Re: [asterisk-users] Peer is UNREACHABLE
> I have a problem and I hope someone can help me... > I configured an Asterisk on a VM to serve more accounts and act as a proxy to > other SIP-providers. > > The first account running on my phone works without any problem. > A second account, running on the phone of my wife, is always UNREACHABLE. > I can just see in the log: > > [May 28 21:48:46] NOTICE[3646]: chan_sip.c:22933 sip_poke_noanswer: Peer > '004935111' is now UNREACHABLE! Last qualify: 0 > > In the CLI I can see: > > Name/username HostDyn Nat ACL Port Status > 004935111/0049351 192.168.200.11 D 5060 UNREACHABLE > 004935122/0049351 192.168.200.10 D 5060 OK (17 ms) > 004935133 (Unspecified)D 5060 UNKNOWN > 1234 (Unspecified)D 5060 UNKNOWN > messagenet/1234567890 212.97.59.765061 Unmonitored > pbxanika/004935172.16.34.132 5060 Unmonitored > pbxfax/0049351333 172.16.34.132 5060 Unmonitored > pbxluca/0049351222 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 configuration file, if they are needed. > What kind of phone are we talking about, both yours that works and your wife's that does not? Can you ping the unreachable phone and does it respond to a ping? Many phones will have a network test function built in to them to help you determine if the phone is properly connected to the network. Do you see anything in the asterisk logs or the logs of the phone itself (providing the phone puts logs somewhere) that indicate a failure to register or to resolve the ip address of the asterisk server?-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Peer is UNREACHABLE
Kevin Larsen schrieb: > What kind of phone are we talking about, both yours that works and your > wife's that does not? Right! > Can you ping the unreachable phone and does it respond to a ping? I can ping both phones from the VM > Many phones will have a network test function built in to them to help you > determine if the phone is properly connected to the network. Unfortunately not that... I tried with Twinkle from my PC, using the same account of my wife (configured IDENTICALLY to my account, just another username). It don't work... I presume, I configured something wrong in Asterisk... > Do you see anything in the asterisk logs or the logs of the phone itself > (providing the phone puts logs somewhere) that indicate a failure to > register or to resolve the ip address of the asterisk server? Unfortunately not... Just UNREACHABLE... Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Peer is UNREACHABLE
> > What kind of phone are we talking about, both yours that works and your > > wife's that does not? > > Right! > > > Can you ping the unreachable phone and does it respond to a ping? > > I can ping both phones from the VM > > > Many phones will have a network test function built in to them to help you > > determine if the phone is properly connected to the network. > > Unfortunately not that... > I tried with Twinkle from my PC, using the same account of my wife > (configured IDENTICALLY to my account, just another username). It don't > work... > I presume, I configured something wrong in Asterisk... > > > Do you see anything in the asterisk logs or the logs of the phone itself > > (providing the phone puts logs somewhere) that indicate a failure to > > register or to resolve the ip address of the asterisk server? > > Unfortunately not... Just UNREACHABLE... Can you post the Manufacturer and Model of your phones (both of them if they are different)? That will help us look up what diagnostics/log files there might be on the phones. Does the Twinkle software on the PC show any error messages? If you watch the CLI in asterisk, does anything go by in there regarding a failed registration? If I get one of my phones programmed with an incorrect username/secret, it will try to register with the server, but can't. Those failed registrations do show up in the CLI. Double check that you are not mistyping the credentials somewhere. If you do post the relevant parts of your config in here, you might want to obscure the secret.-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Peer is UNREACHABLE
Kevin Larsen schrieb: > Can you post the Manufacturer and Model of your phones (both of them if > they are different)? That will help us look up what diagnostics/log files > there might be on the phones. Of course! My phone is a Thomson ST2022 and my wife has a KE1020A > Does the Twinkle software on the PC show any error messages? Nope, just trying and then say "unable to connect"... > If you watch the CLI in asterisk, does anything go by in there regarding a > failed registration? If I get one of my phones programmed with an > incorrect username/secret, it will try to register with the server, but > can't. Those failed registrations do show up in the CLI. That's very strange... I expected these errors, but in the console I can't see anything... SOMETIMES, but just sometimes, if the phone of my wife tries to connect, I see something like "connecting from 192.168.200.11" (I can't find the error message anymore), and then: [May 28 21:46:27] NOTICE[3592] chan_sip.c: Peer '0049351222' is now UNREACHABLE! Last qualify: 0 > Double check that you are not mistyping the credentials somewhere. If you > do post the relevant parts of your config in here, you might want to > obscure the secret. Which part of the configuration do you need? Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Peer is UNREACHABLE
I'd start by turning on sip debugging in asterisk >sip set debug ip [your_phone_ip] and use tcpdump or wireshark to see what the OS sees tcpdump host [your_phone_ip] and udp port 5060 On 15-05-28 03:58 PM, Luca Bertoncello wrote: Hi list! I have a problem and I hope someone can help me... I configured an Asterisk on a VM to serve more accounts and act as a proxy to other SIP-providers. The first account running on my phone works without any problem. A second account, running on the phone of my wife, is always UNREACHABLE. I can just see in the log: [May 28 21:48:46] NOTICE[3646]: chan_sip.c:22933 sip_poke_noanswer: Peer '004935111' is now UNREACHABLE! Last qualify: 0 In the CLI I can see: Name/username HostDyn Nat ACL Port Status 004935111/0049351 192.168.200.11 D 5060 UNREACHABLE 004935122/0049351 192.168.200.10 D 5060 OK (17 ms) 004935133 (Unspecified)D 5060 UNKNOWN 1234 (Unspecified)D 5060 UNKNOWN messagenet/1234567890 212.97.59.765061 Unmonitored pbxanika/004935172.16.34.132 5060 Unmonitored pbxfax/0049351333 172.16.34.132 5060 Unmonitored pbxluca/0049351222 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 configuration file, if they are needed. Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Peer is UNREACHABLE
Darryl Moore schrieb: > I'd start by turning on sip debugging in asterisk > >sip set debug ip [your_phone_ip] Really destroying SIP dialog '490d1996593c8e11217828b71aae5c4d@172.16.34.133' Method: OPTIONS Reliably Transmitting (no NAT) to 192.168.200.11:5060: OPTIONS sip:0049351222@192.168.200.11:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.34.133:5060;branch=z9hG4bK13db26f5;rport Max-Forwards: 70 From: "asterisk" ;tag=as1215345d To: Contact: Call-ID: 78f3a0d0145f3dfa630a5e7c506142d6@172.16.34.133 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.5-0ubuntu1.4 Date: Thu, 28 May 2015 20:39:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 repeated in loop... Help that? 192.168.200.11 is the IP of the phone of my wife, and 172.16.34.133 the IP of the Asterisk server. Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Peer is UNREACHABLE
> Darryl Moore schrieb: > > > I'd start by turning on sip debugging in asterisk > > >sip set debug ip [your_phone_ip] > > Really destroying SIP dialog '490d1996593c8e11217828b71aae5c4d@172. > 16.34.133' Method: OPTIONS > Reliably Transmitting (no NAT) to 192.168.200.11:5060: > OPTIONS sip:0049351222@192.168.200.11:5060 SIP/2.0 > Via: SIP/2.0/UDP 172.16.34.133:5060;branch=z9hG4bK13db26f5;rport > Max-Forwards: 70 > From: "asterisk" ;tag=as1215345d > To: > Contact: > Call-ID: 78f3a0d0145f3dfa630a5e7c506142d6@172.16.34.133 > CSeq: 102 OPTIONS > User-Agent: Asterisk PBX 1.6.2.5-0ubuntu1.4 > Date: Thu, 28 May 2015 20:39:02 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO > Supported: replaces, timer > Content-Length: 0 > > repeated in loop... > Help that? > > 192.168.200.11 is the IP of the phone of my wife, and 172.16.34.133 > the IP of the Asterisk server. > 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.-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Peer is UNREACHABLE
Kevin Larsen 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, not connected to my Asterisk... If I connect it to the other VM with AsteriskNOW I can call my Twinkle, but NOT my phone connected on my Asterisk, using the "proxy". I can see that in the log: [May 28 22:49:51] WARNING[4135]: chan_sip.c:12800 check_auth: username mismatch, have <1234>, digest has [May 28 22:49:51] NOTICE[4135]: chan_sip.c:20083 handle_request_invite: Failed to authenticate device "Test1" ;tag=as6dd12e05 Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Peer is UNREACHABLE
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? On 15-05-28 04:51 PM, Luca Bertoncello wrote: Kevin Larsen 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, not connected to my Asterisk... If I connect it to the other VM with AsteriskNOW I can call my Twinkle, but NOT my phone connected on my Asterisk, using the "proxy". I can see that in the log: [May 28 22:49:51] WARNING[4135]: chan_sip.c:12800 check_auth: username mismatch, have <1234>, digest has [May 28 22:49:51] NOTICE[4135]: chan_sip.c:20083 handle_request_invite: Failed to authenticate device "Test1" ;tag=as6dd12e05 Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Peer is UNREACHABLE
> No, I'm not sure. > And no, I can't make any call, right now... At least, not connected to my > Asterisk... > If I connect it to the other VM with AsteriskNOW I can call my Twinkle, but > NOT my phone connected on my Asterisk, using the "proxy". > I can see that in the log: > > [May 28 22:49:51] WARNING[4135]: chan_sip.c:12800 check_auth: username > mismatch, have <1234>, digest has > [May 28 22:49:51] NOTICE[4135]: chan_sip.c:20083 handle_request_invite: > Failed to authenticate device "Test1" ;tag=as6dd12e05 > I know from your previous email that you are new to Asterisk. Have you created a dialplan that would allow you to call from one extension to another without going through your phone company? That is to say, call from your phone through Asterisk to your wife's phone? You have two parts that you need to have in place for the basics to work. You need your sip.conf in order to tell asterisk what devices and phone trunks you have and you need extensions.conf to tell Asterisk how to route calls. Since you are new to this, you can start by getting the two phones to both register (sounds like one of them is and one probably is not). Then you get to where you can dial from one phone to the other and vice versa. From there you can add in the telephone company lines and the ability to dial in and out to the world. I am still curious why you have both an Asterisk setup and an AsteriskNow setup? Is that just to play around with? At the end of the day you should just need one or the other.-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Peer is UNREACHABLE
Darryl Moore 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 => 004935:MYSECRET@pbxluca/004935
register => 0049351222:MYSECRET@pbxfax/0049351222
register => 0049351333:MYSECRET@pbxanika/0049351333
register => 44:MYSECRET@messagenet/44
[pbxluca]
type=peer
defaultuser=004935
secret= MYSECRET
dtmfmode=rfc2833
host=172.16.34.132
context=luca_incoming
outboundproxy=172.16.34.132
port=5060
fromuser=004935
fromdomain=172.16.34.132
usereqphone=yes
canreinvite=no
insecure=invite
[pbxfax]
type=peer
defaultuser=0049351222
secret= MYSECRET
dtmfmode=rfc2833
host=172.16.34.132
context=fax_incoming
outboundproxy=172.16.34.132
port=5060
fromuser=0049351222
fromdomain=172.16.34.132
usereqphone=yes
canreinvite=no
insecure=invite
[pbxanika]
type=peer
defaultuser=0049351333
secret= MYSECRET
dtmfmode=rfc2833
host=172.16.34.132
context=anika_incoming
outboundproxy=172.16.34.132
port=5060
fromuser=0049351333
fromdomain=172.16.34.132
usereqphone=yes
canreinvite=no
insecure=invite
[messagenet]
type=peer
defaultuser=44
secret=MYSECRET
dtmfmode=rfc2833
host=sip.messagenet.it
context=messagenet_incoming
outboundproxy=sip.messagenet.it
port=5061
fromuser=44
fromdomain=sip.messagenet.it
usereqphone=yes
canreinvite=no
insecure=invite
Here my extensions.conf:
[stdexten]
include => luca_incoming
include => fax_incoming
include => anika_incoming
include => messagenet_incoming
[luca_incoming]
exten => _004935,1,Verbose(2,Call for Luca)
exten => _004935,n,Dial(SIP/004935)
exten => _004935,n,Hangup
[fax_incoming]
exten => _0049351222,1,Verbose(2,Call for FAX)
exten => _0049351222,n,Dial(SIP/0049351222)
exten => _0049351222,n,Hangup
[anika_incoming]
exten => _0049351333,1,Verbose(2,Call for Anika)
exten => _0049351333,n,Dial(SIP/0049351333)
exten => _0049351333,n,Hangup
[messagenet_incoming]
exten => _44,1,Verbose(2,Call from Messagenet)
exten => _44,n,Dial(SIP/004935)
exten => _44,n,Hangup
[myproxy]
exten => _X.,1,Verbose(2,Call from ${CALLERID(num)} to ${EXTEN})
exten => _X.,n,GotoIf($["${CALLERID(num)}" = "004935"]?dialluca)
exten => _X.,n,GotoIf($["${CALLERID(num)}" = "0049351222"]?dialfax)
exten => _X.,n,GotoIf($["${CALLERID(num)}" = "0049351333"]?dialanika)
exten => _X.,n,Dial(SIP/pbxluca/${EXTEN},30,r)
exten => _X.,n,Hangup
exten => _X.,n(dialluca),Verbose(2,Outgoing using pbxluca)
exten => _X.,n(dialluca),Dial(SIP/pbxluca/${EXTEN},30,r)
exten => _X.,n,Hangup
exten => _X.,n(dialfax),Verbose(2,Outgoing using pbxfax)
exten => _X.,n(dialfax),Dial(SIP/pbxfax/${EXTEN},30,r)
exten => _X.,n,Hangup
exten => _X.,n(dialanika),Verbose(2,Outgoing using pbxanika)
exten => _X.,n(dialanika),Dial(SIP/pbxanika/${EXTEN},30,r)
exten => _X.,n,Hangup
And here my users.conf:
[004935]
fullname = luca
secret = MYSECRET
dahdichan = 1
hassip = yes
hasiax = no
hash323 = no
hasmanager = no
callwaiting = no
context = myproxy
host = dynamic
dtmfmode=rfc2833
canreinvite=no
sendrpid=pai
type=friend
nat=force_rport,comedia
qualify=yes
qualifyfreq=60
transport=Auto
avpf=no
force_avp=no
icesupport=no
encryption=no
callgroup=
pickupgroup=
dial=SIP/004935
[0049351222]
fullname = fax
secret = MYSECRET
dahdichan = 1
hassip = yes
hasiax = no
hash323 = no
hasmanager = no
callwaiting = no
context = myproxy
host = dynamic
dtmfmode=rfc2833
canreinvite=no
sendrpid=pai
type=friend
nat=force_rport,comedia
qualify=yes
qualifyfreq=60
transport=Auto
avpf=no
force_avp=no
icesupport=no
encryption=no
callgroup=
pickupgroup=
dial=SIP/0049351222
[0049351333]
fullname = anika
secret = MYSECRET
dahdichan = 1
hassip = yes
hasiax = no
hash323 = no
hasmanager = no
callwaiting = no
context = myproxy
host = dynamic
dtmfmode=rfc2833
canreinvite=no
sendrpid=pai
type=friend
nat=force_rport,comedia
qualify=yes
qualifyfreq=60
transport=Auto
avpf=no
force_avp=no
icesupport=no
encryption=no
callgroup=
pickupgroup=
dial=SIP/0049351333
Now I see this: if I call my phone (004935) from Twinkle it works.
If I call it from the phone of my wife, logged in on the same AsteriskNOW of
Twinkle and able to speak with Twinkle, it does NOT work and I see that in the
Log of my Asterisk:
== Using SIP RTP CoS mark 5
[May 28 23:05:59] WARNING[4135]: chan_sip.c:12800 check_auth: username
mismatch, have <1234>, digest has
[May 28 23:05:59] NOTICE[4135]: chan_sip.c:20083 handle_request_invite: Failed
to authenticate device "Test1" ;tag=as7855ffe5
(the phone of my wife is now logged in on AsteriskNOW with the user "1234" and
try
to call my phone with the same number I use from Twinkle, which works).
Very puzzled...
Thanks
Luca Bertoncello
(lucab...@lucabert.de)
--
__
Re: [asterisk-users] Peer is UNREACHABLE
Kevin Larsen schrieb: > I am still curious why you have both an Asterisk setup and an AsteriskNow > setup? Is that just to play around with? At the end of the day you should > just need one or the other. Just why I need a second SIP-provider to check if all works, when Deutsche Telekom activate the new line... So I installed AsteriskNOW on a VM and configured it to serve a couple of number. Then I installed Asterisk on a second VM and configured it to connect to AsteriskNOW (later will be Telekom) and Messagenet. Dialplan and the other configuration were already sent... Thanks Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Peer is UNREACHABLE
I think your phone may be trying to register with the username '1234',
while your sip configuration is expecting 'luca'. Can you try changing
your phone registration credentials to use 'luca'? Can you give us a sip
transcript when you try to place a call from it?
On 15-05-28 05:09 PM, Luca Bertoncello wrote:
Darryl Moore 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 => 004935:MYSECRET@pbxluca/004935
register => 0049351222:MYSECRET@pbxfax/0049351222
register => 0049351333:MYSECRET@pbxanika/0049351333
register => 44:MYSECRET@messagenet/44
[pbxluca]
type=peer
defaultuser=004935
secret= MYSECRET
dtmfmode=rfc2833
host=172.16.34.132
context=luca_incoming
outboundproxy=172.16.34.132
port=5060
fromuser=004935
fromdomain=172.16.34.132
usereqphone=yes
canreinvite=no
insecure=invite
[pbxfax]
type=peer
defaultuser=0049351222
secret= MYSECRET
dtmfmode=rfc2833
host=172.16.34.132
context=fax_incoming
outboundproxy=172.16.34.132
port=5060
fromuser=0049351222
fromdomain=172.16.34.132
usereqphone=yes
canreinvite=no
insecure=invite
[pbxanika]
type=peer
defaultuser=0049351333
secret= MYSECRET
dtmfmode=rfc2833
host=172.16.34.132
context=anika_incoming
outboundproxy=172.16.34.132
port=5060
fromuser=0049351333
fromdomain=172.16.34.132
usereqphone=yes
canreinvite=no
insecure=invite
[messagenet]
type=peer
defaultuser=44
secret=MYSECRET
dtmfmode=rfc2833
host=sip.messagenet.it
context=messagenet_incoming
outboundproxy=sip.messagenet.it
port=5061
fromuser=44
fromdomain=sip.messagenet.it
usereqphone=yes
canreinvite=no
insecure=invite
Here my extensions.conf:
[stdexten]
include => luca_incoming
include => fax_incoming
include => anika_incoming
include => messagenet_incoming
[luca_incoming]
exten => _004935,1,Verbose(2,Call for Luca)
exten => _004935,n,Dial(SIP/004935)
exten => _004935,n,Hangup
[fax_incoming]
exten => _0049351222,1,Verbose(2,Call for FAX)
exten => _0049351222,n,Dial(SIP/0049351222)
exten => _0049351222,n,Hangup
[anika_incoming]
exten => _0049351333,1,Verbose(2,Call for Anika)
exten => _0049351333,n,Dial(SIP/0049351333)
exten => _0049351333,n,Hangup
[messagenet_incoming]
exten => _44,1,Verbose(2,Call from Messagenet)
exten => _44,n,Dial(SIP/004935)
exten => _44,n,Hangup
[myproxy]
exten => _X.,1,Verbose(2,Call from ${CALLERID(num)} to ${EXTEN})
exten => _X.,n,GotoIf($["${CALLERID(num)}" = "004935"]?dialluca)
exten => _X.,n,GotoIf($["${CALLERID(num)}" = "0049351222"]?dialfax)
exten => _X.,n,GotoIf($["${CALLERID(num)}" = "0049351333"]?dialanika)
exten => _X.,n,Dial(SIP/pbxluca/${EXTEN},30,r)
exten => _X.,n,Hangup
exten => _X.,n(dialluca),Verbose(2,Outgoing using pbxluca)
exten => _X.,n(dialluca),Dial(SIP/pbxluca/${EXTEN},30,r)
exten => _X.,n,Hangup
exten => _X.,n(dialfax),Verbose(2,Outgoing using pbxfax)
exten => _X.,n(dialfax),Dial(SIP/pbxfax/${EXTEN},30,r)
exten => _X.,n,Hangup
exten => _X.,n(dialanika),Verbose(2,Outgoing using pbxanika)
exten => _X.,n(dialanika),Dial(SIP/pbxanika/${EXTEN},30,r)
exten => _X.,n,Hangup
And here my users.conf:
[004935]
fullname = luca
secret = MYSECRET
dahdichan = 1
hassip = yes
hasiax = no
hash323 = no
hasmanager = no
callwaiting = no
context = myproxy
host = dynamic
dtmfmode=rfc2833
canreinvite=no
sendrpid=pai
type=friend
nat=force_rport,comedia
qualify=yes
qualifyfreq=60
transport=Auto
avpf=no
force_avp=no
icesupport=no
encryption=no
callgroup=
pickupgroup=
dial=SIP/004935
[0049351222]
fullname = fax
secret = MYSECRET
dahdichan = 1
hassip = yes
hasiax = no
hash323 = no
hasmanager = no
callwaiting = no
context = myproxy
host = dynamic
dtmfmode=rfc2833
canreinvite=no
sendrpid=pai
type=friend
nat=force_rport,comedia
qualify=yes
qualifyfreq=60
transport=Auto
avpf=no
force_avp=no
icesupport=no
encryption=no
callgroup=
pickupgroup=
dial=SIP/0049351222
[0049351333]
fullname = anika
secret = MYSECRET
dahdichan = 1
hassip = yes
hasiax = no
hash323 = no
hasmanager = no
callwaiting = no
context = myproxy
host = dynamic
dtmfmode=rfc2833
canreinvite=no
sendrpid=pai
type=friend
nat=force_rport,comedia
qualify=yes
qualifyfreq=60
transport=Auto
avpf=no
force_avp=no
icesupport=no
encryption=no
callgroup=
pickupgroup=
dial=SIP/0049351333
Now I see this: if I call my phone (004935) from Twinkle it works.
If I call it from the phone of my wife, logged in on the same AsteriskNOW of
Twinkle and able to speak with Twinkle, it does NOT work and I see that in the
Log of my Asterisk:
== Using SIP RTP CoS mark 5
[May 28 23:05:59] WARNING[4135]: chan_sip.c:12800 check_auth: username mismatch, have
<1234>, digest has
[May 28 23:05:59] NOTICE[4135]: chan_sip.c:20083 handle_request_invite: Failed to a
Re: [asterisk-users] Peer is UNREACHABLE
Darryl Moore schrieb: > I think your phone may be trying to register with the username '1234', > while your sip configuration is expecting 'luca'. Can you try changing > your phone registration credentials to use 'luca'? Can you give us a sip > transcript when you try to place a call from it? Well, right now this phone USES the username 1234, on the AsteriskNOW (the "later Telekom"). I really don't know why it tries to authenticate to my "own Asterisk"... What I see right now, if I try to connect the phone of my wife to "my own Asterisk": -- Registered SIP '0049351222' at 192.168.200.11 port 5060 [May 28 23:46:01] NOTICE[1350]: chan_sip.c:22933 sip_poke_noanswer: Peer '0049351222' is now UNREACHABLE! Last qualify: 0 But, as I said, right now the phone is connected to the AsteriskNOW... Well, now I must sleep... Hope someone can suggest me something that I can try tomorrow. Thanks a lot Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Peer is UNREACHABLE
Darryl Moore schrieb:
> I think your phone may be trying to register with the username '1234',
> while your sip configuration is expecting 'luca'. Can you try changing
> your phone registration credentials to use 'luca'? Can you give us a sip
> transcript when you try to place a call from it?
Well, another information (then I **MUST** go sleep...):
I tried to use my mobile phone logging to my "own Asterisk" with the login
data of my wife's telefon.
Now this user is REACHABLE... So I think, it was a problem on her phone...
I can't call and receive calls. I think, that it's a problem of my Dialplan.
If I try to call the mobile phone from AsteriskNOW (later: "the world"), I
see that in Asterisk's log ("my own Asterisk"):
== Using SIP RTP CoS mark 5
[May 29 00:07:49] NOTICE[1106]: chan_sip.c:20163 handle_request_invite: Call
from '004935' to extension '0049351222' rejected because
extension not found.
That's very strange, since I call from Twinkle and it has the number "1234"...
If I call my mobile phone using my VoIP-phone (connected on the same "my own
Asterisk") I get that:
== Using SIP RTP CoS mark 5
== Call from 004935 to 0049351222
== Outgoing using pbxluca
== Using SIP RTP CoS mark 5
== Using SIP RTP CoS mark 5
[May 29 00:09:25] WARNING[1106]: chan_sip.c:12800 check_auth: username
mismatch, have <004935>, digest has <0049351222> [May 29
00:09:25] NOTICE[1106]: chan_sip.c:20083 handle_request_invite: Failed to
authenticate device "004935"
;tag=as058adbf2 == Everyone is
busy/congested at this time (1:0/1/0) == Spawn extension (myproxy,
0049351222, 9) exited non-zero on 'SIP/004935-0004'
Maybe this is the same problem, since I didn't configured my own Asterisk to
manage "internal calls" (since I don't need to call my wife on VoIP... :D)
And, last but not least, if I try to call from my mobile phone Twinkle I get
this:
== Using SIP RTP CoS mark 5
== Call from 0049351222 to 1234
== Outgoing using pbxanika
== Using SIP RTP CoS mark 5
== Everyone is busy/congested at this time (1:0/1/0)
== Spawn extension (myproxy, 1234, 15) exited non-zero on
'SIP/0049351222-0006'
And if I try to call my VoIP-phone I get that:
== Using SIP RTP CoS mark 5
== Call from 0049351222 to 004935
== Outgoing using pbxanika
== Using SIP RTP CoS mark 5
== Using SIP RTP CoS mark 5
[May 29 00:12:02] WARNING[1106]: chan_sip.c:12800 check_auth: username
mismatch, have <0049351222>, digest has <004935>
[May 29 00:12:02] NOTICE[1106]: chan_sip.c:20083 handle_request_invite: Failed
to authenticate device "0049351222"
;tag=as193c26b0
== Everyone is busy/congested at this time (1:0/1/0)
== Spawn extension (myproxy, 004935, 15) exited non-zero on
'SIP/0049351222-000a'
Maybe can these information help someone helping me?
Thanks a lot!
Luca Bertoncello
(lucab...@lucabert.de)
--
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Re: [asterisk-users] Peer is UNREACHABLE
Maybe shut off qualify for the peer? I think I tried twinkle a few years ago and it didna (yes didna) like the qualify packet. the sip options qualify packet is only needed to keep the UDP state tables in a firewall if the peer is remote -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Peer is UNREACHABLE
Zitat von Adrian Serafini : Maybe shut off qualify for the peer? I think I tried twinkle a few years ago and it didna (yes didna) like the qualify packet. the sip options qualify packet is only needed to keep the UDP state tables in a firewall if the peer is remote Well, the same happens with my wife's phone... I'll try later again... Regards Luca Bertoncello (lucab...@lucabert.de) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] peer is UNREACHABLE when using XLITE
Hans-Henrik Andresen wrote: > Hi, > > I have 3 friends trying to connect to my Asterisk using x-lite, all > of them are using 3 dif. adsl-provider. > > For each of them I got this in sip.conf: > > disallow=all > allow=ulaw > allow=alaw > allow=ilbc > allow=g729 > allow=g723.1 > > > [seholm] > type=friend > secret=** > auth=md5 > nat=yes > host=dynamic > reinvite=no > canreinvite=no > qualify=1000 > dtmfmode=inband > callerid=Svend Erik Holm <60> > context=sip > > They can all connect, but in asterisk I got this > > *CLI> Mar 7 09:20:26 NOTICE[278546]: chan_sip.c:5846 > sip_poke_noanswer: Peer 'henrikoglone' is now UNREACHABLE! > > And sip show peers show this > seholm 83.88.89.122(D) 255.255.255.255 5060 UNREACHABLE > > They can make calls TO me, but of cause the pickup wont be sent to > them, so asterisk shut down the channel after 5 sec. or so. > > If they are using sjphone it works, test with gs. analog adaptor also > works. Most likely NAT sessions are too short. In another words ports at which devices operated are closed to soon by firewall/router/pc, hence * can not "find" devices. > > If I use a x-lite on same lan as asterisk, and if they make a VPN to > my asterisk it works. > > (We had tried to use stun-server as well) > > Any clue ? > > > /Hans-Henrik Andresen > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] peer is UNREACHABLE when using XLITE
Senad Jordanovic wrote: Hans-Henrik Andresen wrote: Hi, I have 3 friends trying to connect to my Asterisk using x-lite, all of them are using 3 dif. adsl-provider. For each of them I got this in sip.conf: disallow=all allow=ulaw allow=alaw allow=ilbc allow=g729 allow=g723.1 [seholm] type=friend secret=** auth=md5 nat=yes host=dynamic reinvite=no canreinvite=no qualify=1000 If the client turns UNREACHABLE, you might want to change the qualify= setting to qualify=yes, that defaults to two seconds, instead of one second that you have here. http://www.voip-info.org/tiki-index.php?page=Asterisk%20sip%20qualify Check with 'sip show peers' how long turn-around-time you have to the clients after that. /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] peer is UNREACHABLE when using XLITE
RTP is default port 8000 in X-lite. Try Forward port 8000 UDP to *, and see how that goes. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hans-Henrik Andresen Sent: 07 March 2004 08:28 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] peer is UNREACHABLE when using XLITE Hi, I have 3 friends trying to connect to my Asterisk using x-lite, all of them are using 3 dif. adsl-provider. For each of them I got this in sip.conf: disallow=all allow=ulaw allow=alaw allow=ilbc allow=g729 allow=g723.1 [seholm] type=friend secret=** auth=md5 nat=yes host=dynamic reinvite=no canreinvite=no qualify=1000 dtmfmode=inband callerid=Svend Erik Holm <60> context=sip They can all connect, but in asterisk I got this *CLI> Mar 7 09:20:26 NOTICE[278546]: chan_sip.c:5846 sip_poke_noanswer: Peer 'henrikoglone' is now UNREACHABLE! And sip show peers show this seholm 83.88.89.122(D) 255.255.255.255 5060 UNREACHABLE They can make calls TO me, but of cause the pickup wont be sent to them, so asterisk shut down the channel after 5 sec. or so. If they are using sjphone it works, test with gs. analog adaptor also works. If I use a x-lite on same lan as asterisk, and if they make a VPN to my asterisk it works. (We had tried to use stun-server as well) Any clue ? /Hans-Henrik Andresen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
