Try adding nat=yes to your config.. Also if you want to make SIP to SIP extension calls and don't want to fight with the NAT set canreinvite=yes to canreinvite=no.. Finally set dtmfmode=info for the GS phones.. Later..> Hi there! > I installed the BudgetTone (GrandStream) on my LAN without any problems. > Then, I moved it to another location using a D-Link NAT. > I opened 5060 (SIP) and 5000 to 5008 for RTP. I also fixed the IP address > of the BudgetTone. > When I receive a call on my Asterisk, it would ring my FXS as before. > However, after I pick up, it hangs within a few seconds (Hungup Zap1-1 in > the log). > The configuration I have in * is the following: > sip.conf > ----------- > [general] > port=5060 > context=sip > maxexpirey=3600 > defaultexpirey=60 > disallow=all > allow=ulaw > allow=gsm > [1000] > contet=sip > type=friend > username=1000 > secret=????? (not the real one) > host=dynamic > mailbox=1000 > canreinvite=yes > dtmfmode=rfc2833 > > I did not change the above configuration when I moved the budgetTone from > the LAN to the Internet (Wan). > I am not using a "register" statement in the sip.conf and I am wondering if > I need to. > I did change the sip server IP address in the Grandstream configuration. > > I suspect my problem is with the router (NAT). I don't quite understand the > symetric discussions but I downloaded a paper to learn more. Right now, all > my public and private ports are the same. > > Regards, > Uriel >-- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze
Hi there! I installed the BudgetTone (GrandStream) on my LAN without any problems. Then, I moved it to another location using a D-Link NAT. I opened 5060 (SIP) and 5000 to 5008 for RTP. I also fixed the IP address of the BudgetTone. When I receive a call on my Asterisk, it would ring my FXS as before. However, after I pick up, it hangs within a few seconds (Hungup Zap1-1 in the log). The configuration I have in * is the following: sip.conf ----------- [general] port=5060 context=sip maxexpirey=3600 defaultexpirey=60 disallow=all allow=ulaw allow=gsm [1000] contet=sip type=friend username=1000 secret=????? (not the real one) host=dynamic mailbox=1000 canreinvite=yes dtmfmode=rfc2833 I did not change the above configuration when I moved the budgetTone from the LAN to the Internet (Wan). I am not using a "register" statement in the sip.conf and I am wondering if I need to. I did change the sip server IP address in the Grandstream configuration. I suspect my problem is with the router (NAT). I don't quite understand the symetric discussions but I downloaded a paper to learn more. Right now, all my public and private ports are the same. Regards, Uriel -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030924/59c0c6f0/attachment.htm
have you tried to put nat=yes in the user definition in sip.conf ? Also, the * server is on a public IP? Matteo Il mer, 2003-09-24 alle 15:35, Uriel Carrasquilla ha scritto:> Hi there! > I installed the BudgetTone (GrandStream) on my LAN without any > problems. Then, I moved it to another location using a D-Link NAT. > I opened 5060 (SIP) and 5000 to 5008 for RTP. I also fixed the IP > address of the BudgetTone. > When I receive a call on my Asterisk, it would ring my FXS as before. > However, after I pick up, it hangs within a few seconds (Hungup Zap1-1 > in the log). > The configuration I have in * is the following: > sip.conf > ----------- > [general] > port=5060 > context=sip > maxexpirey=3600 > defaultexpirey=60 > disallow=all > allow=ulaw > allow=gsm > [1000] > contet=sip > type=friend > username=1000 > secret=????? (not the real one) > host=dynamic > mailbox=1000 > canreinvite=yes > dtmfmode=rfc2833 > > I did not change the above configuration when I moved the budgetTone > from the LAN to the Internet (Wan). > I am not using a "register" statement in the sip.conf and I am > wondering if I need to. > I did change the sip server IP address in the Grandstream > configuration. > > I suspect my problem is with the router (NAT). I don't quite > understand the symetric discussions but I downloaded a paper to learn > more. Right now, all my public and private ports are the same. > > Regards, > Uriel >-- Brancaleoni Matteo <mbrancaleoni@espia.it> Espia - Emmegi Srl
A plain wireless dlink dsl router. Stephen Varga wrote:>On Thu, 2003-09-25 at 10:42, Michael Koehler wrote: > > >>Sorry, but my * is behind NAT and i have no problems with SIP, and it >>even works with NAT to NAT and without forwarding ports or similar >>effords. >> >> >>Michael >> >> > > >What kinda box/device is doing the NAT? > >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users > > > >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030925/3bdb1231/attachment.htm
Europe (Germany) and US (Calif.) Dave Cotton wrote:>On Thu, 2003-09-25 at 19:10, Michael Koehler wrote: > > >>Looks interesting, got retail prices? >> >> > >Where are you in the world? > > >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030925/037f1326/attachment.htm
Hi I have that error messages, what does it mean? *CLI> WARNING[65545]: File chan_sip.c, Line 435 (retrans_pkt): Maximum retries exceeded on call 051c345b67bbb1fb1bf19905727d015c@127.0.0.1 for seqno 102 (Request) WARNING[65545]: File chan_sip.c, Line 435 (retrans_pkt): Maximum retries exceeded on call 1b8b002425c5d97234992de354bf5892@192.168.0.32 for seqno 102 (Request) WARNING[65545]: File chan_sip.c, Line 435 (retrans_pkt): Maximum retries exceeded on call 051c345b67bbb1fb1bf19905727d015c@192.168.0.32 for seqno 102 (Request) WARNING[65545]: File chan_sip.c, Line 435 (retrans_pkt): Maximum retries exceeded on call 051c345b67bbb1fb1bf19905727d015c@127.0.0.1 for seqno 103 (Request) WARNING[65545]: File chan_sip.c, Line 435 (retrans_pkt): Maximum retries exceeded on call 051c345b67bbb1fb1bf19905727d015c@127.0.0.1 for seqno 104 (Request) miklos
I had this problem when I changed the IP of one of the * boxes. Did not see it on the other boxes. Have you changed the IP of your * box since compiling * first time? Senad
On Thu, 2003-09-25 at 12:54, Michael Koehler wrote:> A plain wireless dlink dsl router.Do you know the model number and the software version? I am trying to understand how it is making the appropriate adjustments to allow the connection to work. Thanks, Steve
This means that your Request is not confirmed after N tries. This could happen for example) when your NAT is not working with your VoIP provider. Other reasons possible (e.g. no internet connection) listas iPfone wrote:>Hi > >I have that error messages, what does it mean? > >*CLI> WARNING[65545]: File chan_sip.c, Line 435 (retrans_pkt): Maximum >retries exceeded on call 051c345b67bbb1fb1bf19905727d015c@127.0.0.1 for >seqno 102 (Request) >WARNING[65545]: File chan_sip.c, Line 435 (retrans_pkt): Maximum retries >exceeded on call 1b8b002425c5d97234992de354bf5892@192.168.0.32 for seqno 102 >(Request) >WARNING[65545]: File chan_sip.c, Line 435 (retrans_pkt): Maximum retries >exceeded on call 051c345b67bbb1fb1bf19905727d015c@192.168.0.32 for seqno 102 >(Request) >WARNING[65545]: File chan_sip.c, Line 435 (retrans_pkt): Maximum retries >exceeded on call 051c345b67bbb1fb1bf19905727d015c@127.0.0.1 for seqno 103 >(Request) >WARNING[65545]: File chan_sip.c, Line 435 (retrans_pkt): Maximum retries >exceeded on call 051c345b67bbb1fb1bf19905727d015c@127.0.0.1 for seqno 104 >(Request) > >miklos > >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users > > > >
Michael: is it a D-link on both NAT? the one for * and the one for the Grand Stream? Uriel -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Michael Koehler Sent: Thursday, September 25, 2003 12:55 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] SIP / GrandStream Configuration A plain wireless dlink dsl router. Stephen Varga wrote: On Thu, 2003-09-25 at 10:42, Michael Koehler wrote: Sorry, but my * is behind NAT and i have no problems with SIP, and it even works with NAT to NAT and without forwarding ports or similar effords. Michael What kinda box/device is doing the NAT? _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030925/31fd3c5d/attachment.htm
Hi! Thaanks the problem was the same, now i?m using a static ip and all is working fine. regards ----- Original Message ----- From: "Senad Jordanovic" <senad@boltblue.com> To: <asterisk-users@lists.digium.com> Sent: Thursday, September 25, 2003 4:07 PM Subject: RE: [Asterisk-Users] ERROR MESSAGE> I had this problem when I changed the IP of > one of the * boxes. Did not see it on the other boxes. > > Have you changed the IP of your * box since compiling * first time? > > Senad > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users >