Re: [asterisk-users] asterisk and maybe a freepbx question
OK, that tells me something, I will disable pjsit for now, learn about it and try again. On Sun, 09 Jan 2022 06:39:55 -0500, John Harragin wrote: > > [1 ] > [1.1 ] > You can also set up multiple physical or vlan(ed) interfaces and bind sip > to one and pjsip to the other - then you have to set up the appropriate > interface routing too for both inbound and outbound packets which takes a > good understanding of your network topology and the locations of your > respective devices. You might be able to do it with multiple addresses on > your interface too (although I haven't tried it). > > All of the packets have to be presented to the appropriate channel > otherwise get discarded. You can't set it up so if a packet is from a > device not registered with pjsip, it gets passed to chan_sip to try. > > For me, I had both channel types running on production machines while I > migrated to pjsip or when not being able to figure out how to set up some > property in pjsip that you had running in sip. Each time I've had to do > this, eventually I was able get it all running within pjsip. I also already > had multiple vlans configured for my servers (with voip exclusive to one). > > The short story is that it is easier to learn how to get things working > within pjsip than learning the tricky networking setup. > > > On Sun, Jan 9, 2022 at 2:49 AM Duncan Turnbull > wrote: > > > > > > > > > > > > On 9/01/2022, at 7:11 PM, John Covici wrote: > > > > > > On Sat, 08 Jan 2022 19:17:57 -0500, > > > Antony Stone wrote: > > >> > > >>> On Sunday 09 January 2022 at 00:50:27, John Covici wrote: > > >>> > > >>> Hi. I am using asterisk 18.3 and freepbx. > > >> > > >> Hm, which version of FreePBX uses Asterisk 18.3? > > >> > > >>> How can both sip and pjsip be listening at port 5060 at the same time > > >> > > >> They can't. > > >> > > >> One might be on TCP and the other on UDP, but you can't have them both > > >> listening on the same port with the same protocol. > > > > In freepbx you enable chan sip or pjsip or both and set what ports they use > > > > The choices are either in advanced settings or sip settings > > > > Disable pjsip and reset the chan_sip port to 5060 or use pjsip. With them > > both enabled sometimes odd things happen but it will still work. You will > > get lots of error messages though > > > > > > >> > > >>> for instance I get: > > >>> > > >>> [2022-01-08 17:08:59] SECURITY[244351] res_security_log.c: > > >>> > > SecurityEvent="FailedACL",EventTV="2022-01-08T17:08:59.957-0500",Severity=" > > >>> > > Error",Service="PJSIP",EventVersion="1",AccountID="anonymous",SessionID="20 > > >>> 25076022",LocalAddress="IPV4/UDP/166.84.7.53/5060 > > ",RemoteAddress="IPV4/UDP/ > > >>> 45.134.144.118/5823 > > ",ACLName="registrar_attempt_without_configured_aors" > > >> > > >> What makes you think chan_sip and pjsip are both listening on UDP 5060? > > >> > > >>> I would like pjsit not to listen,till I figure out how to configure > > >>> the thing, so my logs don't fill up with messages. > > >>> > > >>> Thanks in advance for any suggestions. > > >> > > >> As far as I recall using FreePBX, there is a selector for the SIP > > protocol to > > >> tell it whether you want it to use pjsip or chan_sip. I don't think it > > even > > >> supports using both at the same time, so simply make sure that is set > > to > > >> chan_sip and you should be fine. > > >> > > >> On the other hand, why do you need to learn "how to configure the > > thing" if > > >> you're using FreePBX? Part of the whole point is that it does the > > fiddly > > >> techie sutff in the background for you, and you just need to use the > > personnel- > > >> department-friendly web GUI. > > > > > > This is what I thought as well, I just generated one trunk using the > > > old chan_sip and expected nothing from pjsit, yet I get all kinds of > > > errors like > > > [2022-01-08 17:08:59] WARNING[487628] res_pjsip_registrar.c: Endpoint > > > 'anonymous' (45.134.144.118:5823) has no configured AORs > > > > > > so I am very confused as to why this is happening. > > > > > > -- > > > Your life is like a penny. You're going to lose it. The question is: > > > How do > > > you spend it? > > > > > > John Covici wb2una > > > cov...@ccs.covici.com > > > > > > -- > > > _ > > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > > > > > Check out the new Asterisk community forum at: > > https://community.asterisk.org/ > > > > > > New to Asterisk? Start here: > > > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > > > > > asterisk-users mailing list > > > To UNSUBSCRIBE or update options visit: > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > > _ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > > > Check out the new Asterisk
Re: [asterisk-users] asterisk and maybe a freepbx question
> On 9/01/2022, at 7:11 PM, John Covici wrote: > > On Sat, 08 Jan 2022 19:17:57 -0500, > Antony Stone wrote: >> >>> On Sunday 09 January 2022 at 00:50:27, John Covici wrote: >>> >>> Hi. I am using asterisk 18.3 and freepbx. >> >> Hm, which version of FreePBX uses Asterisk 18.3? >> >>> How can both sip and pjsip be listening at port 5060 at the same time >> >> They can't. >> >> One might be on TCP and the other on UDP, but you can't have them both >> listening on the same port with the same protocol. In freepbx you enable chan sip or pjsip or both and set what ports they use The choices are either in advanced settings or sip settings Disable pjsip and reset the chan_sip port to 5060 or use pjsip. With them both enabled sometimes odd things happen but it will still work. You will get lots of error messages though >> >>> for instance I get: >>> >>> [2022-01-08 17:08:59] SECURITY[244351] res_security_log.c: >>> SecurityEvent="FailedACL",EventTV="2022-01-08T17:08:59.957-0500",Severity=" >>> Error",Service="PJSIP",EventVersion="1",AccountID="anonymous",SessionID="20 >>> 25076022",LocalAddress="IPV4/UDP/166.84.7.53/5060",RemoteAddress="IPV4/UDP/ >>> 45.134.144.118/5823",ACLName="registrar_attempt_without_configured_aors" >> >> What makes you think chan_sip and pjsip are both listening on UDP 5060? >> >>> I would like pjsit not to listen,till I figure out how to configure >>> the thing, so my logs don't fill up with messages. >>> >>> Thanks in advance for any suggestions. >> >> As far as I recall using FreePBX, there is a selector for the SIP protocol >> to >> tell it whether you want it to use pjsip or chan_sip. I don't think it even >> supports using both at the same time, so simply make sure that is set to >> chan_sip and you should be fine. >> >> On the other hand, why do you need to learn "how to configure the thing" if >> you're using FreePBX? Part of the whole point is that it does the fiddly >> techie sutff in the background for you, and you just need to use the >> personnel- >> department-friendly web GUI. > > This is what I thought as well, I just generated one trunk using the > old chan_sip and expected nothing from pjsit, yet I get all kinds of > errors like > [2022-01-08 17:08:59] WARNING[487628] res_pjsip_registrar.c: Endpoint > 'anonymous' (45.134.144.118:5823) has no configured AORs > > so I am very confused as to why this is happening. > > -- > Your life is like a penny. You're going to lose it. The question is: > How do > you spend it? > > John Covici wb2una > cov...@ccs.covici.com > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and maybe a freepbx question
On Sat, 08 Jan 2022 19:17:57 -0500, Antony Stone wrote: > > On Sunday 09 January 2022 at 00:50:27, John Covici wrote: > > > Hi. I am using asterisk 18.3 and freepbx. > > Hm, which version of FreePBX uses Asterisk 18.3? > > > How can both sip and pjsip be listening at port 5060 at the same time > > They can't. > > One might be on TCP and the other on UDP, but you can't have them both > listening on the same port with the same protocol. > > > for instance I get: > > > > [2022-01-08 17:08:59] SECURITY[244351] res_security_log.c: > > SecurityEvent="FailedACL",EventTV="2022-01-08T17:08:59.957-0500",Severity=" > > Error",Service="PJSIP",EventVersion="1",AccountID="anonymous",SessionID="20 > > 25076022",LocalAddress="IPV4/UDP/166.84.7.53/5060",RemoteAddress="IPV4/UDP/ > > 45.134.144.118/5823",ACLName="registrar_attempt_without_configured_aors" > > What makes you think chan_sip and pjsip are both listening on UDP 5060? > > > I would like pjsit not to listen,till I figure out how to configure > > the thing, so my logs don't fill up with messages. > > > > Thanks in advance for any suggestions. > > As far as I recall using FreePBX, there is a selector for the SIP protocol to > tell it whether you want it to use pjsip or chan_sip. I don't think it even > supports using both at the same time, so simply make sure that is set to > chan_sip and you should be fine. > > On the other hand, why do you need to learn "how to configure the thing" if > you're using FreePBX? Part of the whole point is that it does the fiddly > techie sutff in the background for you, and you just need to use the > personnel- > department-friendly web GUI. This is what I thought as well, I just generated one trunk using the old chan_sip and expected nothing from pjsit, yet I get all kinds of errors like [2022-01-08 17:08:59] WARNING[487628] res_pjsip_registrar.c: Endpoint 'anonymous' (45.134.144.118:5823) has no configured AORs so I am very confused as to why this is happening. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and maybe a freepbx question
On Sunday 09 January 2022 at 00:50:27, John Covici wrote: > Hi. I am using asterisk 18.3 and freepbx. Hm, which version of FreePBX uses Asterisk 18.3? > How can both sip and pjsip be listening at port 5060 at the same time They can't. One might be on TCP and the other on UDP, but you can't have them both listening on the same port with the same protocol. > for instance I get: > > [2022-01-08 17:08:59] SECURITY[244351] res_security_log.c: > SecurityEvent="FailedACL",EventTV="2022-01-08T17:08:59.957-0500",Severity=" > Error",Service="PJSIP",EventVersion="1",AccountID="anonymous",SessionID="20 > 25076022",LocalAddress="IPV4/UDP/166.84.7.53/5060",RemoteAddress="IPV4/UDP/ > 45.134.144.118/5823",ACLName="registrar_attempt_without_configured_aors" What makes you think chan_sip and pjsip are both listening on UDP 5060? > I would like pjsit not to listen,till I figure out how to configure > the thing, so my logs don't fill up with messages. > > Thanks in advance for any suggestions. As far as I recall using FreePBX, there is a selector for the SIP protocol to tell it whether you want it to use pjsip or chan_sip. I don't think it even supports using both at the same time, so simply make sure that is set to chan_sip and you should be fine. On the other hand, why do you need to learn "how to configure the thing" if you're using FreePBX? Part of the whole point is that it does the fiddly techie sutff in the background for you, and you just need to use the personnel- department-friendly web GUI. Antony. -- "Good health" is merely the slowest rate at which you can die. Please reply to the list; please *don't* CC me. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and maybe a freepbx question
On Sat, 8 Jan 2022, John Covici wrote: How can both sip and pjsip be listening at port 5060 at the same time... They can't. One application per address/port pair. You can configure pjsip to bind to another address and/or port while you figure it out the configuration. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk and maybe a freepbx question
Hi. I am using asterisk 18.3 and freepbx. How can both sip and pjsip be listening at port 5060 at the same time, for instance I get: [2022-01-08 17:08:59] SECURITY[244351] res_security_log.c: SecurityEvent="FailedACL",EventTV="2022-01-08T17:08:59.957-0500",Severity="Error",Service="PJSIP",EventVersion="1",AccountID="anonymous",SessionID="2025076022",LocalAddress="IPV4/UDP/166.84.7.53/5060",RemoteAddress="IPV4/UDP/45.134.144.118/5823",ACLName="registrar_attempt_without_configured_aors" I would like pjsit not to listen,till I figure out how to configure the thing, so my logs don't fill up with messages. Thanks in advance for any suggestions. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici wb2una cov...@ccs.covici.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
