Asterisk 13.6.0 + Incredible GUI 12.0.30 (Incredible PBX 13-12.3 for Raspberry Pi 2)
IPFire 2.19 (x86_64) - Core Update 110
In the following setup, VoIP does not work
Connection to outside world:
german Telekom VDSL100/40
connection done by Fritz!Box 7580 with firewall IPFire as "exposed host"
IPFire manages 192.168.80.0 (LAN) and 192.168.81.0 (WLAN) (+ several VPNs)
Asterisk is in LAN (192.168.80.12)
LAN and WLAN can "see" each other)
between FritzBox (DSL termination) and IPFire there is a transfer net (192.168.178.1 = FritzBox, 192.168.178.2 = ipfire)
Old setup (see list) did work!!
Starting Point was a working setup with the identical topology, but following changes:
The Asterisk server did not change.
Asterisk Server is an "Incredible PBX 12.0.70" (Asterisk 13.x on Debian 7, combined with modified FreePBX WebGUI)
DSL16+ to VDSL imples change from ISDN to VoIP
The old setup already had VoIP running (not with Telekom) with SIPGate, PersonalVoIP and IAXTEL.
After the changes the following happened:
"VoiP dos not work" means more specifically:
The SIP REGISTER works - alle DIDs/CIDs (number) are shown as "registered" both in Asterisk and in the Provider Webpages.
Calling in: the SIP INVITE packages do arrive at the Asterisk server (can be seen in "sngrep" on the Astraisk server). Asterisk answers, but this answer never reaches the VoIP provider.
(i.e.: phone rings, you pick up, Asterisks send OK, but the phone on the other side (the one calling in) still plays the ringing tone until timeout is reached.)
Calling out: Asterisk sends INVITE (to the correct address), never gets an answer, called phone does not ring, calling phone does not play ringing tone.
Well, I thought, it may be routing - so I tried to connect the smartphone from the internal network not to Asterisk, but directly to Telekom (through IPFire) and - works perfectly in both dirfections.
Oh well ... :'(
IPFire: blue and green are allowed to see each other (default: forward). IPFire Firewall options: "SIP application helper" enables.
Even setting NAT routes/rules directly pointing to Asterisk does not help.
Unfortunately, I do not have sngrep on IPFire - and as well not on FritzBox. :-(
The fact that the smartphones work directly (but using STUN, did not try without yet) point to an error in Astreisk.
The fact the the formerly working Non-Telekom SIP trunks stopped working after the changes but without any change in Asterisk point to an error in the IPFire/FritzBox combination, as nothing else changed.
Now I ran out of options.
Any ideas?
Kind regards
----------
some log files can be supplied, they are too long for this board
IPFire 2.19 (x86_64) - Core Update 110
In the following setup, VoIP does not work
Connection to outside world:
german Telekom VDSL100/40
connection done by Fritz!Box 7580 with firewall IPFire as "exposed host"
IPFire manages 192.168.80.0 (LAN) and 192.168.81.0 (WLAN) (+ several VPNs)
Asterisk is in LAN (192.168.80.12)
LAN and WLAN can "see" each other)
between FritzBox (DSL termination) and IPFire there is a transfer net (192.168.178.1 = FritzBox, 192.168.178.2 = ipfire)
Old setup (see list) did work!!
Starting Point was a working setup with the identical topology, but following changes:
- replaced old DSL16+ by new VDSL100
- therefore had to replace old Fritz!Box 3370 by new 7580
- replaced old IPCop (hardware died) by new IPFire
- replaced old ISDN POTS with new VoIP Trunk Telekom
- several VoIP trink dod NOT change, but stopped working (SIPGate, PersonalVoIP)
The Asterisk server did not change.
Asterisk Server is an "Incredible PBX 12.0.70" (Asterisk 13.x on Debian 7, combined with modified FreePBX WebGUI)
DSL16+ to VDSL imples change from ISDN to VoIP
The old setup already had VoIP running (not with Telekom) with SIPGate, PersonalVoIP and IAXTEL.
After the changes the following happened:
- VoIP (SIP) stopped working for SIPGate, IAXTEL and PersonalVoIIP using Asterisk - which worked before, no changes in config at all in Asterisk
- VoIP (SIP) does not work with Telecom
- VoIP/SIP DOES work from any Smartphone in BLUE and Softphone in GREEN, but only when connecting DIRECTLY through IPFire and FritzBox to the outside VoIP provider
- VoIP does not work when connecting using Asterisk, not even those trunks that DID work before the changes
"VoiP dos not work" means more specifically:
The SIP REGISTER works - alle DIDs/CIDs (number) are shown as "registered" both in Asterisk and in the Provider Webpages.
Calling in: the SIP INVITE packages do arrive at the Asterisk server (can be seen in "sngrep" on the Astraisk server). Asterisk answers, but this answer never reaches the VoIP provider.
(i.e.: phone rings, you pick up, Asterisks send OK, but the phone on the other side (the one calling in) still plays the ringing tone until timeout is reached.)
Calling out: Asterisk sends INVITE (to the correct address), never gets an answer, called phone does not ring, calling phone does not play ringing tone.
Well, I thought, it may be routing - so I tried to connect the smartphone from the internal network not to Asterisk, but directly to Telekom (through IPFire) and - works perfectly in both dirfections.
Oh well ... :'(
IPFire: blue and green are allowed to see each other (default: forward). IPFire Firewall options: "SIP application helper" enables.
Even setting NAT routes/rules directly pointing to Asterisk does not help.
Unfortunately, I do not have sngrep on IPFire - and as well not on FritzBox. :-(
The fact that the smartphones work directly (but using STUN, did not try without yet) point to an error in Astreisk.
The fact the the formerly working Non-Telekom SIP trunks stopped working after the changes but without any change in Asterisk point to an error in the IPFire/FritzBox combination, as nothing else changed.
Now I ran out of options.
Any ideas?
Kind regards
----------
some log files can be supplied, they are too long for this board