similar to: Still true: only first peer matched on incoming call?

Hi! When using multiple identities on one physical phone (Snom 320), I get check_auth: username mismatch, have <7705>, digest has <7736> messages when placing a call from a different account than the first one. From reading the asterisk source, I can see that the problem is that peer authentication is not matched against username, but against ip/port. I need to have multiple

I'm getting dozens of these at a very high rate: [Jan 20 09:15:27] NOTICE[16958]: chan_sip.c:12687 check_auth: Correct auth, but based on stale nonce received from '"" <sip:121 at gnat.com>;tag=as5f1a9480' [Jan 20 09:15:28] NOTICE[16958]: chan_sip.c:12687 check_auth: Correct auth, but based on stale nonce received from '"" <sip:130 at

Ahh. Seen that before! That suggests to me that you don't have your sip.conf records setup right. What's your sip.conf look like? On 15-05-28 04:51 PM, Luca Bertoncello wrote: > Kevin Larsen <kevin.larsen at pioneerballoon.com> schrieb: > >> The phone you gave your wife is really old. Are you sure it supports SIP >> OPTIONS? Can you make a call in or out to it?

> No, I'm not sure. > And no, I can't make any call, right now... At least, not connected to my > Asterisk... > If I connect it to the other VM with AsteriskNOW I can call my Twinkle, but > NOT my phone connected on my Asterisk, using the "proxy". > I can see that in the log: > > [May 28 22:49:51] WARNING[4135]: chan_sip.c:12800 check_auth: username

Kevin Larsen <kevin.larsen at pioneerballoon.com> schrieb: > The phone you gave your wife is really old. Are you sure it supports SIP > OPTIONS? Can you make a call in or out to it? If you can, it is more > likely that it just doesn't support that and you can't use a qualify > statement. No, I'm not sure. And no, I can't make any call, right now... At least,

Call from 'sip' to extension '+1xxxyyyzzzz' rejected because extension not found in context 'out'. But [out] exten => s,1,NoOp( this is the extension: ${EXTEN}) exten => s,n,Answer() exten => s,n(weasels),PlayBack(weasels-eaten-phonesys) ........ If I set "s" to "_." it works. Shouldn't "s" work here? Is it because the

I think your phone may be trying to register with the username '1234', while your sip configuration is expecting 'luca'. Can you try changing your phone registration credentials to use 'luca'? Can you give us a sip transcript when you try to place a call from it? On 15-05-28 05:09 PM, Luca Bertoncello wrote: > Darryl Moore <darryl at moores.ca> schrieb: >

I have two Asterisks connected using SIP. One is acting as a SIP "server", the other as a SIP "client". This almost works; but calls from 50607795 are rejected with this error: check_auth: username mismatch, have <50607796>, digest has <50607795> On the "client" I have these accounts configured in sip.conf: register => 50607795:test at

Darryl Moore <darryl at moores.ca> schrieb: > Ahh. Seen that before! That suggests to me that you don't have your > sip.conf records setup right. > > What's your sip.conf look like? Well, here what I wrote in my sip.conf: register => 00493511111111:MYSECRET at pbxluca/00493511111111 register => 00493512222222:MYSECRET at pbxfax/00493512222222 register =>

In SIP setting on the e71 I set the public user name as 1995 at 10.10.11.180. There is a sip.conf context [1995] On the asterisk CLI I get: Registration from '<sip:%201995 at 10.10.11.180:5060>' failed for '10.10.11.98' - No matching peer found So I changed the sip.conf context to [%201995] Then: [2009-11-19 20:44:28] WARNING[14371]: chan_sip.c:11797 check_auth:

Hi, Has anyone had a problem setting up two registrations (on the same Asterisk server) on one Polycom phone? When the user tries to make a call on the 2nd line, it works fine. But when they try the first line, the CLI says:- Using INVITE request as basis request - 9f5fe9a5-215d0f3a-b2fbe6b7 at 192.168.1.138 Found peer client _202' <--- Which is incorrect, it should be client_201. And

I get this when my brother in law tries to call in from his box to mine. WARNING[4855]: chan_sip.c:12675 check_auth: username mismatch, have <100>, digest has <s> or after changing the register line: WARNING[4855]: chan_sip.c:12675 check_auth: username mismatch, have <100>, digest has <199> I have done everything I can think of and still failure. Currently the

On 11.9, trying to set up a callcentric peer: sip debug: > <--- SIP read from UDP:204.11.192.161:5060 ---> > INVITE sip:1777<myccid>@10.10.11.180:5060 SIP/2.0 > v: SIP/2.0/UDP 204.11.192.161:5060;branch=z9hG4bK-6104e46aaaaef4249814d16a2ffb990d > f: <sip:<calling number>@66.193.176.35>;tag=3606475083-968127 > t:

I`ve been noticing alot of those messages in the CLI lately: Apr 5 11:18:02 NOTICE[25593]: chan_sip.c:6444 check_auth: stale nonce received from '<sip:reg-1@pbx.domain.com> I haven't changed my configuration in ages. What could be the cause of this suddent appearance? Mike -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello everyone, I am just starting a project for school in RoR and I am a complete n00b ^^ Here''s my problem : I need to get the user''s login stored in the session but for some reason I cannot. Here''s my login method code : def login if request.post? @user = User.find_by_username(params[:login]) if @user and @user.password_is? params[:password]

Hi masters! I've this Asterisk 1.4.15 running. yesterday I had to change the firewall schema that I had before. I use to have a FW that would be my network FW/Proxy and do the NATs for Asterisk. This FW was receiving too many requests from my LAN and it was making the Asterisk 'cut' the calls or reach very high latency. Yesterday I added a new FW just for this Asterisk. The same

Hi all, I've got a Polycom 501 that I just can't seem to get lines 2 and 3 to work on.? Line 1 works fine. When my user tries to use line 2 or 3 to dial out, they get a fast busy signal and I get this error message on the console: =============================================================================== *CLI> [Jul? 7 09:49:36] WARNING[26513]: chan_sip.c:12729 check_auth:

Scenario: [Asterisk Server] on routed/public IP \/ /\ \/ /\ \/ /\ \/ /\ \/ [Draytek Router] --> internal IP's \/ /\ \/ /\ \/ /\ \/ /\ \/ [internal Network 192.x.x.x] [IP10s] + [IP10s] + [Softphones] Everything is good as long as only *1* phone is registered from the internal network. The minute another phone goes online registration is dropped and the sip debug complains:

I have an * server running 1.0.9 on a FC3 machine. I connect around 44 BT102 phones to it and 6 Sipura 2000 units. Everything is working great but lately I have seen the following error: Aug 4 13:33:24 WARNING[31907]: chan_sip.c:4826 check_auth: Stale nonce received from '<sip:4000@148.235.174.85>' Aug 4 13:33:24 WARNING[31907]: chan_sip.c:4826 check_auth: Stale nonce received

I can't get a test T.38 fax between 2 1.6.2 machines, using app _fax and spandsp pre17 and 20100501. The machines can't seem to get connected. send side extensions.conf: [fax-tx-test] exten=>s,1,NoOp(Context fax-tx-test) exten=>s,n,SendFAX(${FaxFile}.tif) exten=>s,n,HangUp() exten=>h,1,NoOp(FAXSTATUS: ${FAXSTATUS} FAXERROR: ${FAXERROR} FAXMODE: ${FAXMODE})