileage may vary. Use the echo test to evaluate the best setting. ;silencesuppression = yes ;silencethreshold = 1000 ; extensions.conf ; [local] exten = 111,1,Dial(CONSOLE/dsp) exten = 111,2,Hangup ANY ONE CAN HELP REGARD THIS ISSUE THANKS IN ADVANCE Regards Murali___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] One Question
Hi friends, I used following commands to configure my zaptel card 1.modprobe zaptel 2.modprobe wct1xxp 3.ztcfg -vvv 4.zttool the problem is when I type zttool command it shows RED Digium Wildcard T100P T1/PRI Card 0 my zaptel.conf look like this span=1,1,0,esf,b8zs bchan=1-23 dchan=24 loadzone=us defaultzone=us the above 5 lines only placed in my zaptel.conf file can any one suggest how to get ok signal. thanks in advance Regards Murali___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] (no subject)
hi all, can anyone give solution for this. wct1xxp - Digium Wildcard T100P T1/PRI Card 0 zttool gives RED Digium Wildcard T100P T1/PRI Card 0 my zaptel.conf look like this span=1,1,0,esf,b8zs bchan=1-23 dchan=24 loadzone=us defaultzone=us the above 5 lines only placed in my zaptel.conf file Regards Murali___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Red Alarm - Config Zaptel card
hi all, can anyone give solution for this. wct1xxp - Digium Wildcard T100P T1/PRI Card 0 zttool gives RED Digium Wildcard T100P T1/PRI Card 0 my zaptel.conf look like this span=1,1,0,esf,b8zs bchan=1-23 dchan=24 loadzone=us defaultzone=us the above 5 lines only placed in my zaptel.conf file Regards Murali___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] one doubt
Hi all, Im using asterisk. I have one doubt. Im running asterisk in one machine(RedHat9.0) running firefly softphone in 3 windows machine I hv 3 users in sip.conf like 1001, 2001 3001 appropriate entry for those users are also include in extensions.conf like -- [mainmenu] exten = 1001,1,Dial(SIP/1001,20,r) exten = 1001,2,Congestion exten = 1001,103,Busy exten = 2001,1,Dial(SIP/2001,20,r) exten = 2001,2,Congestion exten = 2001,103,Busy exten = 3001,1,Dial(SIP/3001,20,r) exten = 3001,2,Congestion exten = 3001,103,Busy I called 1001 from 2001. 1001 got call from 2001. He attend the call. the call is going on. user 3001 try to call 1001. NOW 1001 got call from 3001. eventhough he is speaking with user 2001. Is it correct? When 1001 is talking with 2001. how he will get call from 3001 or any other. I think its wrong. The user 3001 must get message Busy. I need suggestion from any one. please Thanks in advance Regards Murali___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium TE410P and RedHat Enterprise Server 3.0
Hi, I installed TE410P card on RedHat9.0 successfully. If u want any help contact me. On Tue, 17 Aug 2004 Roland Zagler wrote : Hello! has anyone already successfully installed Digium TE410P card on RedHat Enterprise Server 3.0? Roland Zagler mailto:[EMAIL PROTECTED] @fog smart partners ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Regards Murali
[Asterisk-Users] about sip.conf
HI all, Is there any possible to add sip entry 7004 from CLI without open sip.conf like [7004] type=friend username=7004 secret=123 canreinvite=no host=dynamic dtmfmode=rfc2833 mailbox=11 nat=yes Thanks in advance Regards Murali
Re: [Asterisk-Users] Asterisk Receptionist manager program.
Dear, Kyle Hagan wrote: We are writing a program using the manager for * for our receptionist to use once the system go live. If anyone is interested in helping us with testing please let me know. We are designing it for a touch screen monitor for her to do transfers, see whose on the phone and a few other features. Its in the development stage and has bugs. but I think its gonna be really good. If your interested please let me know. Im gonna be putting up a site for downloading if there is enough interest. Thats a good experience for me too, please do the needful. make it fast, keep me updated [EMAIL PROTECTED] We are considering writing a SIP client build into the program at a later time. Kyle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PostgreSQL
Dear, Check whether you have enable tcp/ip socket connection in your Postgres config. postgresql.conf, if yes, see whether u have respective user and password strategy 'trust'. Fabio Donaggio wrote: Hi to all!! Here's my problem: [cdr_pgsql.so] = (PostgreSQL CDR Backend) == Parsing '/etc/asterisk/cdr_pgsql.conf': Found May 26 17:21:35 ERROR[16384]: cdr_pgsql.c:298 my_load_module: cdr_pgsql: Unable to connect to database server localhost. Calls will not be logged! May 26 17:21:35 ERROR[16384]: cdr_pgsql.c:299 my_load_module: cdr_pgsql: Reason: could not connect to server: Connection refused Is the server running on localhost and accepting TCP/IP connections on port 5432? Anyone can help me??? Anyone have some suggest about this or about how to connect PostgreSQL to Asterisk??? Thanks! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] [Fwd: Answer App hanging in I4L]
Original Message Subject: Answer App hanging in I4L Date: Tue, 25 May 2004 13:49:50 +0530 From: Murali Krishnan [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Organization: bk SYSTEMS (P) LTD., To: [EMAIL PROTECTED] Hi, Anyone using ISDN4Linux (Eicon Diva Hisax ) card. If yes, please help me out. After configuring extension.conf and modem.conf I could make outward calls correctly from gnophone and kphone. Still the inward call to the configured MSN is correctly reaching Asterisk and also to the configured Context. But the issue is, it was hanging on 'Answer' application and throwing out 'Unable to Spawn Extension (vpk, s, 1) . . When I debug, found that Asterisk is issuing the following AT commands while answering the call. ATA (expecting VCON ) AT+VRX+VTX (expecting CONNECT ) In the above sequence, I found that after giving ATA, without waiting for VCON it is giving the AT+VRX+VTX command and getting CONNECT successfully, but according to voice communication CONNECT without VCON would fail and hanging up the line. Though all the above things are my Points on debugging, my basic issue is to successfully ANSWER an incoming call. Please throw some lights. Regards Murali Krishnan.S [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Troubles with Kphone]
Original Message Subject: Re: [Asterisk-Users] Troubles with Kphone Date: Tue, 25 May 2004 15:44:15 +0530 From: Murali Krishnan [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Organization: bk SYSTEMS (P) LTD., To: [EMAIL PROTECTED] References: [EMAIL PROTECTED] enano wrote: Hi , I'm triying to use kphone 4.02, but when i'm make a call the programs doesn't respond any command, so i can't hear any sound .. in sip.conf that's my codec config: disallow=all allow=gsm allow=ulaw allow=ilbc and the kphone give the follow : SipClient: Sending: 06:46:28.116 ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.2;rport CSeq: 6121 ACK To: sip:[EMAIL PROTECTED];tag=as12aab0bf From: ivan2 sip:[EMAIL PROTECTED];tag=7F6911ED Call-ID: [EMAIL PROTECTED] Content-Length: 0 User-Agent: kphone/4.0.2 Contact: ivan2 sip:[EMAIL PROTECTED];transport=udp res_search: NO result ! res_search: NO result ! SipClient: Sending to '192.168.0.3:5060' SipCallMember: localStatusUpdated: 200 CallAudio: Using GSM for output CallAudio: Sending to remote site 192.168.0.3:19696 UDPMessageSocket::SetTOS: Operation not permitted CallAudio: OSS device already open (readwrite) anyone can help me ?? thanks Ivan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Check the following things. 1. Make sure your sound card is configured properly for record/playback - if not, do it with either kmix and test with gnome-sound-recorder 2. Make sure your identity is configured in sip.conf and extension.conf correctly 3. Make sure kphone is registered with Asterisk File-Identity - see whether 'Unregister' is there, (means you are registered ) 4. Watch for Asterisk Messages for any clue. ( asterisk -vc ) 5. Make sure the destination extension you are dialing from kphone has proper dialplan sequence in extension.conf 6. If you have OSS sound configuration, immediately switch to ALSA. - visit alsa-project.org and search docs for your card type. Compile and install the packages. ( this OSS would be the major headache if you are not getting sound ). If you are registered with Asterisk and your sound card is proper, and you configured your destination extension routing properly in extension.conf everything should work fine. Get back with success. Regards Murali Krishnan. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Answer App hanging in I4L
Dear Jason, Thanks for your response. Here below is the configuration section of 'vpk' context in extension.conf. start [vpk] exten = s, 1, Answer exten = s, 2, SetMusicOnHold(default) exten = s, 3, DigitTimeout,5 exten = s, 4, ResponseTimeout, 5 exten = s, 5, Background(bks_wlcmmenu) exten = s, 6, Playback(invalid) exten = s, 7, Hangup [default] include = vpk end note, I have context=vpk in modem.conf Just to throw light. But I suppose, the extension configuration is not the problem, because I could see ( from asterisk messages ), the dial plan sequence is going correctly if I comment 'Answer' application. ( though not useful ) I suppose the problem could be in 'Answer' ing mechanism for the call. Please pin point the issue. Thanks in Advance Jason Williams wrote: At 16:00 25/05/2004 +0530, you wrote: But the issue is, it was hanging on 'Answer' application and throwing out 'Unable to Spawn Extension (vpk, s, 1) . . Do you have an extension s in context vpk ? Can you provide the relevant section from the extensions.conf Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
