Re: [Asterisk-Users] Re: DNS SRV records
On 08-06 18:06, Duane wrote: Darren Edmundson wrote: My argument isn't about the standards or other software in general, my argument is how asterisk (and in this case only asterisk) comes, that is with SRV *disabled*, and the fact many people wouldn't understand what it's for, or why they should enable it. The documentation, well what documentation there is, simply isn't coherent enough, or detailed enough to explain these things, and the few lines in the config file certainly doesn't explain anything either... ;srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; Note: Asterisk only uses the first host in SRV records Joe Public stumbles upon asterisk, not a clue what all the features and modules and what not is for, do you think he'll be clued up enough to remove the leading semicolon??? That is what my point is, until asterisk has it enabled by default, and all the current user base use a version of asterisk that supports it properly is there all that much point in promoting it so heavily? Now how many pieces of MTA software out of the box have MX record lookups disabled??? I'd hazard a guess at none... In your own sandbox, feel free to do whatever you want. If the companies you promote asterisk to are going to call only you, feel free to promote whatever you want to them. But if any of them want to be interoperable with the rest of SIP-world, please STOP telling them to use A records instead of SRV. That way you are forcing others to do hacks they do not want to do ! SRV is essential for SIP, whether you like it or not, and there are companies out there that rely on it. Moreover, SIP is not just asterisk. You have absolutely no idea how many other SIP implementations are out there, you have no idea whether they follow RFC3265 or not, you have no idea how many companies rely or SRV. Nobody knows this and it is clear that your arguments are based on what do you *THINK* others do. And this is very bad because it kills interoperability. So, please, stop promoting using of A records instead of SRV, that would make our life a little bit easier, thank you. Jan. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Spring VON Wrap Up
On 05-04 14:35, Steven Sokol wrote: TCP/TLS would be used for the SIP messaging which handles call setup, teardown, and other non-Realtime functions. The voice stream will still be handled via RTP which is a UDP-based protocol. The reason for doing the call setup as TCP is to allow for TLS encryption. The SIP messages themselves are simply bits of ASCII text (much like SMTP messages). Currently Asterisk does SIP over UDP only (I think...). In order to support SIPS (Secure SIP, like HTTPS) we need to build a version of chan_sip (or chan_sip2 ;-) that supports SIP over TCP. The voice stream will remain UDP an therefore not succumb to enormous delay. There are some more reasons -- transport of big SIP messages and avoiding network congestion among them. SIP message can get pretty big when XML encoded documents (presence documents, for example) are attached. TCP does not fit everywhere. It is still advantageous to let SIP phones use UDP when communicating with a proxy because the proxy does not have to keep a list of opened connections which is very resource consuming (just consider that you have 10 users using the same proxy -- that can be easily achieved using single server). On the other hand, TCP is useful for proxy-to-proxy communication, especially when there is bigger amount of traffic between proxies. In this case TCP head blocking is really not a problem because the sender gets constant feedback from the remote party and can retransmit the lost segment in a short time. (There was a technical report on this published by Henning Schulzrinne). Jan. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Draytek SIP phones are broken
Hello, if you have a Draytek SIP phone, please check if the phone doesn't flood your server with SIP REGISTER messages. Draytek phones are broken and keep sending REGISTER messages after receiving 200 OK (even if expires value is long enough). Several such phones are flooding iptel.org public servers these days. If you have direct contact to Draytek developers, please send it to me. thanks, Jan. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] KPhone working
On 13-01 12:17, Maciek Kaminski wrote: Steve wrote: Hi, If anyone else had a problem I got kphone to work with Asterisk. I have problems with kphone + Asterisk. KPhone does not seem to ACK invites, ie. KPhone --- sends INVITE -- Asterisk KPhone -- sends 101 Trying --- Asterisk KPhone -- sends 202 OK --- Asterisk KPhone --- does not send ACK Could you, please, send me SIP message dumps of this ? Jan. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] SIP response 403 That is ugly
It means that the username in From and the username in digest
credentials are different.
The reason for this test is that we do not want our users to pretend
that they are somebody else. Without this test it would be possible to
put [EMAIL PROTECTED] in From and all phones will display it, although
real username in digest credentials (which is verified by the proxy) is
different.
Jan.
On 11-12 11:34, jerk face wrote:
I am trying to make an outgoing call using an iptel
account using Asterisk. I have followed a how-to for
asterisk and iptel found at
http://www.voip-info.org/tiki-index.php?page=Asterisk%20sip%20client%20SER
I am getting the following error message:
Got SIP response 403 That is ugly -- use From=id next
time (OB) back from 195.37.77.101
I'm not quite sure what that means. Does anybody know
what I might have done wrong?
Here is my configuration:
sip.conf
register = account:[EMAIL PROTECTED]/
[iptel]
type=friend
username=account
secret=passwd
host=iptel.org
extensions.conf
exten=_3.,1,SetCallerID(myNumber)
exten=_3.,2,SetCIDName(myName)
exten=_3.,3,Dial,SIP/${EXTEN:[EMAIL PROTECTED]
exten=_3.,4,Playback(pbx-invalid)
exten=_3.,5,Hangup
Some guy from the iptel mailing list told me:
It means you use different user names
in From and the authentication header.
I don't know what he's talking about so if anybody
could point me in the right direction, that would be
appreciated.
Thank you for your time.
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Re: [Asterisk-Users] Symmetric RTP
Yes. Jan. On 26-11 22:16, Olle E. Johansson wrote: Anyone that knows if the Asterisk SIP channel supports symmetric RTP? /O ___ 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] Symmetric RTP
On 27-11 15:14, Olle E. Johansson wrote: Jan Janak wrote: On 26-11 22:16, Olle E. Johansson wrote: Anyone that knows if the Asterisk SIP channel supports symmetric RTP? Yes. Followup question: Both as a SIP UA (Client) and as a SIP proxy? I don't know, I tried asterisk as a SIP UA behind a NAT with SER in the public internet and it worked. So at least the SIP UA part works. Jan. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Symmetric RTP
I tested the following scenario: private network| public internet SIP Phone 1 --- Asterisk --- NAT --- SER --- SIP Phone 2 and it worked. I was able to make calls from phone 1 to phone 2 and vice versa. Jan. On 27-11 16:37, David Luyens wrote: Have you tried SER to * in the same setup? David -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Jan Janak Verzonden: donderdag 27 november 2003 15:26 Aan: [EMAIL PROTECTED] Onderwerp: Re: [Asterisk-Users] Symmetric RTP On 27-11 15:14, Olle E. Johansson wrote: Jan Janak wrote: On 26-11 22:16, Olle E. Johansson wrote: Anyone that knows if the Asterisk SIP channel supports symmetric RTP? Yes. Followup question: Both as a SIP UA (Client) and as a SIP proxy? I don't know, I tried asterisk as a SIP UA behind a NAT with SER in the public internet and it worked. So at least the SIP UA part works. Jan. ___ 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] Radius on *
On 17-11 16:33, Jeremy McNamara wrote: Sebastian Nocetti wrote: Does Asterisk support Radius accounting? No and there is absolutely no need for it to. RADIUS is not anything that should have ever been deployed in a VoIP environment. You would be surprised how many people find RADIUS support useful in SIP Express Router and actually _use_ it. It is one of the most desired features, I am not kidding. Also IETF recently decided to standardize using of RADIUS with SIP, mainly because there is a huge user base. Jan. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP protocol bug ???
Asterisk was wrong. Every SIP message can be challenged with 401 or 407,
depending on who is challenging.
If you send a REGISTER message then you can get 407 Proxy
Authentication Required from any proxy along the path of the message.
You can also get 401 Unauthorized from registrar.
The same for INVITE, you can get a 407 Proxy Authentication Required
from a proxy and you can also get 401 Unauthorized from a PSTN gateway, for
example.
The rule of thumb is: If a SIP network element forwards the request then
it will use 407, if it is the final destination for the request (PSTN
gw, registrar, user agent) then it will use 401.
One message can be also challenged several times before it get's to its
final destination.
Jan.
On 07-11 09:35, mtm spm wrote:
Hi Olle,
--- Olle E. Johansson [EMAIL PROTECTED] wrote:
The first Invite is without credentials, since
digest authentication needs input
from the server to create credentials.
This is also what I understood too from rfc.
I was just confused becouse in the Asterisk code
there was something like this:
case 401: /* Not authorized on REGISTER */
if (p-registry !strcasecmp(msg, REGISTER)) {
if ((p-authtries 1) || do_register_auth(p, req,
WWW-Authenticate, Authorization)) {
ast_log(LOG_NOTICE, Failed to authenticate on
REGISTER to '%s'\n, get_header(p-initreq, From));
p-needdestroy = 1;
} elsep-needdestroy = 1;
So, only the case of registration was handled in 401.
However I just added something like:
if(!strcasecmp(msg, INVITE)) {
if ((p-authtries 1) || do_proxy_auth2(p, req,
INVITE, 1)) {
ast_log(LOG_NOTICE, Failed to authenticate on
INVITE to '%s'\n, get_header(p-initreq, From));
p-needdestroy = 1;
}
and I implemented the new function do_proxy_auth2 as a
modified version of do_proxy_auth to work with
Authorization instead of Proxy-Authorization and I
got the thing to work fine.
However, I wanted to check with others who was wrong:
Asterisk or my SIP provider.
Is this the right thing to do in respect to the
standard?
Tnx,
MTM
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Re: [Asterisk-Users] SIP protocol bug ???
On 07-11 13:17, John Todd wrote: From what I can understand of the issue you describe, it sounds like the problem resides on the remote side, and not Asterisk's side. You are sending an invalid request in your first query, and the remote side is sending Unauthorized, meaning that it believes you have supplied credentials, but they are the wrong credentials. This is the end of the conversation, since both sides have given their final words on the subject. Unauthorized means that the message contained no credentials or the server was unable to verify the credentials. When a user agent (asterisk) gets an Unauthorized message then it is supposed to retry with proper credentials. What arguably _should_ be happening is that the remote SIP host should be sending 407 Proxy Authentication Required, but it's not. Therefore, Asterisk is behaving correctly. This is not a bug in Asterisk. That depends on the type of the remote host. Registrars, PSTN gateways, and user agents send 401, proxies send 407. In any case asterisk should be able to handle both for any type of message except ACK and CANCEL (which can not be challenged). Jan. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk behind NAT to SIP provider
I experimented a little bit and Asterisk behind NAT with SIP works. I created
an account at iptel.org and use that account for outbound SIP traffic from
Asterisk.
I am using [EMAIL PROTECTED], all the SIP traffic will be sent to
iptel.org proxy and the proxy will take care of NAT traversal. Currently
I forward all numbers begining with 3 to iptel.org beucase I don't know
how to create fall-back rule that will match when there are no other
rules (neither i nor _. works for me).
In the other direction, calls to [EMAIL PROTECTED] get translated to
[EMAIL PROTECTED] and user jan registered at the asterisk box will
receive them.
To able able to call anywhere through iptel.org, From header field must
contain iptel.org so fromdomain parameter is necesarry in [iptel]
section.
Testing scenario was as follows:
[Caller][*]---[NAT][iptel.org (public inet)][NAT]---[Callee]
and vice versa.
sip.conf and extensions.conf follow. I have no previous experience in
configuriing asterisk so maybe the config files are not the best ones, I
simply took John Todd's config files and tweaked them a bit, it seems to
work for me.
To iptel.org proxy asterisk looks like a normal SIP user agent behind
NAT. iptel.org is running SER with extended nathelper and RTP proxy.
Jan.
;
; SIP Configuration for Asterisk
;
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to
context = from-sip ; Default for incoming calls
;
register = asterisk:[EMAIL PROTECTED]/jan ; Register with a SIP
provider
[iptel]
type=friend
username=asterisk
secret=password
fromdomain=iptel.org
host=iptel.org
[jan]
type=friend
username=jan
host=dynamic
canreinvite=no
extensions.conf:
[from-sip]
exten = jan,1,Dial(SIP/jan)
exten = jan,2,Hangup
exten = _3.,1,SetCallerID(jan)
exten = _3.,2,SetCIDName(Jan Janak)
exten = _3.,3,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _3.,4,Playback(invalid)
exten = _3.,5,Hangup
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Re: [Asterisk-Users] Asterisk behind NAT to SIP provider
Try to register your Asterisk with iptel.org. There is a public SIP proxy running with support for NAT traversal. To do so, disable any NAT traversal features in Asterisk (the proxy can detect that you are behind a NAT and will modify the messages accordingly). You don't have to setup anything in your NAT box. Let me know if that doesn't work (I will need your username in that case to find out why). Jan. On 24-10 23:00, Jonathan Hogg wrote: On 24/10/2003 19:31, rnc Info Lists wrote: I have the same problem and have solved it by using iaxtel.com. Asterisk talks to IAXtel quite well on inbound and outbound from behind my NAT router. Yeah, I got that working as a test that Asterisk could successfully route calls in and out to my extensions, but I need a PSTN gateway service that can offer numbers in London and NY. I'm talking to a UK provider, but they only do SIP at the moment. I'm working with one of their tech guys to see if they can support IAX via an Asterisk installation at their end. Jonathan -- Jonathan Hogg Director, Technology Seventh Wave Systems Ltd. 4-14 Tabernacle Street London EC2A 4LU Telephone: +44 20 7074 0423 http://www.seventh-wave-systems.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] what is the best codec for low bandwidth? for quality?
From my experience iLBC is unbeatable on lossy and slow links. I have been in situations where no other codec (GSM, Speex, G.729) worked and iLBC was still fairly usable. Jan. On 22-10 23:29, Matthew Simpson wrote: The number of codecs is overwhelming to me. What do ya'll consider the best codec for conserving bandwidth? [I realize at the cost of quality] Secondly, what do you think the best codec for voice quality is? Yours, Matthew ___ 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] Survey: Grandstream improvements.........
On 22-10 15:38, Michael T Farnworth wrote: In fact I believe a SIP client doesn't have to support TCP, but fortunately I believe the Grandstream does. RFC3261 compliant SIP clients must support TCP. Jan. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] */SER/FW
On 15-10 22:51, Uriel Carrasquilla wrote: is it safe to say that if you have SER on the public side of the Internet, then you can deal with Asterisk behind a NAT plus the UA (SIP Phones) behind a NAT as well when you force SER to be STATEFUL (i.e. the state of the call is maintained)? My challenge is that I have Asterisk in places where I don't have access to a public IP address. Regards, Uriel If you have SER in the public internet along with RTP proxy ( http://www.portaone.com/~sobomax/rtpproxy.tar ) then you can can deal with any symmetric SIP user agent behind NAT. In some cases SER will force RTP proxy, but it tries to minimize using the rtp proxy. I don't know if asterisk is symmetric, most existing sip phones are. SER can't be call stateful, it's a SIP proxy. Jan. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SER vs STUND with Asterisk..
On 16-10 00:55, John Todd wrote: [...] SIP system administrators. The hopefully-soon-to-be-approved ICE RFC's will make things even easier by testing even the RTP ports, but it will be some time before we see clients with that functionality built in. I am personally a little bit skeptic about the implementation of the ICE draft. It is very complex and hard to implement, just realize that you have to run STUN on every port you are going to use. It is maybe more complicated than SIP itself. But on the other hand it works really everywhere and the draft is well written, one can get a good overview how NATs complicate our life :-). Very good results can be achieved with symmetric SIP user agents, such user agents can be made to work behind any NAT quite easily when the SIP server is in the public internet. Fortunatelly most existing SIP phones and user agents are symmetric. 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)
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] Mini-PC box to run server
A couple of links: http://www.pogolinux.com/ http://www.linuxdevices.com/products/PD6582676142.html http://www.linux-works.com/ http://www.soldam.com/ http://www.uclinux.org/ucsimm/ http://www.sun.com/hardware/serverappliances/index.html Hope that helps. Jan. On 08-10 13:00, Chris Albertson wrote: Looking for a low-end PC that has following characteristics 1) headless meaning no video card, keyboard, or mouse 2) At least three PCI slots or two slots and built in Ethernet 3) Very low or zero noise fan (convection cooled?) 4) Small physical size. 5) Low cost I don't mind building a PC from parts, the big thing is the case. Standard mid-tower is way to bulky noisey I assume any PC sold today has enought CPU power to run a small asterisk system = 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] the g729 situation
Ask the producers to implement iLBC: http://www.globalipsound.com/products/iLBCfreeware.php It is free and from my experience one of the best codecs available. Grandstream promised to implement it in the future :-). X-lite and kphone ( http://www.wirlab.net/kphone ) support it. Jan. On 03-10 08:31, Uriel Carrasquilla wrote: If we avoid g729, what options do we have for SIP/Budgetone/Grandstream when communicating to * via the Internet and still have something comparable to GSM? Uriel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jan Rychter Sent: Thursday, October 02, 2003 2:15 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] the g729 situation LDM == Louis-David Mitterrand [EMAIL PROTECTED] writes: LDM Having purchased a license for 5 g729 channels on Digium's web LDM shop I thought registration and installation would be a snap. NOT. LDM I followed registration instructions to the letter but it failed LDM with that message: LDM ERROR! Your Internet connection is probably behind a proxy and the LDM Registration program can't communicate with our server LDM Which is stupid as my * box is a firewall and sits directly on the LDM Internet whith no restrictions from in-out. I must say I'm impressed that people are brave enough to (1) accept the long, restrictive and sometimes outright scary (did you read the parts about credit card charges, or the definition of G.729 software in connection with Improvement by Licensee?) licensing agreement and (2) run a binary module that touches strange parts of the machine and communicates that information over the network to a third party. I also feel sorry for Digium, because they have to take the heat from unhappy users. IMHO this codec should be avoided at all cost. --J. ___ 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 text messages with Asterisk
On 03-10 12:45, Luis Vazquez wrote: WipeOut wrote: Luis Vazquez wrote: Hello all, I'm new to this list and starting with Asterisk. Have any of you have tried a SIP client (like Microsoft messenger) to sent text messages and voice through an Asterix server? Is this possible or the Asterix server simply can't manage this kind of traffic? Regards, Luis Are you asking if Asterisk supports SIMPLE, I have not head of any support for it.. If you are asking if Asterisk can be a M$ IM server then I am sure the answer is almost definately not.. Later.. You are right, i am searching for SIMPLE support with Asterisk, M$ IM is just a (bad?) example client that can handle voice and text messages in the same application. I would like a lot to have a free client (or at least not M$'s) which could handle this over SIP. Kphone for linux: http://www.wirlab.net/kphone Jan. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX vs SIP
Hello, On 19-09 19:48, WipeOut . wrote: Also IAX does not care about NAT so a situation like.. AST--NAT--Internet--NAT--AST ..will work fine.. SIP will have problems in a setup like this without the use of specialised NAT routers.. I am wondering how setup like this could work with IAX (or any other protocol) when symmetric NATs are used. If you have two different NATs then direct connection is not possible between hosts behind those two NATs. You have to do some kind of provisioning of the NAT boxes (i.e. port forwarding). Jan. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP registration
Hello, I don't know if it is the problem, but the message below is syntactically invalid, there must be space between the name token in From and To (704) and the URI, i.e. correct From should look like this: From: 704 sip:[EMAIL PROTECTED];tag=230b0-e0 instead of this: From: 704sip:[EMAIL PROTECTED];tag=230b0-e0 Jan. On 19-09 08:38, Sergio Serrano Revuelto wrote: I have the same problem, Asterisk debug is the next: REGISTER sip:AVANZADA7 SIP/2.0 Call-ID: [EMAIL PROTECTED] From: 704sip:[EMAIL PROTECTED];tag=230b0-e0 To: 704sip:[EMAIL PROTECTED] CSeq: 101 REGISTER Via: SIP/2.0/UDP 192.168.0.154:5060 Contact: sip:[EMAIL PROTECTED]:5060 Max-Forwards: 70 Expires: 1800 Supported: timer Content-Length: 0 11 headers, 0 lines Using latest request as basis request Sending to 192.168.0.154 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.0.154:5060 From: 704sip:[EMAIL PROTECTED];tag=230b0-e0 To: 704sip:[EMAIL PROTECTED];tag=as539680e1 Call-ID: [EMAIL PROTECTED] CSeq: 101 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.0.154:5060 DEBUG[12301]: File chan_sip.c, Line 835 (__sip_destroy): Destorying call '[EMAIL PROTECTED]' 10 headers, 0 lines Reliably Transmitting: OPTIONS sip:192.168.0.154 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.207:5060;branch=z9hG4bK160c3fcc From: asterisk sip:[EMAIL PROTECTED];tag=as6c232c12 To: sip:192.168.0.154 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 192.168.0.154:5060 Sip read: SIP/2.0 200 OK Call-ID: [EMAIL PROTECTED] From: asterisksip:[EMAIL PROTECTED];tag=as6c232c12 To: sip:192.168.0.154 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 192.168.0.207:5060;branch=z9hG4bK160c3fcc Supported: timer Allow: REFER,UPDATE,INFO,MESSAGE,OPTIONS Accept: application/sdp Accept-Encoding: Accept-Language: en;q=0.8 User-Agent: Netergy MicroElectronics Content-Length: 0 My sip.conf is the next: [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = outgoing ; Default for incoming calls disallow=all allow=alaw tos=lowdelay [704] type=friend username=704 secret=704 host=192.168.0.154 dtmfmode=inband mailbox=704 callerid=704 context=outgoing reinvite=no canreinvite=no qualify=300 nat=1 ANY IDEA ABOUT THIS? srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Hielke Christian Braun Enviado el: jueves, 18 de septiembre de 2003 19:05 Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] SIP registration Hello, try to change [siptestphone] to [atrg613test] in sip.conf. Maybe that helps. Regards, Christian. On Thu, Sep 18, 2003 at 05:52:19PM +0200, Tais M. Hansen wrote: Hi, I'm having problems letting a SIP endpoint register at Asterisk. Here's the debug output from Asterisk: ... sip.conf: [general] port=5060 bindaddr=s.s.s.s context=cxnet-in tos=lowdelay [siptestphone] type=friend user=atrg613test host=dynamic defaultip=c.c.c.c ___ 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 registration
No, it is not something you can fix by tweaking the configuration files, you should complain to the authors of the user agent. Anyway, it is a minor problem and I guess that most implementations can overcome it, but you should at least report it to the authors. Jan. On 19-09 09:17, Sergio Serrano Revuelto wrote: Thanks, my phone has the next sip setting. Can you help me with correct parameters with the below sip.conf? SIP Server Settings * Server Address: (IP or FQDN) * Port: * Domain Name: * Send Registration Request: (true or false) Gateway Settings Dial Plan: Transport: (UDP tor TCP ) Phone Number: CallerID Name: Port: AEC: (On or OFF) User Name: Password: Thanks for all srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Jan Janak Enviado el: viernes, 19 de septiembre de 2003 8:59 Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] SIP registration Hello, I don't know if it is the problem, but the message below is syntactically invalid, there must be space between the name token in From and To (704) and the URI, i.e. correct From should look like this: From: 704 sip:[EMAIL PROTECTED];tag=230b0-e0 instead of this: From: 704sip:[EMAIL PROTECTED];tag=230b0-e0 Jan. On 19-09 08:38, Sergio Serrano Revuelto wrote: I have the same problem, Asterisk debug is the next: REGISTER sip:AVANZADA7 SIP/2.0 Call-ID: [EMAIL PROTECTED] From: 704sip:[EMAIL PROTECTED];tag=230b0-e0 To: 704sip:[EMAIL PROTECTED] CSeq: 101 REGISTER Via: SIP/2.0/UDP 192.168.0.154:5060 Contact: sip:[EMAIL PROTECTED]:5060 Max-Forwards: 70 Expires: 1800 Supported: timer Content-Length: 0 11 headers, 0 lines Using latest request as basis request Sending to 192.168.0.154 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.0.154:5060 From: 704sip:[EMAIL PROTECTED];tag=230b0-e0 To: 704sip:[EMAIL PROTECTED];tag=as539680e1 Call-ID: [EMAIL PROTECTED] CSeq: 101 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.0.154:5060 DEBUG[12301]: File chan_sip.c, Line 835 (__sip_destroy): Destorying call '[EMAIL PROTECTED]' 10 headers, 0 lines Reliably Transmitting: OPTIONS sip:192.168.0.154 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.207:5060;branch=z9hG4bK160c3fcc From: asterisk sip:[EMAIL PROTECTED];tag=as6c232c12 To: sip:192.168.0.154 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 192.168.0.154:5060 Sip read: SIP/2.0 200 OK Call-ID: [EMAIL PROTECTED] From: asterisksip:[EMAIL PROTECTED];tag=as6c232c12 To: sip:192.168.0.154 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 192.168.0.207:5060;branch=z9hG4bK160c3fcc Supported: timer Allow: REFER,UPDATE,INFO,MESSAGE,OPTIONS Accept: application/sdp Accept-Encoding: Accept-Language: en;q=0.8 User-Agent: Netergy MicroElectronics Content-Length: 0 My sip.conf is the next: [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = outgoing ; Default for incoming calls disallow=all allow=alaw tos=lowdelay [704] type=friend username=704 secret=704 host=192.168.0.154 dtmfmode=inband mailbox=704 callerid=704 context=outgoing reinvite=no canreinvite=no qualify=300 nat=1 ANY IDEA ABOUT THIS? srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Hielke Christian Braun Enviado el: jueves, 18 de septiembre de 2003 19:05 Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] SIP registration Hello, try to change [siptestphone] to [atrg613test] in sip.conf. Maybe that helps. Regards, Christian. On Thu, Sep 18, 2003 at 05:52:19PM +0200, Tais M. Hansen wrote: Hi, I'm having problems letting a SIP endpoint register at Asterisk. Here's the debug output from Asterisk: ... sip.conf: [general] port=5060 bindaddr=s.s.s.s context=cxnet-in tos=lowdelay [siptestphone] type=friend user=atrg613test host=dynamic defaultip=c.c.c.c ___ 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
