similar to: asterisk 1.8.8 - caller ID not working.

I'm trying Asterisk 10.0 (as 8.x is not passing PSTN CallerID) and Asterisk 10.0 is no better. I'm still getting: WARNING[12295]: chan_sip.c:14446 check_auth: username mismatch, have <11>, digest has <pstn-1270> NOTICE[12295]: chan_sip.c:22769 handle_request_invite: Failed to authenticate device "KMIEC Z" <sip:7804715665 at 10.0.0.110>;tag=1c1222950155 Anybody

I'm having problem passing Caller ID to asterisk from AudioCodes MP-114 (FXO) AudioCodes is passing the caller ID to Asterisk but Asterisk is trying to interpret it as authentication: [Dec 31 11:41:07] WARNING[9752]: chan_sip.c:8553 check_auth: username mismatch, have <pstn-5665>, digest has <pstn-1270> [Dec 31 11:41:07] NOTICE[9752]: chan_sip.c:14316 handle_request_invite:

I'm using "insecure=invite" with two different dial plans, it it working with one dial plan but not with the other. What other parameters could influence "insecure=invite" In sip.conf below "insecure=invite" is working OK [pstn-1270] type=friend secret=spa3k username=voice-1270 mailbox=369 host=dynamic insecure=invite canreinvite=no disallow=all allow=ulaw

Does the sort order matter in sip.conf file? I know sort order might effect: allow=ulaw allow=alaw but does it matter where I place: insecure=invite ? The reason I'm asking is that I've loaded almost two identical (sip.conf and extension.conf) files on the same asterisk server and with one set insecure=invite is working correctly. When I load the second set of dial plan (sip.conf and

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,

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 =>

We have a caller ID from our phone provider "Shaw Cable" (digital phone) and it was working OK until recently. I get an error: WARNING[6769]: chan_sip.c:8553 check_auth: username mismatch, have <4>, digest has <pstn-4444> NOTICE[6769]: chan_sip.c:14316 handle_request_invite: Failed to authenticate user THELMA <sip:7804789998 at 10.10.0.103>;tag=50e17675d59121c4o1 at

Hello, everything goes well on asterisk 1.4.30, but with asterisk 1.6.2.11 I get the following : [Sep 12 18:59:29] WARNING[2066]: chan_sip.c:12738 check_auth: username mismatch, have <329909006666>, digest has <3291119600> [Sep 12 18:59:29] NOTICE[2066]: chan_sip.c:20082 handle_request_invite: Failed to authenticate device "0473990000" <sip:0473990000 at

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 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

I'm having problem passing Caller ID to asterisk from AudioCodes MP-114 (FXO) AudioCodes is passing the caller ID to Asterisk but Asterisk is trying to interpret it as authentication: [Dec 31 11:41:07] WARNING[9752]: chan_sip.c:8553 check_auth: username mismatch, have <pstn-5665>, digest has <pstn-1270> [Dec 31 11:41:07] NOTICE[9752]: chan_sip.c:14316 handle_request_invite: Failed

Darryl Moore <darryl at moores.ca> schrieb: > I'd start by turning on sip debugging in asterisk > >sip set debug ip [your_phone_ip] Really destroying SIP dialog '490d1996593c8e11217828b71aae5c4d at 172.16.34.133' Method: OPTIONS Reliably Transmitting (no NAT) to 192.168.200.11:5060: OPTIONS sip:00493512222222 at 192.168.200.11:5060 SIP/2.0 Via: SIP/2.0/UDP

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

Hi everyone, I noticed that script/backgroudrb requires config/environment which causes the backgroundrb server as well as the log worker to ''bloat'' to 35MB each. I am kind of sensitive to memory issues, so I patched the code and essentially moved the require of environment from script/backgroundrb to the meta_worker. Everything seems good and now both backgroundrb server and

Hi list, I have installed SIPp into my server. But not able to used it properly. how to configure with my server ? how to see logs on webpage ? how to start call testing .... when i start SIPp then found verious hits on myserver. *CLI:- * [Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not

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:

I'm receiving the following error over and over, adnauseam: Oct 1 23:59:53 NOTICE[3194]: chan_sip.c:5890 check_auth: stale nonce received from 'CNAME-CID <sip:5551212@192.168.1.X>' Does anyone know what "stale nonce" is? Thanks! Paul Conn -------------- next part -------------- An HTML attachment was scrubbed... URL:

I happened to be in the cli tonight as some (208.122.57.58) initiated a simple attack - just trying to make long distance calls from outside context. Although harmless, this went on for several minutes as the idiot just used up my bandwidth with SIP messages. Here's and example: [2011-12-28 22:53:42] NOTICE[9635]: chan_sip.c:14035 handle_request_invite: Call from '' to extension

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