similar to: can't set up tcp sip - sip connection : digest <s> problem

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

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

I've got PBXNSIP running on a windows box and it is trying to register with my Asterisk box. I can set up one trunk and it works fine, however if I try to setup a second trunk from the same box, there is some sort of authentication issue where Asterisk appears to be confusing which trunk is which. Here is the chunk from my sip.conf: [TEST1] context=STUFF-LD type=friend

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:

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:

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

Greetings, I've noticed a problem that might originate from my Asterisk configuration, could use a hand in sorting it out. Problem is a 488 response from Asterisk whenever it gets RTP/SAVPF profile in the SDP. My current setup has Asterisk Kamailio realtime integration, and Kamailio uses dispatcher to route calls for Asterisk to handle. Now I have only one Asterisk, on the same machine as

I'm trying to setup sipgate on 1.6.1. Following the instructions on the site: http://www.sipgate.com/faq/article/394/How_do_I_configure_Asterisk, I created [sipgate] in sip.conf and a [sipgate-in] in extensions.conf: [sipgate] type=friend secret= ;;SIP_PASSWORD insecure=port,invite defaultuser= ;; SIP-ID fromuser= ;;SIP-ID context=sipgate_in fromdomain=sipgate.com host=sipgate.com

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,

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

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

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 =>

Hello List, I have a little issue with calls placed to a provider declared on sip.conf, because of a not clear (*for me*) behavior of 'remotesecret' parameter. Before continuing, this is my environment: Asterisk: 1.6.2.16.1 OS: CentOS release 5.5 (Final) 2.6.18-194.32.1.el5 Details: I have this block on sip.conf ----- start ---- ... register => john:j0nhp4ss

I'm having a strange problem recently with a Yealink SIP-T28P phone connected to Asterisk 11.4.0 via openvpn. It was working fine for months, and now when I dial anything from the phone, it shows "Forbidden", and the Asterisk console shows: [May 21 10:47:49] NOTICE[28518][C-00000004]: chan_sip.c:25189 handle_request_invite: Failed to authenticate device "Ext 110" <

I just noticed after upgrade from Asterisk 1.4.39 to 1.8.8 my caller ID is not working WARNING[1671]: chan_sip.c:13956 check_auth: username mismatch, have <11>, digest has <pstn-1270> NOTICE[1671]: chan_sip.c:22048 handle_request_invite: Failed to authenticate device "KMIEC Z" <sip:7804715665 at 10.0.0.110>;tag=1c976040515 -- Joseph

I'm trying to setup a simple tcp sip connection based on the toronto osaka example in the Asterisk book. On the remote box (osaka) (1.8.9.0-rc1): [toronto] type=friend transport=tcp secret=welcome context=toronto_incoming host=dynamic disallow=all allow=ulaw sip show peer toronto * Name : toronto Secret : <Set> MD5Secret : <Not set> Remote Secret:

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'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