Project Detail

Asterisk Voicemail.app revision

Asterisk Voicemail.app revision is project number 471725
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Status:

Selected Providers: mkandhuri

Budget: $250-750

Created: 07/19/2009 at 3:00 EDT

Bid Count: 13

Average Bid:
$ 354

08/05/2009 at 3:00 EDT

Project Creator: chrisG00
Employer Rating: 10/10 10/10 10/10 10/10 10/10 10/10 10/10 10/10 10/10 10/10 (1 reviews)

Description

Hello,
We are using asterisk for its voicemail features, connection is by Sip softphones only.

The asterisk 1.6 edition voicemail app does all that we need out of the box, but we need a small change to the vm_exec programming contained in the app_voicemail.c.

the vm_exec asks for a mailbox number (vm-whichbox) and plays back the message inside the mailbox, and then passes the info to leave_voicemail for the user to leave a voicemail.

We need to add to the vm_exec or the leave_voicemail feature, to enable the user to repeat the message in the mailbox, so the new path through the app_voicemail.c script is :-

user dials in to voicemail

user is asked 'which mailbox' ( (vm-whichbox) played back by vm_exec )

user enters mailbox number and mailbox is played back

new code -> user is offered 'press 1 to repeat this message, press 2 to reply to this message'

new code -> if user press 1, message is repeated, (repeated as many times as user requests, max 20)

new code -> if user press 2, the user goes to default path which is 'please leave your message after the tone, when done hangup or press the # key'

thats the change that is needed.

You can take our copy of app_voicemail.c and add revisions to that, or use your own.

Of course we'll need to test the revision, and will ask you to discuss the best method to ensure both parties are happy.

On the point of testing, we have been placing scripts for development for over 8 years, some get completed, some are discarded with a (in some cases) significant loss to us, this work is not of a high enough value to justify any kind of surety to either party, so a degree of trust will need to prevail, for both sides.

Many Thanks

chrisG

Additional files submitted:
app_voicemail.c

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Job Type

See more: add, app, asterisk, code, copy, development, max, number, press, programming, script, sip, small, test, voicemail, work

Messages Posted:0

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PMB

400

3 days

07-19-2009 16:23 EDT

9/10 9/10 9/10 9/10 9/10 9/10 9/10 9/10 9/10 9/10
(1 reviews)

I have exp. in asterisk 1.2 to asterisk 1.6.2 , Please check pm.

Andrew121

400

5 days

07-19-2009 07:09 EDT

(No Feedback Yet)

Ready to go!

magpiesolutions

300

2 days

07-19-2009 08:07 EDT

(No Feedback Yet)

We can do it. We have experience with asterisk.

dworks

400

5 days

07-19-2009 08:38 EDT

(No Feedback Yet)

Hi I have working experience on SIP.

mantisa

500

20 days

07-19-2009 08:43 EDT

(No Feedback Yet)

I am an experience professional expert in C/C++. I provide you professional look. I can assure you a great output of this project. I have reviewed your project and I am ready to provide you a high quality solution for the same and you are most welcome with all of your queries.

lsdumbrella

300

3 days

07-19-2009 12:13 EDT

(No Feedback Yet)

I seems what I can do that.

nevotech

300

15 days

07-19-2009 14:13 EDT

(No Feedback Yet)

Advanced IVR is made by us and used in big telecom project. Please, check PM.

stwrtmk

500

5 days

07-20-2009 06:57 EDT

(No Feedback Yet)

We are Asterisk service provider, we are leaders in providing turnkey solutions in Asterisk * Asterisk: IVR/IP PBX/SOFT SWITCH * Asterisk Gateway Interface (AGI) * End to End VOIP Set-up * VOIP Minutes * Predictive Dialer * Call Centre Solutions * Hospitality Telecom solutions. * Medical Telecom solutions. * ACD for PRI/Analog/GSM Phones * Softphone * SER/billing soft switch * LCR/CDR * Web to call * Click to call * Inter-office/Intra-office Intercom * IP Phones * IP PBX/ Installation / maintenance / configuration of linux systems / servers VOIP Gatekeepers / Phones / devices. * Support for digium / sangoma / rhino E1 / PRI / FXO Digital / analog Telephony Cards /grandstream Regards, AstNext

mukulg2001

400

15 days

07-20-2009 08:52 EDT

(No Feedback Yet)

Hello, I do not have any feedback yet. But I have worked on C and asterisk in my work experiences. Code may be ready between 1 to 15 days once chosen. Depending upon the days for max 15 days I will charge 400 but if the work is finished than that then you can give me less. Thanks Regards

shillin

250

1 day

07-21-2009 21:53 EDT

(No Feedback Yet)

I can do this through the dialplan without having to change the c code and recompiling.

ylber

250

3 days

07-23-2009 18:03 EDT

(No Feedback Yet)

Hi i can do this for you

powerbuilder

300

7 days

07-28-2009 10:58 EDT

(No Feedback Yet)

we are working on customized asterisk development also we have our own developed module like,meetme,voicemail,pbx_config,Dial and etc.

pocosolutions

300

2 days

08-02-2009 13:31 EDT

(No Feedback Yet)

Please check PMB

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