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Cisco call Manager and Asterisk SIP Integration

I need support for integrating Cisco Call Manager with Asterisk using SIP Trunk.

I need to receive the calls from cisco to Asterisk.

PLease bid only if you have prior hands on experience in it.

Beceriler: Asterisk PBX, Cisco, Linux, Ağ Yönetimi, VoIP

Daha fazlasını gör: cisco as5400 asterisk sip, cisco 7970 asterisk sip, call asterisk sip, asterisk sip incoming call, asterisk sip call pbx trixbox elastix dialer, asterisk sip call pbx trixbox elastix bill dialer, asterisk sip incoming call number, asterisk sip header guest call, asterisk sip call, asterisk sip incoming guest call, asterisk website integration scheduler call recording, asterisk sip cisco, set sip cisco 7940 asterisk, cisco communicator asterisk sip, asterisk sip call generator

İşveren Hakkında:
( 4 değerlendirme ) Kottayam, India

Proje NO: #15880178

Bu iş için 5 freelancer ortalamada ₹2401 teklif veriyor

vinskendaj

Hi ,i'm Ervin, and i've 8+ years of experience in system administration. Voip field is one of my knowledge on this field. i'd like to help you out with your integration. have done it in the past. Relevant Skills and E Daha Fazla

in %bids___i_period_sub_35% gün içinde2350%project_currencyDetails_sign_sub_37% %project_currencyDetails_code_sub_38%
(8 Değerlendirme)
3.2
nmsandroid

We have experience of asterisk and other call control software But doesn’t have specific for Cisco call manager If documentation can be provided we can look k into

in %bids___i_period_sub_35% gün içinde5555%project_currencyDetails_sign_sub_37% %project_currencyDetails_code_sub_38%
(14 Değerlendirme)
3.3
cristiionica

Have 1 year experience with Cisco Call Manager, haven't worked on Asterisk PBX before but I have in depth SIP knowledge Relevant Skills and Experience CCNA certificate,3 years experience as a network engineer,1 year e Daha Fazla

1 gün içinde %bids___i_sum_sub_32%%project_currencyDetails_sign_sub_33% INR
(1 Yorum)
2.1
ram87kavi94

A proposal has not yet been provided

1 gün içinde %bids___i_sum_sub_32%%project_currencyDetails_sign_sub_33% INR
(0 Değerlendirme)
0.0
sasuazo

I propose to establish a SIP trunk between asterisk and CUCM, making sure that both calling and called party information (name and number) passes through, as well as dtmf, correct codec selection, etc Relevant Skills Daha Fazla

1 gün içinde %bids___i_sum_sub_32%%project_currencyDetails_sign_sub_33% INR
(0 Değerlendirme)
0.0