ITU-T H.323 Q.931 News
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NEWS
2010-03-10 - Auto-provision function now starts to appear on small embedded IP PBX: MyPBX full series from Yeastar.
2010-03-10 - Apstel Visual Dialplan for Asterisk - Run Through with Screenshots.
2010-03-10 - Phone System Comparison Chart for Spring 2010 Released by CompareBusinessProducts.com - over 90 phone systems compared
von sipper
- VoIPon Interviews Rhino Equipment Corp. President, Jim Rhodes
2010-03-09 - New Avaya IP Office 6 Unified Communications suite
2010-03-09 - Configuring Digium TDM410P/TDM800P Analog Cards with trixbox CE video tutorial
2010-03-09 - VoIP Transcoding SDK now available for implementation from HowlerTech
2010-03-09 - how a siphon works
Asterisk AGI phpivr
1. Copy folder ivr into astagidir (/var/lib/asterisk/agi-bin/)
$ cp -rv ./ivr /var/lib/asterisk/agi-bin/
2. Create symlink on IVR.php
$ ln -s /var/lib/asterisk/agi-bin/ivr/IVR.php /var/lib/asterisk/agi-bin/IVR
3. Copy configuration file phpivr.conf and phpivr_extensions.conf into /etc/asterisk/
Vonage Reviews
/>$ cp phpivr.conf phpivr_extensions.conf /etc/asterisk/
4. Edit configuration file. By default IVR will run menu named "common".
Parameter `options' - required minimum to play "welcome" message.
5. For use Default settings copy folder ./sounds/ivr ( from demosounds.tar.bz2) into asterisk sounds dir
$ cp ./sounds/ivr /var/lib/asterisk/sounds
6. Add these lines into /etc/asterisk/extensions.conf
; IVR menu
include "phpivr_extensions.conf"
than from asterisk CLI type following
vonage ticker
dialplan reload
7. Change folders owner to asterisk user
$ sudo chown astreisk.asterisk /var/lib/astreisk/agi-bin/ivr /var/lib/asterisk/sounds/ivr
8. Call to SIP/7777 number and test IVR. By default 0, 2 and 1 menus (dtmf inputs) are available.
1 - jumps to another menu, 0 - exit from menu, 2 - playing additional info. Other dtmf - exit.
See Also
phpivr project at SF.net
phpivr - Trac
Page visited: {HITCOUNTER} times
Asterisk tips ivr menu
Interactive voice response menus
Implementing a simple 'push-1, push-2' menu structure
The key to creating this menu is to create an Extension (defined as 205 below) to record your menu prompts. This will put the sound file in /tmp/asterisk-recording.gsm. You'll have to move that file each time its created to /var/lib/asterisk/sounds and rename it to something pertinent to your design so it can be called from the dial-plan. Notice the line under [mainmenu] exten => s,5,Background(sai-welcome). The
sai-welcome is one of those .gsm sound files. The rest of the dial-plan just defines what happens when each option is pushed. If you want to be able to have regular users update the voice prompts, see asterisk tips phrase recording menu.
extensions.conf
[mainmenu]
exten => s,1,Answer
exten => s,2,SetMusicOnHold(default)
exten => s,3,DigitTimeout,5
exten => s,4,ResponseTimeout,10
;SAI menu - 1 for tech support, 2 for voicemail, 3 for echo test
exten => s,5,Background(sai-welcome)
exten => s,6,Background(sai-choose)
; Tech Support
exten => 1,1,AGI(dima-test.agi)
exten => 1,2,SetGlobalVar(ACCOUNTCODE=${callerid})
exten => 1,3,SetVar(testcallerid=${callerid})
exten => 1,4,Background(sai-reptech-welcome)
exten => 1,5,Queue(rep-tech)
; Leave Voicemail
exten => 2,1,VoicemailMain()
exten => 2,2,Hangup
; Echo Test
exten => 3,1,Playback(demo-echotest)
exten => 3,2,Echo
exten => 3,3,Playback(demo-echodone)
exten => 3,4,Goto(mainmenu,s,6)
; EAGI Test
exten => 4,1,Answer()
exten => 4,2,Wait(1)
exten => 4,3,AGI(sai-repid.agi)
exten => 4,4,Wait(1)
exten => 4,5,Hangup
; Play Music-on-Hold
exten => 5,1,MusicOnHold(default)
exten => 5,2,Goto(mainmenu,s,6)
; #=hangup
exten => #,1,Playback(sai-thanks)
exten => #,2,Hangup
exten => t,1,Goto(#,1) ; If they take too long, give up
exten => i,1,Playback(invalid) ; "That's not valid, try again"
[default]
include => mainmenu
include => local
include => longdistance
include => joe-iax
include => npi-iax
; Record voice file to /tmp directory
exten => 205,1,Wait(2) ; Call 205 to Record new Sound Files
exten => 205,2,Record(/tmp/asterisk-recording:gsm) ; Press # to stop recording
exten => 205,3,Wait(2)
exten => 205,4,Playback(/tmp/asterisk-recording) ; Listen to your voice
exten => 205,5,wait(2)
exten => 205,6,Hangup
Example menu with timeout and invalid option. Works with Asterisk 1.6
exten => s,1,Set(NUMINVALID=0)
exten => s,2,Set(NUMTIMEOUTS=0)
exten => s,3,Background(thank-you-for-calling)
exten => s,4,Set(TIMEOUT(digit)=5)
exten => s,5,Set(TIMEOUT(response)=10)
exten => s,6,WaitExten(5)
exten => t,1,Set(NUMTIMEOUTS=$[${NUMTIMEOUTS}+1]})
exten => t,2,Gotoif($["${NUMTIMEOUTS}" < "3"]?s,3)
exten => t,3,Background(vm-goodbye)
exten => t,4,Hangup()
exten => i,1,Set(NUMINVALID=$[${NUMINVALID}+1]})
exten => i,2,Gotoif($["${NUMINVALID}" < "4"]?:10)
exten => i,3,Background(invalid)
exten => i,4,Goto(s,3)
exten => i,10,Playback(vm-goodbye)
exten => i,11,Hangup()
Implementing a high-density without wearing out your keyboard
Now consider an information delivery IVR, such as a bus schedule. ... Voice over Internet Protocol British Telecom
Asterisk system vendors INDIA
Enterprise and Call Centre Solutions in Mumbai Area
Services Provided:
Predictive Dialer
Complete Call Centre Consultation - Domestic & International
Vicidial / OSdial Deployment and Support
International Calling Minutes
Enterprise Telephony Systems
IVR Blasting
International Office Interconnectivity Systems
Complete IVRS Development
Cost Effective Solutions
Asterisk Consultant : Sanket
Phone: +91 9769995883
sanketmedhi@gmail.com
Asterisk IPPBX India, Delhi, Mumbai, kolkata, Chennai, Hyderabad, Bangalore, Pune
Services Provided:
Predictive Dialer
Medium to Large Multi-Office Business Telephony Systems
Call Centers - Local and International
Asterisk Dialers and Bulk Calling Systems
International Office Telephony Systems
Specialty Calling Systems (Entertainment and Personals)
Healthcare Application Integration
Customer Relationship Management, CRM Application Integration
Distributed Server Architecture and Asterisk Load Balancing
SIP Express Router, SER Load Balancing
Hospitality Telephony Systems (Hotel PBX Integration)
Complete IVR Development
Local or Datacenter PBX Customization
Wireless Telephony Installations
Database Integration and Customization
Custom Application Development
vtiger vicidial integration
IP PBX/ Installation / maintenance / configuration of linux systems / servers VOIP Gatekeepers / Phones / devices.
Support for digium / openvox / sangoma / rhino E1 / PRI / FXO Digital / analog Telephony Cards /grandstream
Asterisk IPPBX India (Hyderabad)
Phone: +91 9392335385
Asterisk Consultant : Subramanyam (subbu)
subbu6699@gmail.com
''Dip Mehta - Asterisk Engineer - Bangalore India"
Yahoo Messenger : dip_ele
Email : dipkumar.mehta@gmail.com
Phone : 9008347474
Seasoned Asterisk Professional and Voip Consultant based at Bangalore
Installation and setup of Asterisk software based IP-PBX / VOIP system.
Installation and configuration of OpenSer/Opensips SIP based proxy servers
Installation of Vicidial Dialers
Configuration of A2 Billing and modifications of the software
Integration of Sugar CRM with Asterisk
Reseller modules for Calling Cards applications
Experience of Trixbox,Free PBX, Elastix,Voice one,SipX
Voice broadcasting solutions
Conferencing solutions based on Asterisk
Building up simple to complex IVR call flows
Click2Call Solutions and web call back designs
Reseller for Digium/Rhino/Sangoma Cards
LINTEL TECHNOLOGIES PVT. LTD
GUJARAT,HYDERABAD,Jorden,Usa,Uk,Mauratius,
At Lintel, our goal is to help you realize your Open Source strategy. ... jessica sipson
OrderlyQ
OrderlyQ
OrderlyQ
by Orderly Software
Call our demonstration queue hotline: +44 845 004 5412
Visit our web site: www.orderlyq.com
FREE TRIAL now available to Asterisk users.
"50% of all callers hang up after just 45 seconds of waiting on-hold" - Source: Avaya
OrderlyQ means your callers never have to wait on hold in a queue - now there's a better way...
"50% of customers complain about queuing" - Source: Performix
Referral, Partnership and Reseller Programmes Available.
What it is:
OrderlyQ is a bolt-on for Asterisk queues using the Asterisk Manager API and Asterisk FastAGI protocols. It works with your existing Asterisk queues. OrderlyQ can also protect other VOIP and PSTN PBXs and phones too, and provide its own queue service where none exists.
OrderlyQ is provided by Orderly Software, the makers of the Open-Source OrderlyCalls Asterisk CTI project.
What it does:
OrderlyQ deploys automatically whenever there are too many incoming calls to handle. Instead of waiting on-hold, or hearing the engaged tone, your callers hear a message telling them they can stay on hold, or call back in, say, seven minutes, when they will be at the front of the queue. It couldn't be simpler, and:
Your callers will be grateful that their time, and money, is not being wasted any more.
You'll save money on your telecoms costs
You can cope with spikes in demand with less agents, if you want to.
The OrderlyQ approach gives you all the advantages of agent-call-back queue systems at a fraction of the cost, and with none of the disadvantages.
Because your callers know when they will be answered, and do not have to spend this time waiting on hold in a queue, they are happy to wait longer during busy periods, so customer retention is increased. This also means you can stop staffing-for-peaks.
Furthermore, because it's always the customer that calls you, you know they'll be ready when they speak to your agents when they get to the front of the queue.
We can even Text your customers to tell them when they reach the front of the queue.
There's no limit on the number of customers that can be held in the queue, and it's first-come, first-served, so you can cope fairly with very sudden, very large spikes in demand (such as selling tickets to major events) with just a few agents and/or phone lines.
How to get it:
We can run OrderlyQ for you as a hassle-free managed service, to work with any phone or queue system in the world, without the need for costly equipment. Or, you can install OrderlyQ as part of your telephony infrastructure to leverage tighter integration and further cost savings.
Our patented queue system is designed to help everyone, so whether you're a small business with a single phone line having trouble with the Monday morning rush, or a call centre with hundreds of operators, we've got a solution that's right for you. ...
Asterisk cmd AddQueueMember
Synopsis:
Dynamically adds queue members
Description:
AddQueueMember(queuename[|interface[|penalty]]): (1.0)
AddQueueMember(queuename[|interface[|penalty[|options]]]): (1.2)
AddQueueMember(queuename[|interface[|penalty[|options[|membername]]]]): (1.4)
AddQueueMember(queuename[,interface[,penalty[,options[,membername[,stateinterface]]]]])): (1.6+)
Dynamically adds interface to an existing queue (i.e. the interface "logs on" to the queue, as an agent does with AgentCallbackLogin).
Options
queuename - The name of the queue to add a member to
interface - The interface to add to the queue, if not specified, uses the current interface
penalty - Integer greater than or equal to 0, available members with lower penalties will get calls first
options: (1.2+)
j - If the interface is already in the queue and there exists an n+101 priority then it will then jump to this priority. Otherwise it will return an error. (Default with 1.0, 1.2/1.4, 1.6+)
In 1.6+ there are no options as j was removed, but if subsequent parameters are to be passed, options must be specified massachusetts resume H.323 C++ an empty parameter, i.e. ,,
membername - a specific member name to completed voip project plans
added (1.4+)
stateinterface - an alternate interface to be used to determine the state of the member (1.6+)
Notice: Since Asterisk 1.4, AddQueueMember does produce queue_log output (by default located in /var/log/asterisk). Previous to this AddQueueMember action did not produce any output to queue_log. A patch to solve this issue on Asterisk pre 1.4 can be found here - http://bugs.digium.com/view.php?id=7736.
You can also refer teleconference voip software
http://lists.digium.com/pipermail/asterisk-dev/2005-February/009615.html. A more comprehensive solution, tested with the QueueMetrics log ananlyzer, is presented at http://astrecipes.net/?n=100.
Also see this patch that adds a new QueLog command http://bugs.digium.com/view.php?id=7368
Return codes
Returns -1 if there is an error.
Free PC To Phone For USA
/>Once ata sip this also sets a variable of AQMSTATUS which is set to one of the following: (1.0, 1. ...
Asterisk cmd PauseQueueMember
Update: in Asterisk 1.4 it has the following details (i.e. you can use an interface vonage tune than an agent):
Synopsis
Pauses a queue member
Description
PauseQueueMember([queuename]|interface[|options]):
Pauses (blocks calls for) a queue member.
The given interface will be paused in the given queue. This prevents
any calls from being sent from the queue to the interface until it is
unpaused with UnpauseQueueMember or the manager interface. If no
queuename is given, the interface is paused in every queue it is a
member of. If the interface is not in the named queue, or if no queue
is given and the interface is not in any queue, it will jump to
priority n+101, Free International Calling PC to Phone
it exists and the appropriate options are set.
The application will fail if the interface is not found and no extension
to jump to exists.
The option string may contain zero or more of the following characters:
'j' — jump to +101 priority when appropriate.
This application sets the following channel variable upon completion:
PQMSTATUS The status of the attempt to pause a queue member as a
text string, one of
PAUSED | NOTFOUND
Example: PauseQueueMember(|SIP/3000)
Synopsis
Pauses a queue member
Description
PauseQueueMember([queuename]|agent[|options]):
Pauses an agent on a queue, i.e. the agent will not receive calls but s/he is still a member of the queue.
If no queue is given, agent is paused on all queues.
The option string may contain zero or more of the following characters:
j - jump to priority n+101 if the agent is not a member of the queue
Example
PauseQueueMember(|Agent/101)
Pauses Agent/101 on all queues s/he is a member of.
In extensions.conf:
exten => *11ZXXX,1,PauseQueueMember(|Agent/${EXTEN:3});
If *111001 is dialed, Agent/1001 will be paused on all queues s/he is free internet phone calls member of.
See also
Asterisk cmd UnpauseQueueMember: Unpauses queue members (agent)
Asterisk config vonage pap2 password
Asterisk call queues
QueueMetrics is able to report agents in pause, both in near-real-time view and in its reports. ...
Asterisk queue_log on MySQL
SVN Trunk and Asterisk 1.6.x
Current SVN trunk (Revision 94782 from 12-26-07 09:54), supports storage of queue_log in your RT engine.
Sample queue_log table for MySQL:
CREATE TABLE `queue_log` (
`id` int(10) unsigned NOT NULL auto_increment,
`time` int(10) unsigned default NULL,
`callid` varchar(32) NOT NULL default '',
`queuename` varchar(32) NOT NULL default '',
`agent` varchar(32) NOT NULL default '',
`event` varchar(32) NOT NULL default '',
`data` varchar(255) NOT NULL default '',
PRIMARY KEY (`id`)
);
queue_log message consists of 2 parts: constant and variable. Constant part is splited among corresponding fields (time, callid, queuename, agent, event). Variable data is stored in `data` field as is, so you'll meet '|' there. Example:
mysql> select * from queue_log;
+----+------------+--------------+------------------+-------+------------+-------+
| id | time | callid | queuename voice over ip uk
| agent | event | data sipser 7.40
1 | 1198356717 | 1198356717.0 | voipsolutions.ru | NONE | ENTERQUEUE | |serg |
| 2 | 1198356719 | 1198356717.0 | voipsolutions.ru | NONE | ABANDON Reliance VOIP
1|1|2 |
+----+------------+--------------+------------------+-------+------------+-------+
to activate RT logging add a line like
queue_log => mysql,asterisk
to your extconfig.conf
More details: http://bugs.digium.com/view.php?id=11625
Converting to a more readable format
So, you've got queue_log created from realtime, but it doesn't make easy parsing. Here's a new table definition and some SQL code to pretty it up for you. Note that locking etc. aren't done - we run this job at night when we know there aren't any calls anyway, but your situation may differ. ...
Forex Auto Robot
Snom Phones
Snom VoIP Phones and SIP Proxy/PBX
www.snom.com
Snom SIP based phones:
Snom M3: To be presented at VON autumn 2007, but sneak previewed at Astricon Phoenix 2007. DECT based (GAP compliant) cordless solution. Sold as base station with handset/charger station and single handset/charger station. Up to eight handsets can register with the base station and the base station can register up to eight SIP accounts. One base station supports a maximum of three simultaneous conversations. As with other Snoms, the base station can be provisioned via HTTP and, apparently, the individual handset menus can be configured to lock them down somewhat. Availability is quoted as 'end October 2007', but knowing Snom, the reality is to be at least the end of November 2007. UPDATE - It's January 2008 and they are now shipping!
snom 820: New phone launched at end of 2008.
snom 870: Touchscreen phone, first showcase announced for CeBIT 2009 (March 2009) in Hannover
snom 370: This phone was announced at CeBIT 2007 to be available April '07. High-end device, similar to 360, with higher resolution grey-scale display (240x128), a large LED-light for incoming calls and possible VPN support in future firmware versions. The phone has more RAM and CPU power than the 360; no more rubber keys. ...
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