Difference between revisions of "ZAPTEL outgoing calls"
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==Documentation== | ==Documentation== | ||
* [[MOR PRO Manual | '''MOR Manual''']] | * [[MOR PRO Manual | '''MOR Manual''']] | ||
* Call Routing | |||
** [[Call Routing by priority (Manual LCR)]] | |||
** [[Call routing per destination basis]] | |||
** LCR/Tariff change based on call prefix | |||
** [[Integrity Check]] | ** [[Integrity Check]] | ||
** [[Pay Invoice with balance deduction]] | ** [[Pay Invoice with balance deduction]] | ||
Line 116: | Line 120: | ||
** [[Balance in phone]] | ** [[Balance in phone]] | ||
** [[PBX Functions]] | ** [[PBX Functions]] | ||
** [[Hangupcause Codes]] | ** [[Hangupcause Codes]] | ||
** [[Simultaneous call limitation]] | ** [[Simultaneous call limitation]] |
Revision as of 10:50, 23 November 2008
Welcome to Kolmisoft Wiki Page
Guides
Testing
- Testing MOR PRO installation
- Check GUI log
- Call Tracing
- Asterisk CLI
- app_mor.so version
- VoIP Bandwidth Calculator
- Performance Testing with Sipp
- Watch active calls/channels in Asterisk server from CLI
Installation
- MOR PRO Installation
- RMagick installation
- Zaptel installation
- mod_fcgid installation
- G723/G729 Codec installation
- mime-construct installation - not used anymore
- MySQL Replication
- Sangoma Wanpipe installation
- Debian Etch installation
- Centos installation
GUI
- Change folder /mor to another
- Change flag for translation
- MOR GUI Translation
- Remove Manual link from admin GUI
- Lost MOR GUI admin password
DB
Asterisk
- One way audio problems
- Ports which should be opened
- Asterisk under NAT
- Asterisk CLI debug
- Which codecs should I choose for devices
- How to restart Asterisk server
- How to connect to Huawei devices
- How to change RTP port range for Asterisk
- Two SIP listening ports for single Asterisk
Other
- Migration to other server
- Why amounts are without VAT?
- MOR PRO post installation
- MOR PRO upgrade
- Mor user password change HOW-TO
- What is necessary to troubleshoot MOR PRO system
- Where I can get my serial key from
- Change default passwords
- Frequently Asked Questions (FAQ)
- Linksys SPA942 Configuration
- Video call with CounterPath eyeBeam
- How to set static IP in Debian
- Timezone in RoR
- Voice quality
- Sound files
- Send Receive Fax over T38
- Configuring DIDWW
- mor.conf
- NTP: Server Time Sync
- Selinux
- Recommended hardware for MOR server
- Wiki formating guidelines
Documentation
- MOR Manual
- Call Routing
- Call Routing by priority (Manual LCR)
- Call routing per destination basis
- LCR/Tariff change based on call prefix
- Integrity Check
- Pay Invoice with balance deduction
- Transfers with MOR
- How to create H323 providers/devices in the MOR
- PBX connection to MOR
- What is increment
- Cisco Gateway Settings
- Get Google Maps key
- Importing Tariffs from CSV with wrong Regional Settings
- Voicemail
- Why MOR does not allow to delete users
- PayPal
- Margin and Markup
- Provider Configuation
- DIDs
- Balance in phone
- PBX Functions
- Hangupcause Codes
- Simultaneous call limitation
- MOR API
- Number Manipulation
- Asterisk versions not compatible with MOR
- MOR compatible Linux distributions
- Why we do not suggest to use IAX2
- FAQ
- How fast MOR can perform?
- Implementations
- Support System
- Job offer - suspended
- Releases
- MOR Addons
- Auto-Dialer Addon
- Reseller Addon
- Click2Call Addon
- Joomla integration --- Under development
Problems/Troubleshooting
>>> We're sorry, but something went wrong <<<
Asterisk
- Out of idle IAX2 threads
- Asterisk displays wrong time
- SIP Error Codes
- Call rejected by IP: No authority found
- Unable to forward voice frame
- CallerID not passed
- DID is not working
- Calls are not recorded
- Fax is not received
- MOR can't determine who is calling. Make sure accountcode is set for caller (Provider or Device)
- Retrieved 0 adv.rates, max adv.rate: 0.000000, total event price: 0.000000
- One-way audio or not audio at all
- No sound on Voicemail or Playback
- Dropping extra frame of G.729 since we already have a VAD frame at the end
- Comfort noise support incomplete in Asterisk (RFC 3389)
- mysql_reconnect: mor: Unknown connection error: (2006) MySQL server has gone away.
- Call dropped upon connection
- Starting asterisk: Cannot find your TTY
- False Answer - False Ring
- Unable to allocate RTCP socket: Too many open files in system
- Echo
- Asterisk stops working with Internet loss
- Asterisk ended with exit status 1 Asterisk died with code 1
- Unknown signalling method 'pri_cpe'
- ERROR: Could not open H.323 listener port on 1720
- Auto-Dialer: ERROR! No actions found
- sip_poke_noanswer: Peer 'XXX' is now UNREACHABLE!
- Unknown RTP codec 127 received
- CLIR for ZAP channels
- determine_firstline_parts: Bad request protocol Packet
- Maximum retries exceeded on transmission
- RTCP Read too short
- AMI Connection
- Devices
- Zaptel
GUI
- Can't open GUI
- Can't login
- Graphs does not appear
- Can't open phpMyAdmin
- Can't send email
- Can't initiate web callback
- Can't setup callback
- Vouchers are not accepted
- Can't delete device/user - it has calls
- lib/transaction/simple.rb:46: warning: already initialized constant Messages
MySQL
Installation