RE: [Asterisk-Users] SIP / GrandStream Configuration
What we need are the "nuclear scientists" at Nikotel sharing their solution. I am wondering if they are using a Linux/NAT. Regards, Uriel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Stephen Varga Sent: Thursday, September 25, 2003 4:46 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] SIP / GrandStream Configuration On Thu, 2003-09-25 at 15:41, Michael Koehler wrote: > It is not a feature of the router, it is the way SIP is handled with > nikotel.com > > I recently wrote that i'm using just a plain router with my natted > asterisk because "Stephen Varga" wrote that SIP behind > NAT (in relation to asterisk) is impossible. It is possible because > i'm using asterisk this way. > > There is also nothing special to setup with the router for nikotel and > NAT, except you have a firewall and need > straight rules, then you may use port forwarding. Ok maybe I was being to broad in my original statement, so let me clarify. There orginal question was does the scenario SIP Phone --- NAT --- Internet --- NAT --- Asterisk work. In general this can not be easily accomplished, because of the real ip address of the devices get embedded in SDP message during the INVITE process. Most phones can be changed to use the NAT address in this process, so this solves one side of the conversation. However I have not found away to do this in the asterisk software, thus SDP message needs modified to change the ip address to the NATed one outside of * for this to work. For this I have not discovered a reasonable solution. In Mike's case, I am guessing the SDP message is being modified when the packet arrives at the Nikotel's gateway. Which makes this a specialized case. So that still leaves us with a general problem of SIP and NATing on both sides, for the rest of us not having the benefit of the software that nikotel is using to make this scenario work. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP / GrandStream Configuration
Michael: I am working in a second language and I might be loosing some subtle points. Please over communicate to make your points. are you saying that two garden variety D-Link NAT routers working on two ends of the Interent with one end running a SIP/GrandStream IP-Phone and the other running * will work? This is where Stephen stated that it will NOT. You seem to be saying it will work. Is Nikotel doing anything special that allows them to work this type of configuration? Please elaborate. Regards, Uriel -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Michael KoehlerSent: Thursday, September 25, 2003 3:41 PMTo: [EMAIL PROTECTED]Subject: Re: [Asterisk-Users] SIP / GrandStream ConfigurationIt is not a feature of the router, it is the way SIP is handled with nikotel.comI recently wrote that i'm using just a plain router with my natted asterisk because "Stephen Varga" wrote that SIP behindNAT (in relation to asterisk) is impossible. It is possible because i'm using asterisk this way.There is also nothing special to setup with the router for nikotel and NAT, except you have a firewall and needstraight rules, then you may use port forwarding.MichaelStephen Varga wrote: 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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP / GrandStream Configuration
Michael: is it a D-link on both NAT? the one for * and the one for the Grand Stream? Uriel -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Michael KoehlerSent: Thursday, September 25, 2003 12:55 PMTo: [EMAIL PROTECTED]Subject: Re: [Asterisk-Users] SIP / GrandStream ConfigurationA 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP / GrandStream Configuration
Michael: could you share how you configured your GrandStream? for example, did you say "yes" to NAT (without a STUN)? how about in SIP.CONF, how did you configure the remote GrandStream? Regards, Uriel -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Michael KoehlerSent: Thursday, September 25, 2003 10:42 AMTo: [EMAIL PROTECTED]Subject: Re: [Asterisk-Users] SIP / GrandStream ConfigurationSorry, 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.MichaelStephen Varga wrote: On Wed, 2003-09-24 at 21:50, Uriel Carrasquilla wrote: Adam: in reference to my first message, the NAT on the SIP/GS (a D-Link router) has ports 5060 for SIP-registration and RTP ports 5000 to 5008 being forwarded to the Sip/GS. The Asterisk server, also behind another NAT (Linksys), has the same ports opened and forwarded. is it still impossible? URiel Nope, it is not currently possible. * behind a NAT for SIP does not work because the * real IP address is placed in the SDP information, therefore the 'outside' phone can not send the media stream to *. See my answers over the last week for the more details and possible work arounds. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP / GrandStream Configuration
On Thu, 2003-09-25 at 15:41, Michael Koehler wrote: > It is not a feature of the router, it is the way SIP is handled with > nikotel.com > > I recently wrote that i'm using just a plain router with my natted > asterisk because "Stephen Varga" wrote that SIP behind > NAT (in relation to asterisk) is impossible. It is possible because > i'm using asterisk this way. > > There is also nothing special to setup with the router for nikotel and > NAT, except you have a firewall and need > straight rules, then you may use port forwarding. Ok maybe I was being to broad in my original statement, so let me clarify. There orginal question was does the scenario SIP Phone --- NAT --- Internet --- NAT --- Asterisk work. In general this can not be easily accomplished, because of the real ip address of the devices get embedded in SDP message during the INVITE process. Most phones can be changed to use the NAT address in this process, so this solves one side of the conversation. However I have not found away to do this in the asterisk software, thus SDP message needs modified to change the ip address to the NATed one outside of * for this to work. For this I have not discovered a reasonable solution. In Mike's case, I am guessing the SDP message is being modified when the packet arrives at the Nikotel's gateway. Which makes this a specialized case. So that still leaves us with a general problem of SIP and NATing on both sides, for the rest of us not having the benefit of the software that nikotel is using to make this scenario work. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP / GrandStream Configuration
It is not a feature of the router, it is the way SIP is handled with nikotel.com I recently wrote that i'm using just a plain router with my natted asterisk because "Stephen Varga" wrote that SIP behind NAT (in relation to asterisk) is impossible. It is possible because i'm using asterisk this way. There is also nothing special to setup with the router for nikotel and NAT, except you have a firewall and need straight rules, then you may use port forwarding. Michael Stephen Varga wrote: 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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP / GrandStream Configuration
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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP / GrandStream Configuration
I´m doing the same, ix66 > asterisk. Did you registered asterisk in the ix66? Please share your set up, i´m with some truble using ICH . Miklos - Original Message - From: "Dave Cotton" <[EMAIL PROTECTED]> To: "Asterisk List" <[EMAIL PROTECTED]> Sent: Thursday, September 25, 2003 2:03 PM Subject: Re: [Asterisk-Users] SIP / GrandStream Configuration > On Thu, 2003-09-25 at 18:54, Michael Koehler wrote: > > A plain wireless dlink dsl router. > > I'm testing one of these > > http://www.intertex.se > > and my * is behind it. > -- > Dave Cotton <[EMAIL PROTECTED]> > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP / GrandStream Configuration
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?
Re: [Asterisk-Users] SIP / GrandStream Configuration
On Thu, 2003-09-25 at 19:10, Michael Koehler wrote: > Looks interesting, got retail prices? Where are you in the world? -- Dave Cotton <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP / GrandStream Configuration
Looks interesting, got retail prices? Dave Cotton wrote: On Thu, 2003-09-25 at 18:54, Michael Koehler wrote: A plain wireless dlink dsl router. I'm testing one of these http://www.intertex.se and my * is behind it.
Re: [Asterisk-Users] SIP / GrandStream Configuration
On Thu, 2003-09-25 at 18:54, Michael Koehler wrote: > A plain wireless dlink dsl router. I'm testing one of these http://www.intertex.se and my * is behind it. -- Dave Cotton <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP / GrandStream Configuration
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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP / GrandStream Configuration
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 Stephen Varga wrote: On Wed, 2003-09-24 at 21:50, Uriel Carrasquilla wrote: Adam: in reference to my first message, the NAT on the SIP/GS (a D-Link router) has ports 5060 for SIP-registration and RTP ports 5000 to 5008 being forwarded to the Sip/GS. The Asterisk server, also behind another NAT (Linksys), has the same ports opened and forwarded. is it still impossible? URiel Nope, it is not currently possible. * behind a NAT for SIP does not work because the * real IP address is placed in the SDP information, therefore the 'outside' phone can not send the media stream to *. See my answers over the last week for the more details and possible work arounds. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP / GrandStream Configuration
On Thu, Sep 25, 2003 at 12:33:02AM -0400, Uriel Carrasquilla wrote: > Adam: > I believe you. I assume that the RTP is creating a symetric configuration > between * and the SIP phone. The situation we are left to live with is that > * (won't be the Sip phone) can only live in the Internet brave world (and > not behind a firewall). is this acceptable? > Uriel You could set up a tunnel between both NATed networks so they didn't need to use NAT. Assuming there are no IP conflicts. This will add some latency, I'm not sure how much. cheers, Woody > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of Stephen Varga > Sent: Wednesday, September 24, 2003 11:02 PM > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] SIP / GrandStream Configuration > > > On Wed, 2003-09-24 at 21:50, Uriel Carrasquilla wrote: > > Adam: > > in reference to my first message, the NAT on the SIP/GS (a D-Link router) > > has ports 5060 for SIP-registration and RTP ports 5000 to 5008 being > > forwarded to the Sip/GS. > > The Asterisk server, also behind another NAT (Linksys), has the same ports > > opened and forwarded. > > is it still impossible? > > URiel > > Nope, it is not currently possible. * behind a NAT for SIP does not work > because the * real IP address is placed in the SDP information, > therefore the 'outside' phone can not send the media stream to *. See my > answers over the last week for the more details and possible work > arounds. > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users -- Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP / GrandStream Configuration
Adam: I believe you. I assume that the RTP is creating a symetric configuration between * and the SIP phone. The situation we are left to live with is that * (won't be the Sip phone) can only live in the Internet brave world (and not behind a firewall). is this acceptable? Uriel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Stephen Varga Sent: Wednesday, September 24, 2003 11:02 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] SIP / GrandStream Configuration On Wed, 2003-09-24 at 21:50, Uriel Carrasquilla wrote: > Adam: > in reference to my first message, the NAT on the SIP/GS (a D-Link router) > has ports 5060 for SIP-registration and RTP ports 5000 to 5008 being > forwarded to the Sip/GS. > The Asterisk server, also behind another NAT (Linksys), has the same ports > opened and forwarded. > is it still impossible? > URiel Nope, it is not currently possible. * behind a NAT for SIP does not work because the * real IP address is placed in the SDP information, therefore the 'outside' phone can not send the media stream to *. See my answers over the last week for the more details and possible work arounds. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP / GrandStream Configuration
On Wed, 2003-09-24 at 21:50, Uriel Carrasquilla wrote: > Adam: > in reference to my first message, the NAT on the SIP/GS (a D-Link router) > has ports 5060 for SIP-registration and RTP ports 5000 to 5008 being > forwarded to the Sip/GS. > The Asterisk server, also behind another NAT (Linksys), has the same ports > opened and forwarded. > is it still impossible? > URiel Nope, it is not currently possible. * behind a NAT for SIP does not work because the * real IP address is placed in the SDP information, therefore the 'outside' phone can not send the media stream to *. See my answers over the last week for the more details and possible work arounds. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP / GrandStream Configuration
Adam: in reference to my first message, the NAT on the SIP/GS (a D-Link router) has ports 5060 for SIP-registration and RTP ports 5000 to 5008 being forwarded to the Sip/GS. The Asterisk server, also behind another NAT (Linksys), has the same ports opened and forwarded. is it still impossible? URiel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Adam Hart Sent: Wednesday, September 24, 2003 7:27 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] SIP / GrandStream Configuration How will the packets get to the asterisk server? You'd need to forward ports on the NAT device, otherwise it's impossible - Original Message - From: "Uriel Carrasquilla" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Thursday, September 25, 2003 9:48 AM Subject: RE: [Asterisk-Users] SIP / GrandStream Configuration > Very valuable help. It is now working like a champ. > > This is a solution with SIP--NAT---Internet---Asterisk. No problems here. > > What I would like to do next is to move Asterisk behind a NAT as follows > SIP---NAT---Internet---NAT---Asterisk > do I need a STUN server? is there a chance this could work? > The Google results seems to indicate that I will get an ulcer attempting > this step. is that true? > > Regards, > Uriel > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of WipeOut . > Sent: Wednesday, September 24, 2003 9:05 AM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] SIP / GrandStream Configuration > > > 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 > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP / GrandStream Configuration
How will the packets get to the asterisk server? You'd need to forward ports on the NAT device, otherwise it's impossible - Original Message - From: "Uriel Carrasquilla" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Thursday, September 25, 2003 9:48 AM Subject: RE: [Asterisk-Users] SIP / GrandStream Configuration > Very valuable help. It is now working like a champ. > > This is a solution with SIP--NAT---Internet---Asterisk. No problems here. > > What I would like to do next is to move Asterisk behind a NAT as follows > SIP---NAT---Internet---NAT---Asterisk > do I need a STUN server? is there a chance this could work? > The Google results seems to indicate that I will get an ulcer attempting > this step. is that true? > > Regards, > Uriel > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of WipeOut . > Sent: Wednesday, September 24, 2003 9:05 AM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] SIP / GrandStream Configuration > > > 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 > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP / GrandStream Configuration
Very valuable help. It is now working like a champ. This is a solution with SIP--NAT---Internet---Asterisk. No problems here. What I would like to do next is to move Asterisk behind a NAT as follows SIP---NAT---Internet---NAT---Asterisk do I need a STUN server? is there a chance this could work? The Google results seems to indicate that I will get an ulcer attempting this step. is that true? Regards, Uriel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of WipeOut . Sent: Wednesday, September 24, 2003 9:05 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] SIP / GrandStream Configuration 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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP / GrandStream Configuration
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 <[EMAIL PROTECTED]> Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP / GrandStream Configuration
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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP / GrandStream Configuration
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
