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Trixbox tűzfal mögött
SIP with firewall/NAT using Asterisk
Tűzfalbeállítások Asterisk-hez
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# SIP on UDP port 5060. Other SIP servers may need TCP port 5060 as well
iptables -A INPUT -p udp -m udp --dport 5060 -j ACCEPT
# IAX2- the IAX protocol
iptables -A INPUT -p udp -m udp --dport 4569 -j ACCEPT
# IAX - most have switched to IAX v2, or ought to
iptables -A INPUT -p udp -m udp --dport 5036 -j ACCEPT
# RTP - the media stream
iptables -A INPUT -p udp -m udp --dport 10000:20000 -j ACCEPT
# MGCP - if you use media gateway control protocol in your configuration
iptables -A INPUT -p udp -m udp --dport 2727 -j ACCEPT
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Mobile to Lan szabályt a következők szerint kell beállítani
Paraméter | Érték |
|---|---|
CID | * |
URL | Mellék@Asterisk IP |
Egyéb beállítási segédletek
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http://blogs.digium.com/2009/03/28/sip-security/
http://particio.com/~kovzol/kotta/doku.php?id=tegyuek_biztonsagossa_az_asterisket
VOIP szolgáltatók összehasonlítása
A különböző VOIP szolgáltatók ajánlatainak öszehasonlítását itt olvashatjuk.
A Betamax klónok árainak összehasonlítását itt olvashatjuk.
Trixbox hangolása Hyper-V környezetben
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http://www.smartvox.co.uk/astfaq_configbehindnat.htm
CallGroup PickGroup
http://www.freepbx.org/support/documentation/howtos/how-to-use-callgroups-and-pickgroups
Csengőhangok
http://forums.digium.com/viewtopic.php?p=152714
Asterisk Synology NAS modul
Voicemail
Segít abban, hogy ha nincs vonalbontás, akkor automatikusan befejeződhessen a hangposta:
voicemail.conf
maxsilence=10
silencethreshold=256
FAX beállítások SPA ATA készülékek esetén
http://wiki.freeswitch.org/wiki/SPA2102_T38_Howto
Firmware
Latest release apparently buggy (5.2.12). Get version 5.2.10 instead. Read that as T38 DOES NOT WORK on 5.2.12. Comment: I am using latest 5.2.13(004) which has no problem at all. Read the release note of 5.2.12 which fixed a few bugs 5.2.10. One of them I consider major.
Configure SPA2102
Provider and registration as per usual. watch the 0.030 bug on the RTP Packet size (login advanced)
- Voice -> SIP
- RTP Packet Size: 0.020
- Voice -> Line 1 [or 2]
| Preferred Codec | G711u | Second Preferred Codec | G711u |
| Third Preferred Codec | G711u | Use Pref Codec Only | yes |
| Silence Supp Enable | no | Silence Threshold | medium |
| G729a Enable | yes | Echo Canc Enable | no |
| G723 Enable | yes | Echo Canc Adapt Enable | no |
| G726-16 Enable | yes | Echo Supp Enable | no |
| G726-24 Enable | yes | FAX CED Detect Enable | yes |
| G726-32 Enable | yes | FAX CNG Detect Enable | yes |
| G726-40 Enable | yes | FAX Passthru Codec | G711u |
| DTMF Process INFO | yes | FAX Codec Symmetric | yes |
| DTMF Process AVT | Yes | FAX Passthru Method | ReINVITE |
| DTMF Tx Method | Auto | DTMF Tx Mode | Strict |
| DTMF Tx Strict Hold Off Time | 90 | FAX Process NSE | yes |
| Hook Flash Tx Method | None | FAX Disable ECAN | yes |
| Release Unused Codec | yes | FAX Enable T38 | yes |
| FAX T38 Redundancy | 1 | FAX Tone Detect Mode | caller or callee |
In addition to above, you will want to use the following settings:
- Source: cisco manual
On the Same Line Tab:
- Network Jitter Buffer: very high
- Jitter buffer adjustment: disable
- Call Waiting: no
- 3 Way Calling: no
- Echo Canceller: no
- Silence Suppression: no
- Preferred Codec: G.711
- Use Pref. codec Only: Yes
Without the above settings, outbound fax would work (in proxy media mode) some of the time, but T38 negotioation would fail sometimes and never go through. One example was HP's faxback service: 1-888-hpfaxme (1-888-473-2963).
Configure Fax Machine
You'll also need to set your Fax machine to a max speed of 14.4k and a min speed of 7200.
- Source: cisco manual
NAT-olt extension-ök leszakadnak konfiguráció módosítás után
A problémát az asterisk-nél ezen a hibajegyen tartják nyilván
Megoldás
Freepbx esetén
Settings->Asterisk SIP Settings->Other SIP Settings
rtcachefriends=yes
rtsavesysname=yes
rtupdate=yes
VoIP tarifák összehasonlítása
http://voipratetracker.com/compare_rates?ct=hungary&period=month&country_select=Update
2 freepbx/asterisk központ összekapcsolása
http://wiki.freepbx.org/pages/viewpage.action?pageId=4161588
Asterisk SNMP
http://deobfuscate.net/2013/09/05/configuring-snmp-v3-for-asterisk-on-freepbx/
FAX
Asterisk Jitter Buffer
https://sysadminman.net/blog/2013/improving-asterisk-call-quality-with-sip-jitter-buffers-5612
/etc/asterisk/sip_general_custom.conf
jbenable=yes
jbimpl=adaptive