RE: [Asterisk-Users] run with incorrect E1/T1 jumper settings
a better way is to to load the driver with all spans set to E1 by running modprobe wcte11xp t1e1override=15 or edit wcte11xp.c and change 'static int t1e1override = -1;' to 'static int t1e1override = 15;' Diyanat From: Robert Andersson [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] run with incorrect E1/T1 jumper settings Date: Tue, 28 Feb 2006 10:36:57 +0100 Precedence: list Hi, I have installed an TE110P but forgot to change the jumper settings to E1. I don't have easy physical access to ther server at the moment so I wonder if it will be possible to run it without changing the jumper settings with a configuration like below or will it be impossible to use the card at all before I fix the jumper? I can't try it myself yet since the operator isn't ready yet, but I would like to know in advance if it is impossible. bchan=1-15,17-24 dchan=16 instead of bchan=1-15,17-31 dchan=16 best regards Robert ___ --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 ___ --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
RE: [Asterisk-Users] Server Specification
well roughly 80 calls on g729 or 120 on g711, figures may differ in realtime, 100 gb bandwidth may not be sufficient, you will have to know the actual throughput too you should check this tool for bandwidth calculation http://www.asteriskguru.com/bandwidth_calculator.php Diyanat From: Abdul Lateef [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Server Specification Date: Thu, 12 Jan 2006 12:09:51 -0800 (PST) MIME-Version: 1.0 Hi All, I was making plan to set an VoIP Gateway in India. And found some copanies who offered me to host my Asterisk server. I will be appriciated if anyone can suggest me how much simultaneous calls can be handeled with the following server specification? CPU : Dual Intel® Xeon® Processor at 2.8GHz Memory : 512 MB Hard Drive : 2 x 40GB 7.2K RPM Serial ATA Hard Drive Bandwidth : 100GB/MONTH HD Configuration : 2 Hard drives, Motherboard SATA RAID1 : Yes Port : 10/100MBPS SWITCHED VLAN Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 ICQ: 276994704 MSN: [EMAIL PROTECTED] GoogleTalk: [EMAIL PROTECTED] YM!: abdul_zu Doha Qatar http://www.hatif.com __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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 ___ --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
RE: [Asterisk-Users] Transfer to meetme on different server
add to iax.conf on server1 register = username:[EMAIL PROTECTED] on server1 lets say extension 1001 on server1 will transfer the call to extension 1002 on server2 exten = 1001,1,Dial(IAX2/[EMAIL PROTECTED]) ; replace server2 with ip/domain of server2 on server 2 extension 1002 will join a meetme conference room 999 exten = 1002,1,Meetme(999) to choose a dynamic generated room exten = 1002,1,Meetme(|d) Hope that helps Diyanat From: Steven Langley [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] Transfer to meetme on different server Date: Wed, 11 Jan 2006 11:31:54 +0200 Return-Path: [EMAIL PROTECTED] X-OriginalArrivalTime: 11 Jan 2006 09:36:19.0107 (UTC) FILETIME=[79CBBF30:01C61692] Hi there I am using IAX2 based phones and am wondering if the following is possible: 1. User registers with Server 1 2. User dials an extension on Server 1 3. Extension transfers call to an extension on Server 2, which transfers the call to a Meetme conference. If this is possible, would anyone be able to give me an idea how this can be done? Many thanks Steven ___ --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 ___ --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
RE: [Asterisk-Users] Zap hangup issue
You may enable polarity reversal on the line, ask your telco about it then add the following to zapata.conf hanguponpolarityswitch=yes answeronpolarityswitch=yes you can also use a call progress detector such as http://www.broadcastboxes.com/products/CP-2_lit.html Diyanat From: Olivier Taylor [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: 'Asterisk Users Mailing List - Non-Commercial Discussion'asterisk-users@lists.digium.com Subject: [Asterisk-Users] Zap hangup issue Hi all, We are located in Belgium and using an asterisk as internal Pbx. We have many problems with Zap lines, in fact, very often, Zap doesn't release the line after a call or an unanswered call. Any idea is welcome, Olivier ___ --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 ___ --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
RE: [Asterisk-Users] Monitor Logged in Agent's conversation
You can use ChanSpy() module to monitor Diyanat From: Rajkumar S [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Errors-To: [EMAIL PROTECTED] Return-Path: [EMAIL PROTECTED] X-OriginalArrivalTime: 08 Jan 2006 19:40:13.0831 (UTC) FILETIME=[58229970:01C6148B] Hi, Is it possible to monitor conversation of logged in Agents? Currently I am using ZapScan to monitor incoming calls, but I would like to monitor individual agents. raj ___ --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 ___ --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] Meetme user join/leave
Hi The new meetme i feature in asterisk1.2.1 for annoucing user join/leave is good, but the initial steps to record the name and confirm seems lenghty, the user shoudl just say the name and get into the conference, How can i disable the confirmation of the name recorded before entering the conference Diyanat ___ --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] tdm dialout delay
Hello!
I am using asterisk 1.2.1 with a digium TDM card and trying to reduce the
dialout delay to 1/2 secs at the most, i could bring it down from 6/7
seconds to 3/4 seconds by tweaking the config and tone/zone/dtmf settings in
the source, still this is not acceptable as the regular pstn phone takes
less than 1 sec to ring on the called number
Dec 30 03:51:44 VERBOSE[13144] logger.c: -- Executing
Dial(SIP/1010-3211, Zap/g0/011234567) in new stack
Dec 30 03:51:44 DEBUG[13144] rtp.c: Channel 'Zap/1-1' has no RTP, not doing
anything
Dec 30 03:51:44 DEBUG[13144] chan_zap.c: Dialing '011234567'
Dec 30 03:51:44 DEBUG[13004] channel.c: Avoiding initial deadlock for
'Zap/1-1'
Dec 30 03:51:44 DEBUG[13004] channel.c: Avoiding initial deadlock for
'Zap/1-1'
Dec 30 03:51:44 DEBUG[13004] channel.c: Avoiding initial deadlock for
'Zap/1-1'
Dec 30 03:51:44 VERBOSE[13144] logger.c: -- Called g0/011234567
1 sec delay ?
Dec 30 03:51:45 DEBUG[13144] chan_zap.c: Exception on 16, channel 1
Dec 30 03:51:45 DEBUG[13144] chan_zap.c: Got event Hook Transition
Complete(12) on channel 1 (index 0)
3 secs delay ?
Dec 30 03:51:48 DEBUG[13144] chan_zap.c: Exception on 16, channel 1
Dec 30 03:51:48 DEBUG[13144] chan_zap.c: Got event Dial Complete(9) on
channel 1 (index 0)
Dec 30 03:51:48 DEBUG[13144] chan_zap.c: Enabled echo cancellation on
channel 1
Dec 30 03:51:48 VERBOSE[13144] logger.c: -- Zap/1-1 answered
SIP/1010-3211
[EMAIL PROTECTED] asterisk]# ztcfg -vvv
Zaptel Version: SVN-trunk-r880M
Echo Canceller: KB1
Configuration
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
Channel 02: FXS Kewlstart (Default) (Slaves: 02)
Channel 03: FXS Kewlstart (Default) (Slaves: 03)
Channel 04: FXO Kewlstart (Default) (Slaves: 04)
4 channels configured.
zaptel.conf
# Span 1: WCTDM/0 Wildcard TDM400P REV I Board 1
fxsks=1
fxsks=2
fxsks=3
fxoks=4
loadzone= kr
defaultzone = kr
zapata.conf
[channels]
language=en
loadzone =kr
progzone =kr
signalling=fxs_ks
context=from-pstn
group=0
channel = 1
channel = 2
channel = 3
signalling=fxo_ks
context = from-internal
group=1
channel = 4
usecallerid=yes
callerid=asreceived
hidecallerid=no
callwaiting=yes
usecallingpres=yes
;callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=400
rxgain=0.0
txgain=0.0
group=0
callgroup=1
pickupgroup=1
immediate=no
faxdetect=incoming
busydetect=no
hanguponpolarityswitch=yes
answeronpolarityswitch=yes
extensions.conf entry
exten = s,1,Dial(Zap/g0/${EXTEN:1},20,tr)
exten = s,2,Hangup()
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[Asterisk-Users] TE410P E1 Red Alarm
Hello! I have a TE410P quad span card with 4 E1, i am using asterisk 1.2.1, i was using it without any issues earlier with just 1 E1 on span 1 and i recently plugged in 3 more E1's, only span 1 is working, the e1 for span 1 is from a different provider then the rest, the settings are same for both, but i constantly get red alaram on the span 2,3,4, i tried all settings, including the timming source , framing, coding , signalling type etc, without any sucesss what maybe the cause of the red alarm Regards Diyanat lspci -vvv 06:01.0 Communication controller: Unknown device d161:0410 (rev 02) Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr+ Stepping- SERR+ FastB2B- Status: Cap- 66Mhz- UDF- FastB2B- ParErr- DEVSEL=medium TAbort- TAbort- MAbort- SERR- PERR- Latency: 64 Interrupt: pin A routed to IRQ 74 Region 0: Memory at fdff (32-bit, non-prefetchable) [size=128] lsmod Module Size Used byNot tainted wct4xxp78432 124 zaptel183776 250 [wct4xxp] cat /proc/zaptel/* Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 HDB3/CCS/CRC4 1 TE4/0/1/1 Clear (In use) upto 16 TE4/0/1/16 HDLCFCS (In use) upto 31 TE4/0/1/31 Clear (In use) Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 HDB3/CCS/CRC4 RED 32 TE4/0/2/1 Clear (In use) upto 47 TE4/0/2/16 HDLCFCS (In use) upto 62 TE4/0/2/31 Clear (In use) Span 3: TE4/0/3 T4XXP (PCI) Card 0 Span 3 HDB3/CCS/CRC4 RED 63 TE4/0/3/1 Clear (In use) upto 78 TE4/0/3/16 HDLCFCS (In use) upto 93 TE4/0/3/31 Clear (In use) Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 HDB3/CCS/CRC4 RED 94 TE4/0/4/1 Clear (In use) upto 109 TE4/0/4/16 HDLCFCS (In use) upto 124 TE4/0/4/31 Clear (In use) ztcfg -v SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 2: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 3: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) SPAN 4: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) 124 channels configured. zttool Alarms Span OK T4XXP (PCI) Card 0 Span 1 RED T4XXP (PCI) Card 0 Span 2 RED T4XXP (PCI) Card 0 Span 3 RED T4XXP (PCI) Card 0 Span 4 CLI zap show status Description Alarms IRQbpviol CRC4 T4XXP (PCI) Card 0 Span 1OK 0 0 0 T4XXP (PCI) Card 0 Span 2RED0 0 0 T4XXP (PCI) Card 0 Span 3RED0 0 0 T4XXP (PCI) Card 0 Span 4RED0 0 0 alpha*CLI pri show span 1 Primary D-channel: 16 Status: Provisioned, Up, Active Switchtype: EuroISDN Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 1 T305 Timer: 3 T308 Timer: 4000 T313 Timer: 4000 N200 Counter: 3 alpha*CLI pri show span 2 Primary D-channel: 47 Status: Provisioned, In Alarm, Down, Active Switchtype: EuroISDN Type: CPE Window Length: 0/7 Sentrej: 0 SolicitFbit: 0 Retrans: 0 Busy: 0 Overlap Dial: 0 T200 Timer: 1000 T203 Timer: 1 T305 Timer: 3 T308 Timer: 4000 T313 Timer: 4000 N200 Counter: 3 its the same as span for the rest as upto span 4 CLIpri intense debug span 2 Unnumbered frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 M3: 3 P/F: 1 M2: 3 11: 3 [ SABME (set asynchronous balanced mode extended) ] 0 bytes of data Sending Set Asynchronous Balanced Mode Extended same for the rest zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan = 1-15 dchan = 16 bchan = 17-31 span=2,1,0,ccs,hdb3,crc4 bchan = 32-46 dchan = 47 bchan = 48-62 span=3,1,0,ccs,hdb3,crc4 bchan = 63-77 dchan = 78 bchan = 79-93 span=4,1,0,ccs,hdb3,crc4 bchan = 94-108 dchan = 109 bchan = 110-124 loadzone=se defaultzone=se zapata.conf [channels] language=us context=sip switchtype=euroisdn pridialplan=unknown prilocaldialplan=unknown signalling=pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes cidsignalling=dtmf cidstart=ring rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no callerid=asreceived group=1 channel = 1-15 channel = 17-31 group=2 channel = 32-46 channel = 48-62 group=3 channel = 63-77 channel = 79-93 group=4 channel = 94-108 channel = 110-124 ___ --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
RE: [Asterisk-Users] cdr mysql problem
i am using asterisk 1.2.1 with mysql 5 without any issues, please check your configuration again, make sure you have hostname=localhost too and the dbname, user, password are correct [global] hostname=localhost dbname=databasename user=user password=password port=3306 sock=/var/lib/mysql/mysql.sock Diyanat From: Mohammad Shokuie [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] cdr mysql problem Return-Path: [EMAIL PROTECTED] Dear folks, I've just compiled asterisk-addon1.2.1 after installing MySQL and MySQl-devel packages. and adjust my cdr_mysql.conf to use the defined database using username and password. But as soon as starting asterisk i get error messages informing me of error, error message is as follows : cdr_addon_mysql.c : Failed to connect mysql database cdr on localhost and res_config_mysql.c : Failed to connect database server on . Im realy lost and dont know whats wrong. I've checked the connection to MySql in command line using the same user and host and its been connected without any problem. Anyone has any idea whats wrong here. Regards. --- M. Shokuie Nia. _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --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 ___ --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
RE: [Asterisk-Users] Shutting down Asterisk when not in RTP Stream
Do you have 't' or 'T' in the Dial Application? Diyanat From: Douglas Garstang [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: [Asterisk-Users] Shutting down Asterisk when not in RTP Stream Date: Thu, 15 Dec 2005 10:38:22 -0700 MIME-Version: 1.0 I'm very confused about something. I have two phones that have reinvited and have an RTP session open. I confirmed this by running ngrep on the Asterisk box. Asterisk still shows the calls on the console. *CLI sip show channels Peer User/ANRCall ID Seq (Tx/Rx) Form Hold Last Message 192.168.10.125 a00090201 45dfabad1bd 00103/0 ulaw No Tx: ACK 192.168.10.4 a00090101 ca3279d8-3e 00102/1 ulaw No Tx: ACK When I shut asterisk down, the call terminates. I don't understand that. If Asterisk isn't in the RTP path, how can shutting it down terminate an active call? Don't know if it's relevant, but the 192.168.10.4 is an OpenSER box. Thanks. Doug. ___ --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 ___ --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
RE: [Asterisk-Users] getting started
check http://www.asteriskguru.com/tutorials/ Diyanat From: sukrit [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] getting started Date: Fri, 16 Dec 2005 09:06:28 +0530 MIME-Version: 1.0 Hi Guys, Wanted some advice for the docs that you'd recommend someone new to Asterisk to read. I have a good knowledge of Unix and networking, so that part shouldn't be a problem. Cheers, Sukrit.D. -- \|||/ (o o) +ooO-(_)-Ooo---+ | SUKRIT D| www.liqwidkrystal.com| | Email: sukrit-at-liqwidkrystal.com | |--| | MSN:[EMAIL PROTECTED] YAHOO:sd_root | | SKYPE:sukritd | +--+ ( _ ) _| | | |_ (___| |___) ___ --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 ___ --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
RE: [Asterisk-Users] SIP Trunk please help
in the sip.conf have the following enteries
; for regsitering with ser
register:seruser:[EMAIL PROTECTED]:5060;(put ser machine ip:port)
;add a user for the ser machine
[seruser]
type=friend
host=0.0.0.0 ;(put ser machine ip here)
nat=no ;(change as needed )
canreinvite=yes ;(change as needed)
insecure=very ;(change as needed)
disallow=all
allow=ulaw
allow=gsm
context=sip
dtmfmode=rfc2833
in extensions.conf under contect [sip]
[sip]
;replace extension and the priority to macth your dial plan
exten = _X.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED]) ;(seruser is defined in
sip.conf)
Diyanat
From: Ryan Pagquil [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] SIP Trunk please help
Date: Fri, 16 Dec 2005 10:31:24 +0800
MIME-Version: 1.0
Hi,
I've been setting up asterisk for prepaid use. I'm testing to call a SER
registered user from the Asterisk just to simulate the prepaid calls. Now,
I can already contact Asterisk and it prompts me to input my call card
number and after that I dial in the number I want to call (a SER registered
device). My question is how can I implement on sip.conf to use my SER as
the trunk line? So that calls will be forwarded to it. Do I also need to
register asterisk on SER?How?
Please help!
Thanks,
Ryan
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RE: [Asterisk-Users] SIP Trunk please help
yes
$AGI-exec('Dial', SIP/[EMAIL PROTECTED]);
Diyanat
From: Ryan Pagquil [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com,
asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] SIP Trunk please help
Date: Fri, 16 Dec 2005 13:56:09 +0800
MIME-Version: 1.0
X-OriginalArrivalTime: 16 Dec 2005 05:58:00.0170 (UTC)
FILETIME=[AB7B14A0:01C60205]
Hi,
Thanks for the reply... Actually I'm using AGI to do it instead of
defining it on extensions.conf... Would it be the same in extensions.conf?
Should I write $AGI-exec('Dial', 'SIP/[EMAIL PROTECTED]'); to dial it from
AGI script (perl), is this correct?
Thank you very much,
Ryan
At 01:45 PM 12/16/05, Diyanat Ali wrote:
in the sip.conf have the following enteries
; for regsitering with ser
register:seruser:[EMAIL PROTECTED]:5060;(put ser machine ip:port)
;add a user for the ser machine
[seruser]
type=friend
host=0.0.0.0 ;(put ser machine ip here)
nat=no ;(change as needed )
canreinvite=yes ;(change as needed)
insecure=very ;(change as needed)
disallow=all
allow=ulaw
allow=gsm
context=sip
dtmfmode=rfc2833
in extensions.conf under contect [sip]
[sip]
;replace extension and the priority to macth your dial plan
exten = _X.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED]) ;(seruser is defined in
sip.conf)
Diyanat
From: Ryan Pagquil [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] SIP Trunk please help
Date: Fri, 16 Dec 2005 10:31:24 +0800
MIME-Version: 1.0
Hi,
I've been setting up asterisk for prepaid use. I'm testing to
call a SER registered user from the Asterisk just to simulate the prepaid
calls. Now, I can already contact Asterisk and it prompts me to input my
call card number and after that I dial in the number I want to call (a
SER registered device). My question is how can I implement on sip.conf to
use my SER as the trunk line? So that calls will be forwarded to it. Do I
also need to register asterisk on SER?How?
Please help!
Thanks,
Ryan
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RE: [Asterisk-Users] Join when empty problem, in queue
in queues.conf change joinempty = no leavewhenempty = no to joinempty = strict leavewhenempty = strict From: Xavier Gil [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Join when empty problem, in queue Date: Wed, 14 Dec 2005 10:19:08 +0100 (CET) MIME-Version: 1.0 Hi all, when calling to a queue that has no agents logged in we expect to hang up, here is the extensions.conf queue configuration. exten= 2020,1,Answer exten= 2020,2,Ringing exten= 2020,3,Wait(2) exten= 2020,4,Queue(gestoria) exten= 2020,5,Hangup But althougth there isn't any agent it let us enter in the queue. Any idea? Here is the queues.conf: [gestoria] musiconhold = default strategy = ringall servicelevel = 40 context = default timeout = 25 retry = 10 ;weight=0 ;wrapuptime=15 maxlen = 0 announce-frequency = 120 periodic-announce-frequency=60 announce-holdtime = no announce-round-seconds = 10 monitor-format = gsm monitor-join = no joinempty = no leavewhenempty = no eventwhencalled = no eventmemberstatusoff = yes reportholdtime = yes memberdelay = 0 timeoutrestart = no member = Agent/1001 member = Agent/1002 We are using the asterisk from svn repository. __ Renovamos el Correo Yahoo! Nuevos servicios, más seguridad http://correo.yahoo.es ___ --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 ___ --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
