[Asterisk-Users] Nat, SIP, Realtime problem
Group: I have a customer that is running the following Asterisk CVS-HEAD dated 2005-08-18 WhitBox Linux respin 2 mysql Ver 11.18 Distrib 3.23.58 Cisco 7960G We are using the real-time drivers for sip and everything is working great. They have a few employees that use the phones from home on a RR or DSL line. The problem is if they make a call everything works great they hang up and are able to get inbound calls. If they do not make a call for 5 or 10 mins they are unable to get inbound calls. If they dial out again its all working for another 5 or 10 mins. This does not happen to all remote people just a few. Anyone have any ideas what the heck is going on with this? Thanks for your time. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Nat, SIP, Realtime problem
Hall, Eric M. wrote: Asterisk CVS-HEAD dated 2005-08-18 WhitBox Linux respin 2 mysql Ver 11.18 Distrib 3.23.58 Cisco 7960G We are using the real-time drivers for sip and everything is working great. They have a few employees that use the phones from home on a RR or DSL line. The problem is if they make a call everything works great they hang up and are able to get inbound calls. If they do not make a call for 5 or 10 mins they are unable to get inbound calls. If they dial out again its all working for another 5 or 10 mins. This does not happen to all remote people just a few. Using Realtime SIP peers does not allow for NAT Keepalive packets to be sent, so the firewall/NAT devices that those phones are connected to are closing the SIP port hole after an expiration timeout. To fix this, you'll need to upgrade to newer Asterisk (you really should be running 1.2) and use 'realtime caching' for your SIP peers. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Nat, SIP, Realtime problem
I'm using realtime caching. Here is my sip.conf file [general] callerid=unavailable context=default allowguest=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes nat=yes canreinvite=no rtcachefriends=yes allow=ulaw allow=g729 All other information about the sip clint is keep in the db Thanks again! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin P. Fleming Sent: Tuesday, February 14, 2006 11:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Nat, SIP, Realtime problem Hall, Eric M. wrote: Asterisk CVS-HEAD dated 2005-08-18 WhitBox Linux respin 2 mysql Ver 11.18 Distrib 3.23.58 Cisco 7960G We are using the real-time drivers for sip and everything is working great. They have a few employees that use the phones from home on a RR or DSL line. The problem is if they make a call everything works great they hang up and are able to get inbound calls. If they do not make a call for 5 or 10 mins they are unable to get inbound calls. If they dial out again its all working for another 5 or 10 mins. This does not happen to all remote people just a few. Using Realtime SIP peers does not allow for NAT Keepalive packets to be sent, so the firewall/NAT devices that those phones are connected to are closing the SIP port hole after an expiration timeout. To fix this, you'll need to upgrade to newer Asterisk (you really should be running 1.2) and use 'realtime caching' for your SIP peers. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Nat, SIP, Realtime problem
Just wanted to also say this does not happen to all users behind a NAT box on RR or DSL line just a few. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin P. Fleming Sent: Tuesday, February 14, 2006 11:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Nat, SIP, Realtime problem Hall, Eric M. wrote: Asterisk CVS-HEAD dated 2005-08-18 WhitBox Linux respin 2 mysql Ver 11.18 Distrib 3.23.58 Cisco 7960G We are using the real-time drivers for sip and everything is working great. They have a few employees that use the phones from home on a RR or DSL line. The problem is if they make a call everything works great they hang up and are able to get inbound calls. If they do not make a call for 5 or 10 mins they are unable to get inbound calls. If they dial out again its all working for another 5 or 10 mins. This does not happen to all remote people just a few. Using Realtime SIP peers does not allow for NAT Keepalive packets to be sent, so the firewall/NAT devices that those phones are connected to are closing the SIP port hole after an expiration timeout. To fix this, you'll need to upgrade to newer Asterisk (you really should be running 1.2) and use 'realtime caching' for your SIP peers. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Nat Sip Pain
Hi everyone, I decided to have a look at SIP NAT again and I've been at it for a [quite a] few hours but typically nothing is working for me. Actually I'm not sure if SIP and NAT can ever work but some emails on this list do suggest that someone has got it working, once, maybe. I'm experimenting with a ZyXEL 2000W [WiFi Sip phone] which supports Outbound Proxy, STUN and Fake WAN Address on SIP and RTP. I'm using Netfilter (IPTables) on Linux as the Firewall at NAT gateway to the Internet. I'm lacking knowledge in UDP, RTP and SIP - which doesn't help of course. In my experiments the only thing that seems to allow me to make a call is to enter the [public Internet] IP address of my * server into the Outbound Proxy setting in the SIP phone - then it registers and I can make a call but no audio, either direction, is heard. I would have thought that the Outbound Proxy should be inside the NAT gateway but then I read the settings for a Budgetone BEHIND nat on the FWD webpage (http://www.freeworlddialup.com/support/configuration_guide/configure_your_fwd_certified_phone/grandstream_budgetone/outbound_proxy) where they suggest that the Outbound Proxy should be an external Internet public proxy server ? Then I was reading about STUN and what a nice sounding solution it is - so I downloaded and installed the Vivida STUN server - compilation installation was nice and easy and I set the STUN primary IP address port into the SIP phones STUN servers settings. I could see that the SIP phone communicated with the STUN server (lots of stuff about mapping between my local NAT gateway's public IP address and the secondary IP address of the STUN server)... but no registration or [apparent] communication with the * server. I didn't try to do anything with the Fake WAN address.. settings or try to redirect incoming UDP ports from the firewall to the SIP phone because I'm trying to see if its possible to setup a deploy-anywhere SIP phone solution. Needless to say when I don't have any NAT settings on the SIP phone I don't get any registration with the * server (this confuses me too - I'm not sure why I only get registration when I set the * server to be the outbound proxy? Maybe its because the SIP phone sends its local IP in the RTP packets?). Does anyone know how to get NAT SIP working where the SIP phone is behind a NAT server talking to a publicly accessible * server? Thanks for any help! When I run FWD's netcheck on my local PC (also behind the NAT) I get: Internet Connection: Connected, Direct/NAT: Using NAT, NAT type: Port Restricted Nat, NAT UPnP enabled: No, Local IP Address: 192.168.5.10, WAN IP Address: XXX.XXX.XXX.XXX (public IP address), Port 5060: Blocked, port 5082: Blocked. [Maybe] useful Links that I've found on my Nat SIP travels:- http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions - Here VOIP INFO claim that Asterisk as a SIP server outside nat, clients on the inside connecting to Asterisk is solved with with nat tiki-index.php?page=Asterisk+sip+nat=yes and qualify tiki-index.php?page=Asterisk+sip+qualify=xxx in sip.conf tiki-index.php?page=Asterisk+config+sip.conf for the client in most cases. Some clients (X-lite) assist themselves by using STUN tiki-index.php?page=STUN and sending UDP keep-alive packets. Qualify tiki-index.php?page=Asterisk+sip+qualify sends keep-alive packets from Asterisk to the client on the inside. - however I can't get it to work http://www.asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asterisk.html --- Here there is some detail about the NAT= option in sip.conf and firewall NAT types plus some understandable diagrams of why SIP NAT is so much bother. http://www.voip-info.org/wiki-STUN -- The VOIP INFO page about STUN - I don't think I learned much here - except the link to the Vovida STUN server software Asterisk Users - Email from [EMAIL PROTECTED] - 02/July/2005 23:49 Thierry claims that you need to put special MASQUERADE POSTROUTING rules into iptables to make it NAT UDP properly - tried it but didn't work for me Asterisk Users - Email from [EMAIL PROTECTED] - 16/Aug/2005 10:29 Kamran Ahmad sounds like someone who [might have] had SIP NAT working - until it wasn't working BTW My Current SIP sip.conf entry that I'm using for testing (which doesn't work of course!): - [0035314401789] context=PublicSip type=friend port=5060 username=0035314401789 password= callerId=0035314401789 nat=route; assume a NAT connection (note: route doesn't seem to make any difference compared to yes) qualify=yes; keep-alive
Re: [Asterisk-Users] Nat Sip Pain
Mensaje citado por: Derek Conniffe [EMAIL PROTECTED]: Hi, Does anyone know how to get NAT SIP working where the SIP phone is behind a NAT server talking to a publicly accessible * server? Have you tried sip-conntrack-nat for netfilter?. May be could help you. Get pom-ng from www.netfilter.org. Cheers. __ Registrate desde http://servicios.arnet.com.ar/registracion/registracion.asp?origenid=9 y participá de todos los beneficios del Portal Arnet. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Nat Sip Pain
Derek, You said - Needless to say when I don't have any NAT settings on the SIP phone I don't get any registration with the * server (this confuses me too - I'm not sure why I only get registration when I set the * server to be the outbound proxy? Maybe its because the SIP phone sends its local IP in the RTP packets?). SIP is not NAT friendly (unlike IAX) and yes your device will try to send its local IP (in SIP packets), unless in the case of a budgetone phone you set the 'Use NAT IP' to your external IP addr. You will also have to NAT the public ip for the SIP port (5060?) and RTP ports (whatever) to your phones private IP. Must admit not tried it myself, but happy to jointly experiment if you like? ___ Ray ___ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Derek Conniffe Sent: 13 September 2005 12:44 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Nat Sip Pain Hi everyone, I decided to have a look at SIP NAT again and I've been at it for a [quite a] few hours but typically nothing is working for me. Actually I'm not sure if SIP and NAT can ever work but some emails on this list do suggest that someone has got it working, once, maybe. I'm experimenting with a ZyXEL 2000W [WiFi Sip phone] which supports Outbound Proxy, STUN and Fake WAN Address on SIP and RTP. I'm using Netfilter (IPTables) on Linux as the Firewall at NAT gateway to the Internet. I'm lacking knowledge in UDP, RTP and SIP - which doesn't help of course. In my experiments the only thing that seems to allow me to make a call is to enter the [public Internet] IP address of my * server into the Outbound Proxy setting in the SIP phone - then it registers and I can make a call but no audio, either direction, is heard. I would have thought that the Outbound Proxy should be inside the NAT gateway but then I read the settings for a Budgetone BEHIND nat on the FWD webpage (http://www.freeworlddialup.com/support/configuration_guide/configure_yo ur_fwd_certified_phone/grandstream_budgetone/outbound_proxy) where they suggest that the Outbound Proxy should be an external Internet public proxy server ? Then I was reading about STUN and what a nice sounding solution it is - so I downloaded and installed the Vivida STUN server - compilation installation was nice and easy and I set the STUN primary IP address port into the SIP phones STUN servers settings. I could see that the SIP phone communicated with the STUN server (lots of stuff about mapping between my local NAT gateway's public IP address and the secondary IP address of the STUN server)... but no registration or [apparent] communication with the * server. I didn't try to do anything with the Fake WAN address.. settings or try to redirect incoming UDP ports from the firewall to the SIP phone because I'm trying to see if its possible to setup a deploy-anywhere SIP phone solution. Needless to say when I don't have any NAT settings on the SIP phone I don't get any registration with the * server (this confuses me too - I'm not sure why I only get registration when I set the * server to be the outbound proxy? Maybe its because the SIP phone sends its local IP in the RTP packets?). Does anyone know how to get NAT SIP working where the SIP phone is behind a NAT server talking to a publicly accessible * server? Thanks for any help! When I run FWD's netcheck on my local PC (also behind the NAT) I get: Internet Connection: Connected, Direct/NAT: Using NAT, NAT type: Port Restricted Nat, NAT UPnP enabled: No, Local IP Address: 192.168.5.10, WAN IP Address: XXX.XXX.XXX.XXX (public IP address), Port 5060: Blocked, port 5082: Blocked. [Maybe] useful Links that I've found on my Nat SIP travels:- http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions - Here VOIP INFO claim that Asterisk as a SIP server outside nat, clients on the inside connecting to Asterisk is solved with with nat tiki-index.php?page=Asterisk+sip+nat=yes and qualify tiki-index.php?page=Asterisk+sip+qualify=xxx in sip.conf tiki-index.php?page=Asterisk+config+sip.conf for the client in most cases. Some clients (X-lite) assist themselves by using STUN tiki-index.php?page=STUN and sending UDP keep-alive packets. Qualify tiki-index.php?page=Asterisk+sip+qualify sends keep-alive packets from Asterisk to the client on the inside. - however I can't get it to work http://www.asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asteris k.html --- Here there is some detail about the NAT= option in sip.conf and firewall NAT types plus some understandable diagrams of why SIP NAT is so much bother. http://www.voip-info.org/wiki-STUN
Re: [Asterisk-Users] Nat Sip Pain
Hi Ray, It would be great to find a solution which doesn't need modification of the firewall setup (like if it was a customers firewall rather than your own). There is two things I'm wondering about: - 1) Can a Outbound SIP Proxy be a server out on the Internet (i.e. not in the local network this side of the NAT) and does that provide a way to make the SIP via NAT work? * 2) Is STUN a workable solution. There is no problem running a STUN server but can the far side of the STUN connection (Internet) talk with Asterisk and is this a way to make the SIP via NAT work? ** * I would have thought that an Outbound Proxy would need to be inside on the local network (a bastion host rather like a squid server for HTTP) but then I read the FWD documentation about setting the Outbound Proxy for a budgetone to make it work with NAT and their server - the Outbound Proxy they specified was out there on the Internet. ** I've read that Asterisk doesn't currently have STUN support but I'm not sure what that means exactly: I'm not sure if that means Asterisk doesn't have an STUN server built-in or if it means Asterisk is not compatible with an STUN server. Thanks, Derek razza wrote: Derek, You said - Needless to say when I don't have any NAT settings on the SIP phone I don't get any registration with the * server (this confuses me too - I'm not sure why I only get registration when I set the * server to be the outbound proxy? Maybe its because the SIP phone sends its local IP in the RTP packets?). SIP is not NAT friendly (unlike IAX) and yes your device will try to send its local IP (in SIP packets), unless in the case of a budgetone phone you set the 'Use NAT IP' to your external IP addr. You will also have to NAT the public ip for the SIP port (5060?) and RTP ports (whatever) to your phones private IP. Must admit not tried it myself, but happy to jointly experiment if you like? ___ Ray ___ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Derek Conniffe Sent: 13 September 2005 12:44 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Nat Sip Pain Hi everyone, I decided to have a look at SIP NAT again and I've been at it for a [quite a] few hours but typically nothing is working for me. Actually I'm not sure if SIP and NAT can ever work but some emails on this list do suggest that someone has got it working, once, maybe. I'm experimenting with a ZyXEL 2000W [WiFi Sip phone] which supports Outbound Proxy, STUN and Fake WAN Address on SIP and RTP. I'm using Netfilter (IPTables) on Linux as the Firewall at NAT gateway to the Internet. I'm lacking knowledge in UDP, RTP and SIP - which doesn't help of course. In my experiments the only thing that seems to allow me to make a call is to enter the [public Internet] IP address of my * server into the Outbound Proxy setting in the SIP phone - then it registers and I can make a call but no audio, either direction, is heard. I would have thought that the Outbound Proxy should be inside the NAT gateway but then I read the settings for a Budgetone BEHIND nat on the FWD webpage (http://www.freeworlddialup.com/support/configuration_guide/configure_yo ur_fwd_certified_phone/grandstream_budgetone/outbound_proxy) where they suggest that the Outbound Proxy should be an external Internet public proxy server ? Then I was reading about STUN and what a nice sounding solution it is - so I downloaded and installed the Vivida STUN server - compilation installation was nice and easy and I set the STUN primary IP address port into the SIP phones STUN servers settings. I could see that the SIP phone communicated with the STUN server (lots of stuff about mapping between my local NAT gateway's public IP address and the secondary IP address of the STUN server)... but no registration or [apparent] communication with the * server. I didn't try to do anything with the Fake WAN address.. settings or try to redirect incoming UDP ports from the firewall to the SIP phone because I'm trying to see if its possible to setup a deploy-anywhere SIP phone solution. Needless to say when I don't have any NAT settings on the SIP phone I don't get any registration with the * server (this confuses me too - I'm not sure why I only get registration when I set the * server to be the outbound proxy? Maybe its because the SIP phone sends its local IP in the RTP packets?). Does anyone know how to get NAT SIP working where the SIP phone is behind a NAT server talking to a publicly accessible * server? Thanks for any help! When I run FWD's netcheck on my local PC (also behind the NAT) I get: Internet Connection: Connected, Direct/NAT: Using NAT, NAT type: Port Restricted Nat, NAT UPnP enabled: No, Local IP Address: 192.168.5.10, WAN IP Address: XXX.XXX.XXX.XXX
Re: FW: [Asterisk-Users] Nat Sip Pain
Hi Ray, I was wondering if the qualify option is used [in sip.conf] to keep a connection (from the SIP phone inside the firewall to the Asterisk server outside the firewall) open then would the firewall not allow two way communication without incoming port mapping/NAT (providing that the SIP phone started talking first)? I'm not sure about that - I'm being hopeful though :) STUN would be very acceptable to me if it worked though ;) Derek razza wrote: Derek, I'm not an expert in these area's hence the offer to play, but in answer to your questions to the best of my ability - 1. I don't see any reason the outbound proxy cant be in the public domain although this is where the NAT issues start kicking in (especially if you want incoming calls), depending on the number of clients behind the firewall you would have to do lots of port mapping etc. on the router/firewall, could be done but would be painful. 2. Never played with a STUN server, sorry just another point to break in the chain? ___ Ray ___ -Original Message- From: Derek Conniffe [mailto:[EMAIL PROTECTED] Sent: 13 September 2005 17:50 To: Asterisk Users Mailing List - Non-Commercial Discussion; [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Nat Sip Pain Hi Ray, It would be great to find a solution which doesn't need modification of the firewall setup (like if it was a customers firewall rather than your own). There is two things I'm wondering about: - 1) Can a Outbound SIP Proxy be a server out on the Internet (i.e. not in the local network this side of the NAT) and does that provide a way to make the SIP via NAT work? * 2) Is STUN a workable solution. There is no problem running a STUN server but can the far side of the STUN connection (Internet) talk with Asterisk and is this a way to make the SIP via NAT work? ** * I would have thought that an Outbound Proxy would need to be inside on the local network (a bastion host rather like a squid server for HTTP) but then I read the FWD documentation about setting the Outbound Proxy for a budgetone to make it work with NAT and their server - the Outbound Proxy they specified was out there on the Internet. ** I've read that Asterisk doesn't currently have STUN support but I'm not sure what that means exactly: I'm not sure if that means Asterisk doesn't have an STUN server built-in or if it means Asterisk is not compatible with an STUN server. Thanks, Derek razza wrote: Derek, You said - Needless to say when I don't have any NAT settings on the SIP phone I don't get any registration with the * server (this confuses me too - I'm not sure why I only get registration when I set the * server to be the outbound proxy? Maybe its because the SIP phone sends its local IP in the RTP packets?). SIP is not NAT friendly (unlike IAX) and yes your device will try to send its local IP (in SIP packets), unless in the case of a budgetone phone you set the 'Use NAT IP' to your external IP addr. You will also have to NAT the public ip for the SIP port (5060?) and RTP ports (whatever) to your phones private IP. Must admit not tried it myself, but happy to jointly experiment if you like? ___ Ray ___ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Derek Conniffe Sent: 13 September 2005 12:44 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Nat Sip Pain Hi everyone, I decided to have a look at SIP NAT again and I've been at it for a [quite a] few hours but typically nothing is working for me. Actually I'm not sure if SIP and NAT can ever work but some emails on this list do suggest that someone has got it working, once, maybe. I'm experimenting with a ZyXEL 2000W [WiFi Sip phone] which supports Outbound Proxy, STUN and Fake WAN Address on SIP and RTP. I'm using Netfilter (IPTables) on Linux as the Firewall at NAT gateway to the Internet. I'm lacking knowledge in UDP, RTP and SIP - which doesn't help of course. In my experiments the only thing that seems to allow me to make a call is to enter the [public Internet] IP address of my * server into the Outbound Proxy setting in the SIP phone - then it registers and I can make a call but no audio, either direction, is heard. I would have thought that the Outbound Proxy should be inside the NAT gateway but then I read the settings for a Budgetone BEHIND nat on the FWD webpage (http://www.freeworlddialup.com/support/configuration_guide/configure_y o ur_fwd_certified_phone/grandstream_budgetone/outbound_proxy) where they suggest that the Outbound Proxy should be an external Internet public proxy server ? Then I was reading about STUN and what a nice sounding solution it is - so I downloaded
RE: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet)
--- Uriel Carrasquilla [EMAIL PROTECTED] wrote: John: are you aware of any documentation on how to configre SER to be a front-end to Asterisk? I suspect it is very inexpensive to put a SER server in a hosting facility I think the cost is about the same as for putting a web server at a hosting facility. But I don't think you need high bandwidth. SER simply sets up the call. I don't think the audio data actually goes through SER. It goes directly between the two end points. This is the big problem with using Asterisk for SIP. With Asterisk the audio data between two SIP extensions has to actualy go into then out of the Asterisk box. This does not scale well to thousands of users like in a university campus or a comercial SIP service. to forward traffic to multiple Asterisks based on Least Cost Routing. My problem is that my experience is with Asterisk and not with SER. Uriel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of John Todd Sent: Monday, October 13, 2003 8:11 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet) I'm curently looking into using SER to front end SIP calls for Asterisk. Basicaly all SIP users would register with SER not Asterisk and then Asterisk and SER exchange registrations. SER is a very capable SIP router, much more sophisticated than Asterisk as it can look inside packets and route based on what it finds or even re-write packets based on user specified logic. SER is GPL'd and has very good user documentation. Don't know how well the above will work. The claim by the authors or SER that it can handle thousands of calls per second is quite impressive One other nice feature is that SER users can set up their own SIP accounts using a web interface and not needing to edit *.conf files. See here for details http://www.iptel.org/ser/ = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK SER is an excellent option as a front end to Asterisk. It is a true SIP proxy, whereas Asterisk is a hybrid, and SIP has not been the primary focus of Asterisk development. In fact, Asterisk's SIP implementation is very limited (though it is extremely pragmatic.) However, moving to SER does not solve any of the issues about the proxy being behind a NAT, and I believe that SER will have the same problems (though I could be wrong on this; I haven't experimented with SER's ability to work from behind a NAT.) SIP clients work well enough behind NAT (most of them, anyway) but the servers are a different story. I really like SER's third-party addons for account administration; Asterisk is significantly more complex, and probably would not be as easily converted to such a front end. In fact, SER has a very complex routing/scripting language that is not easily administered with a web front end, so I think that SER and Asterisk suffer from the same problems. If someone were to come up with a simple way to administer voicemail.conf and sip.conf from a web tool, that would go far to making Asterisk a bit more user-accessible... JT ___ 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 = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? The New Yahoo! Shopping - with improved product search http://shopping.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet)
Chris Albertson wrote: This is the big problem with using Asterisk for SIP. With Asterisk the audio data between two SIP extensions has to actualy go into then out of the Asterisk box. This does not scale well to thousands of users like in a university campus or a comercial SIP service. http://www.voip-info.org/wiki-Asterisk+sip+reinvite As I understand this Asterisk sets up the call with itself as endpoints, then moves the stream tobypass the PBX and go directly with a SIP reinvite. Some clients does not support this, and with those you have to configure asterisk to stay in the media path for this client with canreinvite=no in SIP.conf. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet)
Uriel - 1) Please stop top-posting. 2) I'm afraid I don't have any data on specifics of creating a front-end. I know how to do it, but my time these days is spent writing lots of other projects that I have been doing. :-) I would suggest you get SER and set it up - it's quite easy, and the documentation on SER itself is very well written, and if you have a good idea of how SIP works you should be able to patch together an appropriate system. However, if you aren't 100% familiar with how SIP works, I would stick to just an Asterisk system; SER doesn't allow for any of the shortcuts that Asterisk has. 3) Use Google and do some searching. I found some quick links with a few of the keywords that would seem obvious, but I don't have enough time to review them... JT John: Thank you for responding. I am in the process of installing SER and hope to have it ready by this weekend. I am in the process of installing some equipment at a local colo. I have to tell you, at the expense of offending you, that I use MS-Outlook and the responses go to the tope of the messages. At work I use Lotus Notes and the same thing happens. Before, I used PROFS (on mainframes) and the same principle applied. All in all, 20+ years of using this principle for e-mails at both work and home. As a matter of fact, I am of the opinion that the response to E-mails should go at the top to save time. However, this is not about me but the * group and the well being of this list. Does anybody else have a strong opinion one way or the other? If it is left to John and myself we have a 1:1 vote. Regards, Uriel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet)
Andre:
This makes a lot of sense. I had used Asterisk in the past to play the role
of Gatekeeper for directing traffic to the appropriate Asterisk acting as a
PSTN gateway. IAX does a heck of a good job in that configuration.
However, with SIP, I have run into nothing but trouble with registrations
falling off.
I have read the SER manual I am going to jump into it, now that I know that
in practice it works and it is not only theory in a manual.
Thank you,
Uriel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Andres
Sent: Tuesday, October 14, 2003 12:43 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on
the Internet)
On Monday 13 October 2003 22:26, Uriel Carrasquilla wrote:
John:
are you aware of any documentation on how to configre SER to be a
front-end
to Asterisk?
Hi Uriel,
At TeleSIP we run a cluster of several geographically distributed SER
Servers
that hande all our SIP Routing. SER is a robust, fast and stable platform
which has worked flawlessly for us. We use * as our company PBX and PSTN
Gateway. Basically what you need to do is to device a numbering plan so
that
SERs routing logic can forward the call to * when it needs to.
For example in ser.cfg you could put something like this:
#
###PSTN ACCESS###
#
if (method==INVITE) {
if (uri=~sip:[EMAIL PROTECTED]) {
log(1, This is a Long Distance Call\n);
route(6);
break;
};
};
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Re: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet)
On 15/10/03 00:15, Uriel Carrasquilla wrote: Does anybody else have a strong opinion one way or the other? If it is left to John and myself we have a 1:1 vote. See how much easier it is to follow the thread of conversation if you quote just enough of the e-mail you're responding to so people know what's going on without having to read through pages of text? Please see RFC 1855: - http://www.faqs.org/rfcs/rfc1855.html Decent mail clients that behave sensibly regarding quoting are easy to come by. You can even set up Outlook to behave vaguely properly and quote using . As a matter of fact, I am of the opinion that the response to E-mails should go at the top to save time. So, it's not worth *your* time organizing your e-mail sensibly, but it's worth everyone else's time having to dig through lines of text to work out what the context is? I find that selfish, at best. Please see the following page (strong words warning). It pretty much sums it all up nicely: - http://thegestalt.org/simon/quoterant.html -- Al Maw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet)
OK OK OK, I got it. See my response inside the body of your E-mail. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Alastair Maw Sent: Tuesday, October 14, 2003 8:44 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet) On 15/10/03 00:15, Uriel Carrasquilla wrote: Does anybody else have a strong opinion one way or the other? If it is left to John and myself we have a 1:1 vote. See how much easier it is to follow the thread of conversation if you quote just enough of the e-mail you're responding to so people know what's going on without having to read through pages of text? [URIEL] - I have to learn how to quote with Outlook. Please see RFC 1855: - http://www.faqs.org/rfcs/rfc1855.html Decent mail clients that behave sensibly regarding quoting are easy to come by. You can even set up Outlook to behave vaguely properly and quote using . As a matter of fact, I am of the opinion that the response to E-mails should go at the top to save time. So, it's not worth *your* time organizing your e-mail sensibly, but it's worth everyone else's time having to dig through lines of text to work out what the context is? I find that selfish, at best. [URIEL] you are absolutely right and I do apologize. Ignorance is not an excuse. Please see the following page (strong words warning). It pretty much sums it all up nicely: - http://thegestalt.org/simon/quoterant.html [URIEL] Thank you. -- Al Maw ___ 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] NAT, SIP
I wish you would take this stuff to personal email, I am tired of wasting my time reading this crap. If you idiots want to give lesions on how YOU would like people to post on list servers _DO_IT_VIA_PERSONAL_EMAIL_!!! None of the rest of us care. This is a personal messages from you to someone else. Stop wastering MY bandwidth. I personal like top posting as I don't have to scroll all the way to the bottom to read what is most of the time one damn sentence. So, it's not worth *your* time organizing your e-mail sensibly, but it's worth everyone else's time having to dig through lines of text to work out what the context is? I find that selfish, at best. Please see the following page (strong words warning). It pretty much sums it all up nicely: - http://thegestalt.org/simon/quoterant.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet)
I'm curently looking into using SER to front end SIP calls for Asterisk. Basicaly all SIP users would register with SER not Asterisk and then Asterisk and SER exchange registrations. SER is a very capable SIP router, much more sophisticated than Asterisk as it can look inside packets and route based on what it finds or even re-write packets based on user specified logic. SER is GPL'd and has very good user documentation. Don't know how well the above will work. The claim by the authors or SER that it can handle thousands of calls per second is quite impressive One other nice feature is that SER users can set up their own SIP accounts using a web interface and not needing to edit *.conf files. See here for details http://www.iptel.org/ser/ = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? The New Yahoo! Shopping - with improved product search http://shopping.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet)
I'm curently looking into using SER to front end SIP calls for Asterisk. Basicaly all SIP users would register with SER not Asterisk and then Asterisk and SER exchange registrations. SER is a very capable SIP router, much more sophisticated than Asterisk as it can look inside packets and route based on what it finds or even re-write packets based on user specified logic. SER is GPL'd and has very good user documentation. Don't know how well the above will work. The claim by the authors or SER that it can handle thousands of calls per second is quite impressive One other nice feature is that SER users can set up their own SIP accounts using a web interface and not needing to edit *.conf files. See here for details http://www.iptel.org/ser/ = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK SER is an excellent option as a front end to Asterisk. It is a true SIP proxy, whereas Asterisk is a hybrid, and SIP has not been the primary focus of Asterisk development. In fact, Asterisk's SIP implementation is very limited (though it is extremely pragmatic.) However, moving to SER does not solve any of the issues about the proxy being behind a NAT, and I believe that SER will have the same problems (though I could be wrong on this; I haven't experimented with SER's ability to work from behind a NAT.) SIP clients work well enough behind NAT (most of them, anyway) but the servers are a different story. I really like SER's third-party addons for account administration; Asterisk is significantly more complex, and probably would not be as easily converted to such a front end. In fact, SER has a very complex routing/scripting language that is not easily administered with a web front end, so I think that SER and Asterisk suffer from the same problems. If someone were to come up with a simple way to administer voicemail.conf and sip.conf from a web tool, that would go far to making Asterisk a bit more user-accessible... JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet)
On 13-10 17:11, John Todd wrote: [...] SER is an excellent option as a front end to Asterisk. It is a true SIP proxy, whereas Asterisk is a hybrid, and SIP has not been the primary focus of Asterisk development. In fact, Asterisk's SIP implementation is very limited (though it is extremely pragmatic.) However, moving to SER does not solve any of the issues about the proxy being behind a NAT, and I believe that SER will have the same problems (though I could be wrong on this; I haven't experimented with SER's ability to work from behind a NAT.) SIP clients work well enough behind NAT (most of them, anyway) but the servers are a different story. SER can can become very helpful when it is run in the public internet and clients are behind NATs. For this case SER contains many NAT helping functions that can rewrite header fields, test if a client comes from behind a NAT, ping clients behind NATs (to keep the NAT binding open) and force RTP proxy usage when necesary. Along with RTP proxy SER can help any *symmetric* SIP user agent to get through NAT. (A symmetric SIP user agent is a user agent that uses the same source port for receiving signalling and media as for sending them. Vast majority of SIP user agents as of today is symmetric, including Windows Messenger, Cisco phones, Grandstream phone a.s.o.). There is also support for proxy behind NAT, but it is mostly untested yet. Jan. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet)
Chris: I am glad to see someone else asking the same question I have been asking myself. As soon as I get my public IP address, I will install SER on the public side and Asterisk behind a NAT (with dynamic IP) to see if I can get around problems I have when my SIP (UA) behind their own NAT on the other side of my Internet connection. If you make any progress, please share. I will do the same. Uriel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Chris Albertson Sent: Monday, October 13, 2003 7:49 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet) I'm curently looking into using SER to front end SIP calls for Asterisk. Basicaly all SIP users would register with SER not Asterisk and then Asterisk and SER exchange registrations. SER is a very capable SIP router, much more sophisticated than Asterisk as it can look inside packets and route based on what it finds or even re-write packets based on user specified logic. SER is GPL'd and has very good user documentation. Don't know how well the above will work. The claim by the authors or SER that it can handle thousands of calls per second is quite impressive One other nice feature is that SER users can set up their own SIP accounts using a web interface and not needing to edit *.conf files. See here for details http://www.iptel.org/ser/ = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? The New Yahoo! Shopping - with improved product search http://shopping.yahoo.com ___ 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] NAT, SIP (was: No sound with SIP Phones on the Internet)
John: are you aware of any documentation on how to configre SER to be a front-end to Asterisk? I suspect it is very inexpensive to put a SER server in a hosting facility to forward traffic to multiple Asterisks based on Least Cost Routing. My problem is that my experience is with Asterisk and not with SER. Uriel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of John Todd Sent: Monday, October 13, 2003 8:11 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet) I'm curently looking into using SER to front end SIP calls for Asterisk. Basicaly all SIP users would register with SER not Asterisk and then Asterisk and SER exchange registrations. SER is a very capable SIP router, much more sophisticated than Asterisk as it can look inside packets and route based on what it finds or even re-write packets based on user specified logic. SER is GPL'd and has very good user documentation. Don't know how well the above will work. The claim by the authors or SER that it can handle thousands of calls per second is quite impressive One other nice feature is that SER users can set up their own SIP accounts using a web interface and not needing to edit *.conf files. See here for details http://www.iptel.org/ser/ = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK SER is an excellent option as a front end to Asterisk. It is a true SIP proxy, whereas Asterisk is a hybrid, and SIP has not been the primary focus of Asterisk development. In fact, Asterisk's SIP implementation is very limited (though it is extremely pragmatic.) However, moving to SER does not solve any of the issues about the proxy being behind a NAT, and I believe that SER will have the same problems (though I could be wrong on this; I haven't experimented with SER's ability to work from behind a NAT.) SIP clients work well enough behind NAT (most of them, anyway) but the servers are a different story. I really like SER's third-party addons for account administration; Asterisk is significantly more complex, and probably would not be as easily converted to such a front end. In fact, SER has a very complex routing/scripting language that is not easily administered with a web front end, so I think that SER and Asterisk suffer from the same problems. If someone were to come up with a simple way to administer voicemail.conf and sip.conf from a web tool, that would go far to making Asterisk a bit more user-accessible... JT ___ 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] NAT, SIP (was: No sound with SIP Phones on the Internet)
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of John Todd Sent: Monday, October 13, 2003 8:11 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet) I'm curently looking into using SER to front end SIP calls for Asterisk. Basicaly all SIP users would register with SER not Asterisk and then Asterisk and SER exchange registrations. SER is a very capable SIP router, much more sophisticated than Asterisk as it can look inside packets and route based on what it finds or even re-write packets based on user specified logic. SER is GPL'd and has very good user documentation. Don't know how well the above will work. The claim by the authors or SER that it can handle thousands of calls per second is quite impressive One other nice feature is that SER users can set up their own SIP accounts using a web interface and not needing to edit *.conf files. See here for details http://www.iptel.org/ser/ = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK SER is an excellent option as a front end to Asterisk. It is a true SIP proxy, whereas Asterisk is a hybrid, and SIP has not been the primary focus of Asterisk development. In fact, Asterisk's SIP implementation is very limited (though it is extremely pragmatic.) However, moving to SER does not solve any of the issues about the proxy being behind a NAT, and I believe that SER will have the same problems (though I could be wrong on this; I haven't experimented with SER's ability to work from behind a NAT.) SIP clients work well enough behind NAT (most of them, anyway) but the servers are a different story. I really like SER's third-party addons for account administration; Asterisk is significantly more complex, and probably would not be as easily converted to such a front end. In fact, SER has a very complex routing/scripting language that is not easily administered with a web front end, so I think that SER and Asterisk suffer from the same problems. If someone were to come up with a simple way to administer voicemail.conf and sip.conf from a web tool, that would go far to making Asterisk a bit more user-accessible... JT At 11:26 PM -0400 10/13/03, Uriel Carrasquilla wrote: From: Uriel Carrasquilla [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet) Reply-To: [EMAIL PROTECTED] Date: Mon, 13 Oct 2003 23:26:59 -0400 John: are you aware of any documentation on how to configre SER to be a front-end to Asterisk? I suspect it is very inexpensive to put a SER server in a hosting facility to forward traffic to multiple Asterisks based on Least Cost Routing. My problem is that my experience is with Asterisk and not with SER. Uriel Uriel - 1) Please stop top-posting. 2) I'm afraid I don't have any data on specifics of creating a front-end. I know how to do it, but my time these days is spent writing lots of other projects that I have been doing. :-) I would suggest you get SER and set it up - it's quite easy, and the documentation on SER itself is very well written, and if you have a good idea of how SIP works you should be able to patch together an appropriate system. However, if you aren't 100% familiar with how SIP works, I would stick to just an Asterisk system; SER doesn't allow for any of the shortcuts that Asterisk has. 3) Use Google and do some searching. I found some quick links with a few of the keywords that would seem obvious, but I don't have enough time to review them... JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet)
On Monday 13 October 2003 22:26, Uriel Carrasquilla wrote:
John:
are you aware of any documentation on how to configre SER to be a front-end
to Asterisk?
Hi Uriel,
At TeleSIP we run a cluster of several geographically distributed SER Servers
that hande all our SIP Routing. SER is a robust, fast and stable platform
which has worked flawlessly for us. We use * as our company PBX and PSTN
Gateway. Basically what you need to do is to device a numbering plan so that
SERs routing logic can forward the call to * when it needs to.
For example in ser.cfg you could put something like this:
#
###PSTN ACCESS###
#
if (method==INVITE) {
if (uri=~sip:[EMAIL PROTECTED]) {
log(1, This is a Long Distance Call\n);
route(6);
break;
};
};
.
.
.
route[6] {
rewritehostport(your_asterisk_box:5050);
if (!t_relay()) {
sl_reply_error();
};
}
Andres
http://www.telesip.net
I suspect it is very inexpensive to put a SER server in a hosting facility
to forward traffic to multiple Asterisks based on Least Cost Routing.
My problem is that my experience is with Asterisk and not with SER.
Uriel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of John Todd
Sent: Monday, October 13, 2003 8:11 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on
the Internet)
I'm curently looking into using SER to front end SIP calls for
Asterisk.
Basicaly all SIP users would register with SER not Asterisk and then
Asterisk and SER exchange registrations.
SER is a very capable SIP router, much more sophisticated than Asterisk
as it can look inside packets and route based on what it finds or even
re-write packets based on user specified logic.
SER is GPL'd and has very good user documentation. Don't know how well
the above will work. The claim by the authors or SER that it can
handle thousands of calls per second is quite impressive
One other nice feature is that SER users can set up their own SIP
accounts using a web interface and not needing to edit *.conf files.
See here for details http://www.iptel.org/ser/
=
Chris Albertson
Home: 310-376-1029 [EMAIL PROTECTED]
Cell: 310-990-7550
Office: 310-336-5189 [EMAIL PROTECTED]
KG6OMK
SER is an excellent option as a front end to Asterisk. It is a
true SIP proxy, whereas Asterisk is a hybrid, and SIP has not been
the primary focus of Asterisk development. In fact, Asterisk's SIP
implementation is very limited (though it is extremely pragmatic.)
However, moving to SER does not solve any of the issues about the
proxy being behind a NAT, and I believe that SER will have the same
problems (though I could be wrong on this; I haven't experimented
with SER's ability to work from behind a NAT.) SIP clients work
well enough behind NAT (most of them, anyway) but the servers are a
different story.
I really like SER's third-party addons for account administration;
Asterisk is significantly more complex, and probably would not be as
easily converted to such a front end. In fact, SER has a very
complex routing/scripting language that is not easily administered
with a web front end, so I think that SER and Asterisk suffer from
the same problems. If someone were to come up with a simple way to
administer voicemail.conf and sip.conf from a web tool, that would go
far to making Asterisk a bit more user-accessible...
JT
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[Asterisk-Users] NAT/SIP solution?
Greetings, I was wondering if somebody is working on a solution to the NAT/SIP-issues? It seems to me that the problem has been identified, is that correct? Just hoping that someone with more skills will provide us with a solution sooner or later... Regards, Stig
Re: [Asterisk-Users] NAT/SIP solution?
Stig Hess wrote: Greetings, I was wondering if somebody is working on a solution to the NAT/SIP-issues? It seems to me that the problem has been identified, is that correct? Just hoping that someone with more skills will provide us with a solution sooner or later... Regards, Stig The solution is to use nat=yes in your sip.conf.. so far this has worked great for me.. Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NAT/SIP solution?
I meant where Asterisk is behing a NAT... sorry for the confusion. Regards, Stig H. - Original Message - From: WipeOut To: [EMAIL PROTECTED] Sent: Sunday, September 28, 2003 8:21 PM Subject: Re: [Asterisk-Users] NAT/SIP solution? Stig Hess wrote: Greetings, I was wondering if somebody is working on a solution to the NAT/SIP-issues? It seems to me that the problem has been identified, is that correct? Just hoping that someone with more skills will provide us with a solution sooner or later... Regards, StigThe solution is to use "nat=yes" in your sip.conf.. so far this has worked great for me..Later..___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NAT/SIP solution?
Stig Hess wrote: I meant where Asterisk is behing a NAT... sorry for the confusion. Regards, Stig H. Oh.. :) Well thats a bigger problem.. and i doubt the Gods of SIP are going to fix it any time soon.. :( Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NAT/SIP solution?
I've heard of this: http://sarp.sourceforge.net I have no idea if it will be of any help, but it's an interesting programme for handleing SIP over NAT. Brian. - Original Message - From: Stig Hess To: [EMAIL PROTECTED] Sent: Sunday, September 28, 2003 2:17 PM Subject: [Asterisk-Users] NAT/SIP solution? Greetings, I was wondering if somebody is working on a solution to the NAT/SIP-issues? It seems to me that the problem has been identified, is that correct? Just hoping that someone with more skills will provide us with a solution sooner or later... Regards, Stig
Re: [Asterisk-Users] NAT/SIP solution?
Stig Hess wrote: I meant where Asterisk is behing a NAT... sorry for the confusion. Hi Stig, If you are able to run * on your NAT'd box, then I have come up with a work around (thanks wasim!!!) that will allow you to run an * box behind your NAT, and still recieve and make SIP calls. I haven't got the whole thing figured out yet in terms of extensions.conf (but I am working on that today, will post on my website later) but this is basically my configuration: remote --TDM400P-- * --IAX-- * --SIP-- remote |-NAT-||FW| So basically the * on the GW machine which is also the NAT / FW box recieves the connection from the SIP remote end, then forwards all the traffic over IAX to the NAT'd * box. I just tested it, and it works fine! Once I get some more complex extensions.conf files setup, I will post them. Thanks, Leif Madsen. BTW: As for just passing SIP through SIP, I believe it's a limitation of the SIP protocol as the RTP ports are different than the connection port, whereas IAX is all the same port for everything (from what I gather) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] NAT/SIP solution?
I might be putting words into Stig's message but I think what he means to ask was the following scenario that causes problems: SIP --- NAT --- Internet --- NAT --- Asterisk Nikotel has a solution and one participant in thi list is doing a trial on a SIP/NAT router (claiming to be the first one in this realm). To answer Stig's question as I understand it: I don't think anybody is working on a solution in this list since the by-pass is to put Asterisk directly on the Internet with its own public-IP address. Regards, Uriel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of WipeOut Sent: Sunday, September 28, 2003 2:21 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] NAT/SIP solution? Stig Hess wrote: Greetings, I was wondering if somebody is working on a solution to the NAT/SIP-issues? It seems to me that the problem has been identified, is that correct? Just hoping that someone with more skills will provide us with a solution sooner or later... Regards, Stig The solution is to use nat=yes in your sip.conf.. so far this has worked great for me.. Later.. ___ 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] NAT/SIP solution?
Greetings, I was wondering if somebody is working on a solution to the NAT/SIP-issues? It seems to me that the problem has been identified, is that correct? Just hoping that someone with more skills will provide us with a solution sooner or later... Regards, Stig http://bugs.digium.com/bug_view_page.php?bug_id=104 This has been identified and is waiting for a solution. Feel free to contribute a patch! JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] NAT/SIP solution?
There are several SIP aware NAT routers. Any Cisco router with a firewall load has SIP aware NAT. There is at least one other brand of SIP aware NAT router out there, but I don't recall the brand. On Sun, 2003-09-28 at 17:26, Uriel Carrasquilla wrote: Nikotel has a solution and one participant in thi list is doing a trial on a SIP/NAT router (claiming to be the first one in this realm). -- Sample configs and more: http://www.fnords.org/~eric/asterisk/ BTEL Consulting +1-850-484-4535 x2111 (Pensacola) +1-504-595-3916 x2111 (New Orleans) +1-877-677-9643 x2111 (Toll Free) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] NAT/SIP solution?
On Mon, 2003-09-29 at 00:00, Eric Wieling wrote: There are several SIP aware NAT routers. Any Cisco router with a firewall load has SIP aware NAT. There is at least one other brand of SIP aware NAT router out there, but I don't recall the brand. On Sun, 2003-09-28 at 17:26, Uriel Carrasquilla wrote: Nikotel has a solution and one participant in thi list is doing a trial on a SIP/NAT router (claiming to be the first one in this realm). http://www.intertex.se -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] NAT/SIP solution?
I had been warned about British sense of humour, but this even a South American like myself find funny. Uriel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of WipeOut Sent: Sunday, September 28, 2003 3:21 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] NAT/SIP solution? Stig Hess wrote: I meant where Asterisk is behing a NAT... sorry for the confusion. Regards, Stig H. Oh.. :) Well thats a bigger problem.. and i doubt the Gods of SIP are going to fix it any time soon.. :( Later.. ___ 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
