Re: [asterisk-users] Peer doesn't answer
on server side no special configuration is needed. To have qos on the sat link, we contact sat link operator, and I think this is the only way to do it. The codec is g729. I´m not sure about the bandwidth, I think we have about 64Kbps allocated, because we almost don´t have concurrent calls. The quality is very good, you listen everything the other part says, but delayed. From landline to sat link, delay is about 2 seconds. With 2way sat link, it goes to 4, 5 seconds. On Wed, Jan 18, 2012 at 9:03 AM, Arthur Stanfield wrote: > Hi Arlen, > > I'm interested in seeing what setup you settled on to get decent voice > quality over the Sat link? Which codec are you using, and what is the > bandwidth usage?. Are you doing just one concurrent call, Or multiple?. > > - > Regards, > AJ Stanfield > > > - Original Message - > From: "Arlen Nascimento" > To: "Asterisk Users Mailing List - Non-Commercial Discussion" < > asterisk-users@lists.digium.com> > Sent: Wednesday, 18 January, 2012 12:29:23 PM > Subject: Re: [asterisk-users] Peer doesn't answer > > Hi guys, > > the problem was too many NATs on the way. > Although the server had a valid ip, it was behind a nat, as soon as I > set ip directly on the server, things worked fine. > Also, despite the huge delay, if the link has qos, the quality is very > good. > > > > On Mon, Jan 16, 2012 at 9:06 AM, Sammy Govind < govoi...@gmail.com > > wrote: > > > I'm only expecting NAT issues if not the latency issues. SIP traces of > any such calls will make more sense. > > > > > On Mon, Jan 16, 2012 at 6:05 PM, Arlen Nascimento < > arlen.nascime...@gmail.com > wrote: > > > the client is aware of the adverse environment and this is the only > solution for him > > > > > On Mon, Jan 16, 2012 at 9:00 AM, Flavio Miranda < > flaviormira...@hotmail.com > wrote: > > > > > Unless you are doing test with SIP under adverse environmet, that is not > the point, but, if you intend to have Communication, you should worry > about this detail. > Basic infra-estructure is the first thing to think in any new project. > > Good luck! > > Att, > > Flavio Roberto Miranda > MSN:flaviormira...@hotmail.com > Skype: flaviormiranda > > > > > Date: Mon, 16 Jan 2012 07:58:34 -0400 > From: arlen.nascime...@gmail.com > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] Peer doesn't answer > > > > It is a satellite connection, so ping is about 500ms. I know it is not > ok to keep a normal conversation, that is not the point. > > > > On Sun, Jan 15, 2012 at 10:34 PM, Flavio Miranda < > flaviormira...@hotmail.com > wrote: > > > > > Hi Arlen, > > A reasonable time to Voip calls is about 250 ms. What about the Ping > test end-to-end ? > > Att, > > Flavio Roberto Miranda > MSN:flaviormira...@hotmail.com > Skype: flaviormiranda > > > > > Date: Sun, 15 Jan 2012 21:53:46 -0400 > From: arlen.nascime...@gmail.com > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] Peer doesn't answer > > > > Hi all, > > i'm implementing an asterisk server that will have several peers > connected by satellite links. > When qualify=yes or some value (from 3000 to 5), 'sip show peers' > shows the peer as unreachable. In this case i can place calls from the > phone in the satellite link, but can't call to it. > When i turn off qualify, the status changes to unmonitored. In this > case, I can make calls in both directions but the call is never > established. The phone keeps ringing until 'ring time' expires even when > I answer the call on the phone/softphone. > > Any thoughts? > > Regards, > > -- Arlen Nascimento > > > -- _ > -- 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 > > > > -- Arlen Nascimento > > > -- __
Re: [asterisk-users] Peer doesn't answer
Hi Arlen, I'm interested in seeing what setup you settled on to get decent voice quality over the Sat link? Which codec are you using, and what is the bandwidth usage?. Are you doing just one concurrent call, Or multiple?. - Regards, AJ Stanfield - Original Message - From: "Arlen Nascimento" To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Wednesday, 18 January, 2012 12:29:23 PM Subject: Re: [asterisk-users] Peer doesn't answer Hi guys, the problem was too many NATs on the way. Although the server had a valid ip, it was behind a nat, as soon as I set ip directly on the server, things worked fine. Also, despite the huge delay, if the link has qos, the quality is very good. On Mon, Jan 16, 2012 at 9:06 AM, Sammy Govind < govoi...@gmail.com > wrote: I'm only expecting NAT issues if not the latency issues. SIP traces of any such calls will make more sense. On Mon, Jan 16, 2012 at 6:05 PM, Arlen Nascimento < arlen.nascime...@gmail.com > wrote: the client is aware of the adverse environment and this is the only solution for him On Mon, Jan 16, 2012 at 9:00 AM, Flavio Miranda < flaviormira...@hotmail.com > wrote: Unless you are doing test with SIP under adverse environmet, that is not the point, but, if you intend to have Communication, you should worry about this detail. Basic infra-estructure is the first thing to think in any new project. Good luck! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda Date: Mon, 16 Jan 2012 07:58:34 -0400 From: arlen.nascime...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Peer doesn't answer It is a satellite connection, so ping is about 500ms. I know it is not ok to keep a normal conversation, that is not the point. On Sun, Jan 15, 2012 at 10:34 PM, Flavio Miranda < flaviormira...@hotmail.com > wrote: Hi Arlen, A reasonable time to Voip calls is about 250 ms. What about the Ping test end-to-end ? Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda Date: Sun, 15 Jan 2012 21:53:46 -0400 From: arlen.nascime...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Peer doesn't answer Hi all, i'm implementing an asterisk server that will have several peers connected by satellite links. When qualify=yes or some value (from 3000 to 5), 'sip show peers' shows the peer as unreachable. In this case i can place calls from the phone in the satellite link, but can't call to it. When i turn off qualify, the status changes to unmonitored. In this case, I can make calls in both directions but the call is never established. The phone keeps ringing until 'ring time' expires even when I answer the call on the phone/softphone. Any thoughts? Regards, -- Arlen Nascimento -- _ -- 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 -- Arlen Nascimento -- _ -- 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 -- Arlen Nascimento -- _ -- 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 l
Re: [asterisk-users] Peer doesn't answer
Hi guys, the problem was too many NATs on the way. Although the server had a valid ip, it was behind a nat, as soon as I set ip directly on the server, things worked fine. Also, despite the huge delay, if the link has qos, the quality is very good. On Mon, Jan 16, 2012 at 9:06 AM, Sammy Govind wrote: > I'm only expecting NAT issues if not the latency issues. SIP traces of any > such calls will make more sense. > > > On Mon, Jan 16, 2012 at 6:05 PM, Arlen Nascimento < > arlen.nascime...@gmail.com> wrote: > >> the client is aware of the adverse environment and this is the only >> solution for him >> >> >> On Mon, Jan 16, 2012 at 9:00 AM, Flavio Miranda < >> flaviormira...@hotmail.com> wrote: >> >>> Unless you are doing test with SIP under adverse environmet, that is >>> not the point, but, if you intend to have Communication, you should worry >>> about this detail. >>> Basic infra-estructure is the first thing to think in any new project. >>> >>> Good luck! >>> >>> Att, >>> >>> Flavio Roberto Miranda >>> MSN:flaviormira...@hotmail.com >>> Skype: flaviormiranda >>> >>> -- >>> Date: Mon, 16 Jan 2012 07:58:34 -0400 >>> From: arlen.nascime...@gmail.com >>> To: asterisk-users@lists.digium.com >>> Subject: Re: [asterisk-users] Peer doesn't answer >>> >>> >>> It is a satellite connection, so ping is about 500ms. I know it is not >>> ok to keep a normal conversation, that is not the point. >>> >>> >>> On Sun, Jan 15, 2012 at 10:34 PM, Flavio Miranda < >>> flaviormira...@hotmail.com> wrote: >>> >>> Hi Arlen, >>> >>> A reasonable time to Voip calls is about 250 ms. What about the Ping >>> test end-to-end ? >>> >>> Att, >>> >>> Flavio Roberto Miranda >>> MSN:flaviormira...@hotmail.com >>> Skype: flaviormiranda >>> >>> -- >>> Date: Sun, 15 Jan 2012 21:53:46 -0400 >>> From: arlen.nascime...@gmail.com >>> To: asterisk-users@lists.digium.com >>> Subject: [asterisk-users] Peer doesn't answer >>> >>> >>> Hi all, >>> >>> i'm implementing an asterisk server that will have several peers >>> connected by satellite links. >>> When qualify=yes or some value (from 3000 to 5), 'sip show peers' >>> shows the peer as unreachable. In this case i can place calls from the >>> phone in the satellite link, but can't call to it. >>> When i turn off qualify, the status changes to unmonitored. In this >>> case, I can make calls in both directions but the call is never >>> established. The phone keeps ringing until 'ring time' expires even when I >>> answer the call on the phone/softphone. >>> >>> Any thoughts? >>> >>> Regards, >>> >>> -- >>> Arlen Nascimento >>> >>> >>> -- _ >>> -- 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 >>> >>> >>> >>> >>> -- >>> Arlen Nascimento >>> >>> >>> -- _ >>> -- 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 doesn't answer
I'm only expecting NAT issues if not the latency issues. SIP traces of any such calls will make more sense. On Mon, Jan 16, 2012 at 6:05 PM, Arlen Nascimento < arlen.nascime...@gmail.com> wrote: > the client is aware of the adverse environment and this is the only > solution for him > > > On Mon, Jan 16, 2012 at 9:00 AM, Flavio Miranda < > flaviormira...@hotmail.com> wrote: > >> Unless you are doing test with SIP under adverse environmet, that is not >> the point, but, if you intend to have Communication, you should worry about >> this detail. >> Basic infra-estructure is the first thing to think in any new project. >> >> Good luck! >> >> Att, >> >> Flavio Roberto Miranda >> MSN:flaviormira...@hotmail.com >> Skype: flaviormiranda >> >> -- >> Date: Mon, 16 Jan 2012 07:58:34 -0400 >> From: arlen.nascime...@gmail.com >> To: asterisk-users@lists.digium.com >> Subject: Re: [asterisk-users] Peer doesn't answer >> >> >> It is a satellite connection, so ping is about 500ms. I know it is not ok >> to keep a normal conversation, that is not the point. >> >> >> On Sun, Jan 15, 2012 at 10:34 PM, Flavio Miranda < >> flaviormira...@hotmail.com> wrote: >> >> Hi Arlen, >> >> A reasonable time to Voip calls is about 250 ms. What about the Ping >> test end-to-end ? >> >> Att, >> >> Flavio Roberto Miranda >> MSN:flaviormira...@hotmail.com >> Skype: flaviormiranda >> >> -- >> Date: Sun, 15 Jan 2012 21:53:46 -0400 >> From: arlen.nascime...@gmail.com >> To: asterisk-users@lists.digium.com >> Subject: [asterisk-users] Peer doesn't answer >> >> >> Hi all, >> >> i'm implementing an asterisk server that will have several peers >> connected by satellite links. >> When qualify=yes or some value (from 3000 to 5), 'sip show peers' >> shows the peer as unreachable. In this case i can place calls from the >> phone in the satellite link, but can't call to it. >> When i turn off qualify, the status changes to unmonitored. In this case, >> I can make calls in both directions but the call is never established. The >> phone keeps ringing until 'ring time' expires even when I answer the call >> on the phone/softphone. >> >> Any thoughts? >> >> Regards, >> >> -- >> Arlen Nascimento >> >> >> -- _ >> -- 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 >> >> >> >> >> -- >> Arlen Nascimento >> >> >> -- _ >> -- 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 >> > > > > -- > Arlen Nascimento > > > -- > _ > -- 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
Re: [asterisk-users] Peer doesn't answer
the client is aware of the adverse environment and this is the only solution for him On Mon, Jan 16, 2012 at 9:00 AM, Flavio Miranda wrote: > Unless you are doing test with SIP under adverse environmet, that is not > the point, but, if you intend to have Communication, you should worry about > this detail. > Basic infra-estructure is the first thing to think in any new project. > > Good luck! > > Att, > > Flavio Roberto Miranda > MSN:flaviormira...@hotmail.com > Skype: flaviormiranda > > -- > Date: Mon, 16 Jan 2012 07:58:34 -0400 > From: arlen.nascime...@gmail.com > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] Peer doesn't answer > > > It is a satellite connection, so ping is about 500ms. I know it is not ok > to keep a normal conversation, that is not the point. > > > On Sun, Jan 15, 2012 at 10:34 PM, Flavio Miranda < > flaviormira...@hotmail.com> wrote: > > Hi Arlen, > > A reasonable time to Voip calls is about 250 ms. What about the Ping test > end-to-end ? > > Att, > > Flavio Roberto Miranda > MSN:flaviormira...@hotmail.com > Skype: flaviormiranda > > -- > Date: Sun, 15 Jan 2012 21:53:46 -0400 > From: arlen.nascime...@gmail.com > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] Peer doesn't answer > > > Hi all, > > i'm implementing an asterisk server that will have several peers connected > by satellite links. > When qualify=yes or some value (from 3000 to 5), 'sip show peers' > shows the peer as unreachable. In this case i can place calls from the > phone in the satellite link, but can't call to it. > When i turn off qualify, the status changes to unmonitored. In this case, > I can make calls in both directions but the call is never established. The > phone keeps ringing until 'ring time' expires even when I answer the call > on the phone/softphone. > > Any thoughts? > > Regards, > > -- > Arlen Nascimento > > > -- _ > -- 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 > > > > > -- > Arlen Nascimento > > > -- _ > -- 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 > -- Arlen Nascimento -- _ -- 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 doesn't answer
Unless you are doing test with SIP under adverse environmet, that is not the point, but, if you intend to have Communication, you should worry about this detail. Basic infra-estructure is the first thing to think in any new project. Good luck! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda Date: Mon, 16 Jan 2012 07:58:34 -0400 From: arlen.nascime...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Peer doesn't answer It is a satellite connection, so ping is about 500ms. I know it is not ok to keep a normal conversation, that is not the point. On Sun, Jan 15, 2012 at 10:34 PM, Flavio Miranda wrote: Hi Arlen, A reasonable time to Voip calls is about 250 ms. What about the Ping test end-to-end ? Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda Date: Sun, 15 Jan 2012 21:53:46 -0400 From: arlen.nascime...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Peer doesn't answer Hi all, i'm implementing an asterisk server that will have several peers connected by satellite links. When qualify=yes or some value (from 3000 to 5), 'sip show peers' shows the peer as unreachable. In this case i can place calls from the phone in the satellite link, but can't call to it. When i turn off qualify, the status changes to unmonitored. In this case, I can make calls in both directions but the call is never established. The phone keeps ringing until 'ring time' expires even when I answer the call on the phone/softphone. Any thoughts? Regards, -- Arlen Nascimento -- _ -- 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 -- Arlen Nascimento -- _ -- 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
Re: [asterisk-users] Peer doesn't answer
basically CLI shows SIP/X called SIP/Y I answer the call on Y but X keeps ringing and then both hangup. On Mon, Jan 16, 2012 at 8:01 AM, Sammy Govind wrote: > Paste some SIP traces of the call while Unmonitored. > > > On Mon, Jan 16, 2012 at 4:58 PM, Arlen Nascimento < > arlen.nascime...@gmail.com> wrote: > >> It is a satellite connection, so ping is about 500ms. I know it is not ok >> to keep a normal conversation, that is not the point. >> >> >> >> On Sun, Jan 15, 2012 at 10:34 PM, Flavio Miranda < >> flaviormira...@hotmail.com> wrote: >> >>> Hi Arlen, >>> >>> A reasonable time to Voip calls is about 250 ms. What about the Ping >>> test end-to-end ? >>> >>> Att, >>> >>> Flavio Roberto Miranda >>> MSN:flaviormira...@hotmail.com >>> Skype: flaviormiranda >>> >>> ------ >>> Date: Sun, 15 Jan 2012 21:53:46 -0400 >>> From: arlen.nascime...@gmail.com >>> To: asterisk-users@lists.digium.com >>> Subject: [asterisk-users] Peer doesn't answer >>> >>> >>> Hi all, >>> >>> i'm implementing an asterisk server that will have several peers >>> connected by satellite links. >>> When qualify=yes or some value (from 3000 to 5), 'sip show peers' >>> shows the peer as unreachable. In this case i can place calls from the >>> phone in the satellite link, but can't call to it. >>> When i turn off qualify, the status changes to unmonitored. In this >>> case, I can make calls in both directions but the call is never >>> established. The phone keeps ringing until 'ring time' expires even when I >>> answer the call on the phone/softphone. >>> >>> Any thoughts? >>> >>> Regards, >>> >>> -- >>> Arlen Nascimento >>> >>> >>> -- _ >>> -- 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 >>> >> >> >> >> -- >> Arlen Nascimento >> >> >> -- >> _ >> -- 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 > -- Arlen Nascimento -- _ -- 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 doesn't answer
Paste some SIP traces of the call while Unmonitored. On Mon, Jan 16, 2012 at 4:58 PM, Arlen Nascimento < arlen.nascime...@gmail.com> wrote: > It is a satellite connection, so ping is about 500ms. I know it is not ok > to keep a normal conversation, that is not the point. > > > > On Sun, Jan 15, 2012 at 10:34 PM, Flavio Miranda < > flaviormira...@hotmail.com> wrote: > >> Hi Arlen, >> >> A reasonable time to Voip calls is about 250 ms. What about the Ping >> test end-to-end ? >> >> Att, >> >> Flavio Roberto Miranda >> MSN:flaviormira...@hotmail.com >> Skype: flaviormiranda >> >> -- >> Date: Sun, 15 Jan 2012 21:53:46 -0400 >> From: arlen.nascime...@gmail.com >> To: asterisk-users@lists.digium.com >> Subject: [asterisk-users] Peer doesn't answer >> >> >> Hi all, >> >> i'm implementing an asterisk server that will have several peers >> connected by satellite links. >> When qualify=yes or some value (from 3000 to 5), 'sip show peers' >> shows the peer as unreachable. In this case i can place calls from the >> phone in the satellite link, but can't call to it. >> When i turn off qualify, the status changes to unmonitored. In this case, >> I can make calls in both directions but the call is never established. The >> phone keeps ringing until 'ring time' expires even when I answer the call >> on the phone/softphone. >> >> Any thoughts? >> >> Regards, >> >> -- >> Arlen Nascimento >> >> >> -- _ >> -- 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 >> > > > > -- > Arlen Nascimento > > > -- > _ > -- 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
Re: [asterisk-users] Peer doesn't answer
It is a satellite connection, so ping is about 500ms. I know it is not ok to keep a normal conversation, that is not the point. On Sun, Jan 15, 2012 at 10:34 PM, Flavio Miranda wrote: > Hi Arlen, > > A reasonable time to Voip calls is about 250 ms. What about the Ping test > end-to-end ? > > Att, > > Flavio Roberto Miranda > MSN:flaviormira...@hotmail.com > Skype: flaviormiranda > > -- > Date: Sun, 15 Jan 2012 21:53:46 -0400 > From: arlen.nascime...@gmail.com > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] Peer doesn't answer > > > Hi all, > > i'm implementing an asterisk server that will have several peers connected > by satellite links. > When qualify=yes or some value (from 3000 to 5), 'sip show peers' > shows the peer as unreachable. In this case i can place calls from the > phone in the satellite link, but can't call to it. > When i turn off qualify, the status changes to unmonitored. In this case, > I can make calls in both directions but the call is never established. The > phone keeps ringing until 'ring time' expires even when I answer the call > on the phone/softphone. > > Any thoughts? > > Regards, > > -- > Arlen Nascimento > > > -- _ > -- 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 > -- Arlen Nascimento -- _ -- 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 doesn't answer
Hi Arlen, A reasonable time to Voip calls is about 250 ms. What about the Ping test end-to-end ? Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda Date: Sun, 15 Jan 2012 21:53:46 -0400 From: arlen.nascime...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Peer doesn't answer Hi all, i'm implementing an asterisk server that will have several peers connected by satellite links. When qualify=yes or some value (from 3000 to 5), 'sip show peers' shows the peer as unreachable. In this case i can place calls from the phone in the satellite link, but can't call to it. When i turn off qualify, the status changes to unmonitored. In this case, I can make calls in both directions but the call is never established. The phone keeps ringing until 'ring time' expires even when I answer the call on the phone/softphone. Any thoughts? Regards, -- Arlen Nascimento -- _ -- 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
[asterisk-users] Peer doesn't answer
Hi all, i'm implementing an asterisk server that will have several peers connected by satellite links. When qualify=yes or some value (from 3000 to 5), 'sip show peers' shows the peer as unreachable. In this case i can place calls from the phone in the satellite link, but can't call to it. When i turn off qualify, the status changes to unmonitored. In this case, I can make calls in both directions but the call is never established. The phone keeps ringing until 'ring time' expires even when I answer the call on the phone/softphone. Any thoughts? Regards, -- Arlen Nascimento -- _ -- 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
