similar to: baffled by defaultuser on aastra 9133i

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:

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: >

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 =>

Upgraded from 16.2.14 to 1.6.2.15 on Fedora 13, with aastra 9133i and 57i. On 9133i and 57i: #<extension># works for a blind transfer. Xfer<extension>Xfer doesn't! All this worked on 1.6.2.14. Nothing useful on cli, verbose 3, DEBUG. Here extension 169 answers an outside call, and tries to transfer it to 145 using the Xfer button: -- SIP/169-0000009c answered

Hi All, I am running Asterisk 1.6.1.2 with realtime for SIP and recently noticed the values in the defaultuser field have been "disappearing". I can't place my finger on what is happening but it appears that when the peer de-registers the defaultuser field is cleared. I will be having a more detailed look through the logs and possibly add more logging to the database but wondered

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

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 I'm sorry. I mailed the same question again. because, it cannot be yet solved. any ideas with asterisk? [Aug 20 14:40:12] WARNING[29315]: chan_sip.c:11806 check_auth: username mismatch, have <aaaa>, digest has aaaa at 192.168.0.1[Aug 20 14:40:12] NOTICE[29315]: chan_sip.c:20479 handle_request_register: Registration from 'aaaa <sip:aaaa at 192.168.0.1>' failed for

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,

Hello, Asterisk seems to try to authenticate incoming INVITE based on the [section] in sip.conf and not the username specified. I just removed the "insecure" option from my sip.conf requesting every connection to be authenticated. I added the match_auth_username=yes in the [general] section for extra security. To make it work, I have to use the same [section] identifier as username.

sip.conf: [imsdroid] type=friend ;;auth=md5 ;;defaultuser=imsdroid secret=mysecret host=dynamic context=cloud-out qualify=60 dtmfmode=auto insecure=port,invite callerid="IMSDroid client" <imsdroid> disallow=all allow=ulaw I've tried with and without defaultuser and secret. sip show peer imsdroid: * Name : imsdroid Secret : <Set> MD5Secret :

This is a small 12 line system, internal extensions 150 - 180. I didn't have a phone on 151. Here's the sip.conf stanza: ;;[151] ;;type=friend ;;context=longdistance ;;callerid="Conf Room" <151> ;;secret=0000 ;;host=dynamic ;;qualify=yes ;;dtmfmode=rfc2833 ;;allow=all ;;defaultuser=151 ;;nat=yes ;;canreinvite=no There's no DISA. And then somehow (how???) ip address

Hi I want to know how to solve below an error case. Uac cant's change username of from and digest header. I tried to put aaaa at 192.168.0.1 on username of sip.conf.but same error returned. [Aug 20 14:40:12] WARNING[29315]: chan_sip.c:11806 check_auth: username mismatch, have <aaaa>, digest has aaaa at 192.168.0.1 [Aug 20 14:40:12] NOTICE[29315]: chan_sip.c:20479

[Jul 8 21:23:49] WARNING[4358]: chan_sip.c:10458 check_auth: username mismatch, have <6001>, digest has <1160> [Jul 8 21:23:49] NOTICE[4358]: chan_sip.c:18529 handle_request_register: Registration from '<sip:6001 at 192.168.1.4 <sip%3A6001 at 192.168.1.4>>' failed for '192.168.1.3' - Username/auth name mismatch sip.conf [6001] user=6001 type = friend

Dear All, Today I upgraded asterisk 1.6 to 1.8. As the result of this when peers trying to register to asterisk the system shows: NOTICE[24698]: chan_sip.c:23417 handle_request_register: Registration from '"50" <sip:50 at 192.168.1.109> <sip:50 at 192.168.1.109>' failed for ' 192.168.1.80:5062' - Wrong password even though on 1.6 everything was OK here is

Greetings. Attempting to connect Asterisk to LDAP database using res_config_ldap module. While trying to register sip client (Ekiga softphone), according to slapd.log, asterisk connects to LDAP server, asks for some attributes to modify (they do exist, and asterisk user has all permissions to do that, etc). And then asterisk application just crashes. Without ldap (using just static users'

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 list someone is trying to hack my server . Is there any way by whcih I can stop hacking of my server except iptables ? I want to stop on the basis of sip.conf account only. bcoz I can't apply iptables rules on server it's remote server of server provider and we used it for making voip call for customers. for the time been i have close all sip accounts. but can't stop for more then

My server is being attached all day and fail2ban is not stopping the attack. I updated stamstamp to match fail2ban requirements. [2010-12-25 18:54:34] NOTICE[15415]: chan_sip.c:21830 handle_request_register: Registration from '"7002" <sip:7002 at x.x.x.x>' failed for '38.108.40.94' - No matching peer found [2010-12-25 18:54:34] NOTICE[15415]: chan_sip.c:21830

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?