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 '0049351111111' is now UNREACHABLE! Last qualify: 0 In the CLI I can see: Name/username Host Dyn Nat ACL Port Status 0049351111111/00493511111 192.168.200.11 D 5060 UNREACHABLE 0049351222222/00493512222 192.168.200.10 D 5060 OK (17 ms) 0049351333333 (Unspecified) D 5060 UNKNOWN 1234 (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 configuration file, if they are needed. Thanks Luca Bertoncello (lucabert at lucabert.de)
> 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 aproxy 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 alwaysUNREACHABLE.> I can just see in the log: > > [May 28 21:48:46] NOTICE[3646]: chan_sip.c:22933 sip_poke_noanswer: Peer > '0049351111111' is now UNREACHABLE! Last qualify: 0 > > In the CLI I can see: > > Name/username Host Dyn Nat ACL Port Status > 0049351111111/00493511111 192.168.200.11 D 5060 UNREACHABLE > 0049351222222/00493512222 192.168.200.10 D 5060 OK (17ms)> 0049351333333 (Unspecified) D 5060 UNKNOWN> 1234 (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, 0offline]> > 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? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150528/b53a6e04/attachment.html>
Kevin Larsen <kevin.larsen at pioneerballoon.com> 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 (lucabert at lucabert.de)
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 > '0049351111111' is now UNREACHABLE! Last qualify: 0 > > In the CLI I can see: > > Name/username Host Dyn Nat ACL Port Status > 0049351111111/00493511111 192.168.200.11 D 5060 UNREACHABLE > 0049351222222/00493512222 192.168.200.10 D 5060 OK (17 ms) > 0049351333333 (Unspecified) D 5060 UNKNOWN > 1234 (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 configuration file, if they are needed. > > Thanks > Luca Bertoncello > (lucabert at lucabert.de) >
Darryl Moore <darryl at moores.ca> schrieb:> I'd start by turning on sip debugging in asterisk > >sip set debug ip [your_phone_ip]Really destroying SIP dialog '490d1996593c8e11217828b71aae5c4d at 172.16.34.133' Method: OPTIONS Reliably Transmitting (no NAT) to 192.168.200.11:5060: OPTIONS sip:00493512222222 at 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" <sip:asterisk at 172.16.34.133>;tag=as1215345d To: <sip:00493512222222 at 192.168.200.11:5060> Contact: <sip:asterisk at 172.16.34.133> Call-ID: 78f3a0d0145f3dfa630a5e7c506142d6 at 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 (lucabert at lucabert.de)