Hello Sebastian, I had already seen this list of the hosts, but it is not active. All servers with which my Asterisk has been communicated are not listed. A port scan, to eventually update the list, found hundreds of servers provided in the address range 217.0.0.0/13 with open port 5060, some were even not found. I think there must be another solution. If I change insecure to insecure=port,invite - could that be a solution? Or should I try to change to res_pjsip (upgrade to Asterisk 13 is no problem)? Has there anyone experience with dynamic ip addresses of Asterisk? Daniel> Am 30.03.2015 um 20:09 schrieb Sebastian Kemper <sebastian_ml at gmx.net>: > > On Mon, Mar 30, 2015 at 06:31:46PM +0200, Daniel Heckl wrote: >> Hello >> >> I use Asterisk 11 with FreePBX 12. Our SIP Provider is Telekom >> Germany. We have sometimes problems with incoming and outgoing calls. >> I hope I can explain it understandable. > > Hello Daniel, > > I'll find myself in the same situation a few weeks from now :-) > >> >> For example, Asterisk sends a REGISTER to 217.0.23.68 (tel.t-online.de >> <http://tel.t-online.de/>), the message is answered with OK and the >> peer is registered. >> >> Usually INVITES comes now from this ip address. All works fine. But >> sometimes INVITES comes from an other IP address, for example >> 217.0.23.100. This request Asterisk responds with 401 Unauthorized. >> >> In the next register procedure REGISTER are sent to the new ip address >> and answered also with OK. But qualify OPTIONS are continue be sent to >> the old ip address. Incoming and outgoing calls are canceled. Outgoing >> calls are answered with Forbidden. >> >> Even if the REGISTER procedure works with the new ip address, the >> peers are connected with the old address. >> >> Waiting doesn?t help, only a ?sip reload? update the ip address of the >> peer. >> >> What is the solution for this problem? How can asterisk update the >> peer? > > I think the solution - for the inbound issue at least - could be to add > more hosts as a peer. Have a looks at this forum post: > > http://www.ip-phone-forum.de/showthread.php?t=268787&p=1999371&viewfull=1#post1999371 > > The user used a template and than he added peers, each with its own IP > address. The provided list was last updated in 2014, though, so I assume > the provider in the meantime has added to that list. > > It looks pretty tedious, though, I mean there could be dozens of IPs > you'd have to add. But I guess this is the way to go with Asterisk 11 > and chan_sip. > > The future looks brighter :-) I read that with pjsip, which I understand > is the replacement for chan_sip, you can have one peer entry and match > an IP range instead of a single host. That should tidy up the dialplan. > > What I'm a little afraid of is the SIP provider using IPs out of a range > that they also use for other services. Maybe out of the same range they > hand out IPs to their customers. I guess we got to be careful :-) > > Kind regards, > Sebastian > >> The Asterisk is local behind a NAT with a firewall, following settings >> are used: >> >> externhost with DynDNS stun with stun.t-online.de >> <http://stun.t-online.de/> nat=yes srvlookup=yes allowguest=no >> trustrpid=no insecure=invite qualify=yes >> >> Thank you! Daniel > >> -- >> _____________________________________________________________________ >> -- 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 > > > -- > _____________________________________________________________________ > -- 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-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150331/00ca37af/attachment.html>
Maybe someone could elaborate on my first question again. If the ip address changes while a REGISTER period, the ip address of the peer isn't been updated. How can asterisk update the ip address of the peer?> Am 31.03.2015 um 12:36 schrieb Daniel Heckl <daniel.heckl at gmail.com>: > > Hello Sebastian, > > I had already seen this list of the hosts, but it is not active. All servers with which my Asterisk has been communicated are not listed. > > A port scan, to eventually update the list, found hundreds of servers provided in the address range 217.0.0.0/13 with open port 5060, some were even not found. I think there must be another solution. > > If I change insecure to insecure=port,invite - could that be a solution? > > Or should I try to change to res_pjsip (upgrade to Asterisk 13 is no problem)? Has there anyone experience with dynamic ip addresses of Asterisk? > > Daniel > >> Am 30.03.2015 um 20:09 schrieb Sebastian Kemper <sebastian_ml at gmx.net>: >> >> On Mon, Mar 30, 2015 at 06:31:46PM +0200, Daniel Heckl wrote: >>> Hello >>> >>> I use Asterisk 11 with FreePBX 12. Our SIP Provider is Telekom >>> Germany. We have sometimes problems with incoming and outgoing calls. >>> I hope I can explain it understandable. >> >> Hello Daniel, >> >> I'll find myself in the same situation a few weeks from now :-) >> >>> >>> For example, Asterisk sends a REGISTER to 217.0.23.68 (tel.t-online.de >>> <http://tel.t-online.de/>), the message is answered with OK and the >>> peer is registered. >>> >>> Usually INVITES comes now from this ip address. All works fine. But >>> sometimes INVITES comes from an other IP address, for example >>> 217.0.23.100. This request Asterisk responds with 401 Unauthorized. >>> >>> In the next register procedure REGISTER are sent to the new ip address >>> and answered also with OK. But qualify OPTIONS are continue be sent to >>> the old ip address. Incoming and outgoing calls are canceled. Outgoing >>> calls are answered with Forbidden. >>> >>> Even if the REGISTER procedure works with the new ip address, the >>> peers are connected with the old address. >>> >>> Waiting doesn?t help, only a ?sip reload? update the ip address of the >>> peer. >>> >>> What is the solution for this problem? How can asterisk update the >>> peer? >> >> I think the solution - for the inbound issue at least - could be to add >> more hosts as a peer. Have a looks at this forum post: >> >> http://www.ip-phone-forum.de/showthread.php?t=268787&p=1999371&viewfull=1#post1999371 >> >> The user used a template and than he added peers, each with its own IP >> address. The provided list was last updated in 2014, though, so I assume >> the provider in the meantime has added to that list. >> >> It looks pretty tedious, though, I mean there could be dozens of IPs >> you'd have to add. But I guess this is the way to go with Asterisk 11 >> and chan_sip. >> >> The future looks brighter :-) I read that with pjsip, which I understand >> is the replacement for chan_sip, you can have one peer entry and match >> an IP range instead of a single host. That should tidy up the dialplan. >> >> What I'm a little afraid of is the SIP provider using IPs out of a range >> that they also use for other services. Maybe out of the same range they >> hand out IPs to their customers. I guess we got to be careful :-) >> >> Kind regards, >> Sebastian >> >>> The Asterisk is local behind a NAT with a firewall, following settings >>> are used: >>> >>> externhost with DynDNS stun with stun.t-online.de >>> <http://stun.t-online.de/> nat=yes srvlookup=yes allowguest=no >>> trustrpid=no insecure=invite qualify=yes >>> >>> Thank you! Daniel >> >>> -- >>> _____________________________________________________________________ >>> -- 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 >> >> >> -- >> _____________________________________________________________________ >> -- 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150331/49c8b4b7/attachment.html>
You have two options for dealing with an IP change during the registration period: 1) set the registration time to shorter period of time to minimize the downtime 2) detect that the IP address has changed via whatever method available, and then issue a "sip reload" CLI command to asterisk, which will cause it to resend registrations immediately. On Tue, Mar 31, 2015 at 1:36 PM, Daniel Heckl <daniel.heckl at gmail.com> wrote:> Maybe someone could elaborate on my first question again. > > If the ip address changes while a REGISTER period, the ip address of the > peer isn't been updated. How can asterisk update the ip address of the peer? > > Am 31.03.2015 um 12:36 schrieb Daniel Heckl <daniel.heckl at gmail.com>: > > Hello Sebastian, > > I had already seen this list of the hosts, but it is not active. All > servers with which my Asterisk has been communicated are not listed. > > A port scan, to eventually update the list, found hundreds of servers > provided in the address range 217.0.0.0/13 with open port 5060, some were > even not found. I think there must be another solution. > > If I change insecure to insecure=port,invite - could that be a solution? > > Or should I try to change to res_pjsip (upgrade to Asterisk 13 is no > problem)? Has there anyone experience with dynamic ip addresses of Asterisk? > > Daniel > > Am 30.03.2015 um 20:09 schrieb Sebastian Kemper <sebastian_ml at gmx.net>: > > On Mon, Mar 30, 2015 at 06:31:46PM +0200, Daniel Heckl wrote: > > Hello > > I use Asterisk 11 with FreePBX 12. Our SIP Provider is Telekom > Germany. We have sometimes problems with incoming and outgoing calls. > I hope I can explain it understandable. > > > Hello Daniel, > > I'll find myself in the same situation a few weeks from now :-) > > > For example, Asterisk sends a REGISTER to 217.0.23.68 (tel.t-online.de > <http://tel.t-online.de/>), the message is answered with OK and the > peer is registered. > > Usually INVITES comes now from this ip address. All works fine. But > sometimes INVITES comes from an other IP address, for example > 217.0.23.100. This request Asterisk responds with 401 Unauthorized. > > In the next register procedure REGISTER are sent to the new ip address > and answered also with OK. But qualify OPTIONS are continue be sent to > the old ip address. Incoming and outgoing calls are canceled. Outgoing > calls are answered with Forbidden. > > Even if the REGISTER procedure works with the new ip address, the > peers are connected with the old address. > > Waiting doesn?t help, only a ?sip reload? update the ip address of the > peer. > > What is the solution for this problem? How can asterisk update the > peer? > > > I think the solution - for the inbound issue at least - could be to add > more hosts as a peer. Have a looks at this forum post: > > > http://www.ip-phone-forum.de/showthread.php?t=268787&p=1999371&viewfull=1#post1999371 > > The user used a template and than he added peers, each with its own IP > address. The provided list was last updated in 2014, though, so I assume > the provider in the meantime has added to that list. > > It looks pretty tedious, though, I mean there could be dozens of IPs > you'd have to add. But I guess this is the way to go with Asterisk 11 > and chan_sip. > > The future looks brighter :-) I read that with pjsip, which I understand > is the replacement for chan_sip, you can have one peer entry and match > an IP range instead of a single host. That should tidy up the dialplan. > > What I'm a little afraid of is the SIP provider using IPs out of a range > that they also use for other services. Maybe out of the same range they > hand out IPs to their customers. I guess we got to be careful :-) > > Kind regards, > Sebastian > > The Asterisk is local behind a NAT with a firewall, following settings > are used: > > externhost with DynDNS stun with stun.t-online.de > <http://stun.t-online.de/> nat=yes srvlookup=yes allowguest=no > trustrpid=no insecure=invite qualify=yes > > Thank you! Daniel > > > -- > _____________________________________________________________________ > -- 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 > > > > -- > _____________________________________________________________________ > -- 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 > > > > -- > _____________________________________________________________________ > -- 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 >-- [image: Digium logo] Scott Griepentrog Digium, Inc ? Software Developer 445 Jan Davis Drive NW ? Huntsville, AL 35806 ? US direct/fax: +1 256 428 6239 ? mobile: +1 256 580 6090 Check us out at: http://digium.com ? http://asterisk.org -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150331/0f9a75ec/attachment.html>
On Tue, Mar 31, 2015 at 12:36:34PM +0200, Daniel Heckl wrote:> Hello Sebastian, > > I had already seen this list of the hosts, but it is not active. All > servers with which my Asterisk has been communicated are not listed. > > A port scan, to eventually update the list, found hundreds of servers > provided in the address range 217.0.0.0/13 with open port 5060, some > were even not found. I think there must be another solution. > > If I change insecure to insecure=port,invite - could that be a > solution?Hello Daniel, I've asked myself that, too. But I don't have access to the connection, yet, so I can't test it right away. Kind regards, Sebastian