search for: pbxluca

...lt:5] GotoIf("Local/0039015222222 at default-0000003c;2", "1?dialluca") in new stack -- Goto (default,0039015222222,13) -- Executing [0039015222222 at default:13] Verbose("Local/0039015222222 at default-0000003c;2", "2,Outgoing call for 0039015222222 using pbxluca") in new stack == Outgoing call for 0039015222222 using pbxluca -- Executing [0039015222222 at default:14] Dial("Local/0039015222222 at default-0000003c;2", "SIP/pbxluca/0039015222222,,RXx") in new stack == Using SIP RTP CoS mark 5 -- Called SIP/pbxluca/00390152...

...l work later, when Deutsche Telekom > > changes my ISDN to VoIP... > > Don't worry, Asterisk works very well with Deutsche Telekom and there > new ip-based connections ... That's a good news... Currenty I configured my sip.conf so: register => 00493511111111:MYSECRET at pbxluca/00493511111111 register => 00493512222222:MYSECRET at pbxfax/00493512222222 register => 00493513333333:MYSECRET at pbxanika/00493513333333 register => 4444444444:MYVERYSECRET at messagenet/4444444444 [pbxluca] type=peer defaultuser=00493511111111 secret= MYSECRET dtmfmode=rfc2833 host=172...

Darryl Moore <darryl at moores.ca> schrieb: > Ahh. Seen that before! That suggests to me that you don't have your > sip.conf records setup right. > > What's your sip.conf look like? Well, here what I wrote in my sip.conf: register => 00493511111111:MYSECRET at pbxluca/00493511111111 register => 00493512222222:MYSECRET at pbxfax/00493512222222 register => 00493513333333:MYSECRET at pbxanika/00493513333333 register => 4444444444:MYSECRET at messagenet/4444444444 [pbxluca] type=peer defaultuser=00493511111111 secret= MYSECRET dtmfmode=rfc2833 host=172.16....

...1234 (Unspecified) D 5060 UNKNOWN messagenet/4444444444 212.97.59.76 5061 Unmonitored pbxanika/00493513333333 172.16.34.132 5060 Unmonitored pbxfax/00493512222222 172.16.34.132 5060 Unmonitored pbxluca/00493511111111 172.16.34.132 5060 Unmonitored 8 sip peers [Monitored: 2 online, 2 offline Unmonitored: 4 online, 0 offline] If I call from my phone (00493511111111) my wife's phone (00493513333333), it works. If I call from my wife's phone (00493513333333) my phone (0...

...or these calls. Of course, but how can I test, if I am an "open relay"? > The calls are being dumped into your default context. It's not matching on > your gotoif statements, so finally it is trying to execute this: > Dial("SIP/192.168.20.120-0000002a", "SIP/pbxluca/000972592603325,,R") in > new stack > > Not sure what trunk pbxluca is, but if that is an outbound trunk, then > this is very bad. The only reason it would fail then is if they have the This is one of my outbound trunk... > outbound dial pattern wrong, which is a sure sign...

Kevin Larsen <kevin.larsen at pioneerballoon.com> schrieb: > Based on SIP packets coming in from IP addresses you don't recognize, > while you may not be hacked, you would seem to have people probing your I think, too, it's someone probing my IP... > system. One thing you can do at the firewall level is restrict inbound sip > communications to only those from your

...moores.ca> schrieb: > >> Ahh. Seen that before! That suggests to me that you don't have your >> sip.conf records setup right. >> >> What's your sip.conf look like? > Well, here what I wrote in my sip.conf: > > register => 00493511111111:MYSECRET at pbxluca/00493511111111 > register => 00493512222222:MYSECRET at pbxfax/00493512222222 > register => 00493513333333:MYSECRET at pbxanika/00493513333333 > register => 4444444444:MYSECRET at messagenet/4444444444 > > [pbxluca] > type=peer > defaultuser=00493511111111 > secret=...

...Call ID Format Hold Last Message Expiry Peer 192.168.200.10 0049351xxxxxxx 9eff88f7-c0a801 (alaw) No Rx: ACK 0049351xxxxxxx 217.0.27.53 03501xxxxxxx 453efbcb7a04f33 (alaw) No Tx: ACK pbxluca 2 active SIP dialogs Call from mobile phone (via VoIP registered in Asterisk): bpi*CLI> sip show channels Peer User/ANR Call ID Format Hold Last Message Expiry Peer 192.168.10.12 0049177xxxxxxx 11b86bd612b71ae (alaw) No Rx:...

...Format Hold > Last Message Expiry Peer > 192.168.200.10 0049351xxxxxxx 9eff88f7-c0a801 (alaw) No > Rx: ACK 0049351xxxxxxx > 217.0.27.53 03501xxxxxxx 453efbcb7a04f33 (alaw) No > Tx: ACK pbxluca > 2 active SIP dialogs > > Call from mobile phone (via VoIP registered in Asterisk): > > bpi*CLI> sip show channels > Peer User/ANR Call ID Format Hold > Last Message Expiry Peer > 192.168.10.12 0049177xxxxxxx 11b86b...

...are trying to use you to dial out to another number. You don't want to pay for these calls. The calls are being dumped into your default context. It's not matching on your gotoif statements, so finally it is trying to execute this: Dial("SIP/192.168.20.120-0000002a", "SIP/pbxluca/000972592603325,,R") in new stack Not sure what trunk pbxluca is, but if that is an outbound trunk, then this is very bad. The only reason it would fail then is if they have the outbound dial pattern wrong, which is a sure sign that you are open in the future to having someone make this k...

Am 13.06.2020 09:30, schrieb Luca Bertoncello: Hi again (again) I noticed right now another strange detail... I made a call using my mobile phone (connected to the Asterisk). The quality was top... Maybe is the problem in a codec used from our phones at homes? Could someone suggest me how to check the codec used by my mobile phone and the codec used by the phones at home? Thanks Luca

Kevin Larsen <kevin.larsen at pioneerballoon.com> schrieb: > The phone you gave your wife is really old. Are you sure it supports SIP > OPTIONS? Can you make a call in or out to it? If you can, it is more > likely that it just doesn't support that and you can't use a qualify > statement. No, I'm not sure. And no, I can't make any call, right now... At least,

...ourse, but how can I test, if I am an "open relay"? > > > The calls are being dumped into your default context. It's not matching on > > your gotoif statements, so finally it is trying to execute this: > > Dial("SIP/192.168.20.120-0000002a", "SIP/pbxluca/000972592603325,,R") in > > new stack > > > > Not sure what trunk pbxluca is, but if that is an outbound trunk, then > > this is very bad. The only reason it would fail then is if they have the > > This is one of my outbound trunk... > > > outbou...

...(Unspecified) D 5060 UNKNOWN messagenet/1234567890 212.97.59.76 5061 Unmonitored pbxanika/00493511111111 172.16.34.132 5060 Unmonitored pbxfax/00493513333333 172.16.34.132 5060 Unmonitored pbxluca/00493512222222 172.16.34.132 5060 Unmonitored 8 sip peers [Monitored: 1 online, 3 offline Unmonitored: 4 online, 0 offline] Asterisk connects to another Test-VM with AsteriskNOW and to the italian provider Messagenet. Can someone suggest me, what can I do? I can send the co...

...68.20.120-0000002a", "0?dialfax") in new stack -- Executing [000972592603325 at default:5] GotoIf("SIP/192.168.20.120-0000002a", "0?dialanika") in new stack -- Executing [000972592603325 at default:6] Dial("SIP/192.168.20.120-0000002a", "SIP/pbxluca/000972592603325,,R") in new stack [Jun 8 21:42:50] WARNING[18981]: app_dial.c:2345 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [000972592603325 at default:7] Hangup("...

Am 23.06.2020 16:22, schrieb Marek Greško: > It seems your problems lie in something other. Most probably it is not > mtu problem. All my suspections are contradicted. If it is true you > have inter vlan voice quality problems, it is definitely something > different. Formerly I assumed you were trying only LTE vs LAN using > internet. I'm not sure what you mean with the last

Would you mind repeating the test with canreinvite=no set for all you phones and mobile phones? What is your upload bitrate? Is it guaranteed? I would try also to test the PMTU: Try: ping -M do -s 2000 ${ip address of the sip server} You should receive icmp asking for lowering the packet size. The LTE phones could have lower MTU and thus overcome PMTU problem. Marek 2020-06-22 21:48

Hi list! Finally I got my Asterisk works with my two phones... It was a problem on my Firewall (for the phone of my wife) and on my Dialplan (for forwarding calls). Now all works as expected, at least in the simulation I did with AsteriskNOW. Hopefully it will work later, when Deutsche Telekom changes my ISDN to VoIP... Well, now I have some time to spend with "fooling"... My phone

...(Unspecified) D 5060 UNKNOWN > messagenet/1234567890 212.97.59.76 5061 Unmonitored > pbxanika/00493511111111 172.16.34.132 5060 Unmonitored > pbxfax/00493513333333 172.16.34.132 5060 Unmonitored > pbxluca/00493512222222 172.16.34.132 5060 Unmonitored > 8 sip peers [Monitored: 1 online, 3 offline Unmonitored: 4 online, 0 offline] > > Asterisk connects to another Test-VM with AsteriskNOW and to the italian > provider Messagenet. > > Can someone suggest me, what...

...(Unspecified) D 5060 UNKNOWN > messagenet/1234567890 212.97.59.76 5061 Unmonitored > pbxanika/00493511111111 172.16.34.132 5060 Unmonitored > pbxfax/00493513333333 172.16.34.132 5060 Unmonitored > pbxluca/00493512222222 172.16.34.132 5060 Unmonitored > 8 sip peers [Monitored: 1 online, 3 offline Unmonitored: 4 online, 0 offline] > > Asterisk connects to another Test-VM with AsteriskNOW and to the italian > provider Messagenet. > > Can someone suggest me, what...