similar to: Failed to authenticate user

Does anybody know if there is an app that will cause similar to occur on users PC? I have a scenario where users will have snom phones on their desks. Ideally when their phone receives a call I need to popup a web browser with a specific url. Any ideas appreciated. Neil on 5/7/05 10:52 PM, Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> wrote:

> -- Started music on hold, class 'default', on > SIP/193.111.200.67-0815c790 > -- outgoing agentcall, to agent '1001', on 'Local/201@sip-0add,1' > -- Called Agent/1001 > -- Executing Dial("Local/201@sip-0add,2", "SIP/101|20|tr") in new > stack > -- Called 101 > -- Agent/1001 is ringing > --

Hi Thomas, Really Thanks for your advice. I have tried your suggestions but problems still remain. My findings is that some Windows, regardless of version, can map to Samba through dial-up networking but some just cannot. I try to find the difference between these Windows but failed. Regards, Neil Thomas Cameron wrote: > Neil - > > The problem you are having is possibly from one of

I made the observation that I'm able to make a call with my SIP client (kphone) even when I'm not registered/authenticated. Of course, when I'm not registered at asterisk, people can't call me, but it's still a huge security hole, that unregistered Clients can make calls. Is there a way to tell asterisk to only allow registered clients making calls? I know about the

Hi all, I'm trying to tighten things up a bit and I seem be be running into something that doesn't make sense to me. I've got 2 contexts, one for customers, and one for guests, that I include into [customers] and [default], in extensions.conf, as below: ============================================================= [default] include = dial_GUEST [customers] include = parkedcalls

I've been using a couple of Polycom 501 phones in my home Asterisk setup. I set up each phone in sip.conf to be static, i.e. host=<phone ip address> so that registration wasn't required. This has worked fine for me for a couple of years. Now I just bought a Polycom 335. Since the 501's are now obsolete, I had to go through the steps required in order to have separate

Ok I think we have some common ground - CSE should choose the aliased pointer over the non-aliased one because we don't want the no-aliasing information to creep outwards from the inlined callsite. I'll put together a patch in the coming days and add y'all as reviewers so you get visibility. Cheers, -Neil. On Wed, Jan 22, 2020 at 4:47 PM Jeroen Dobbelaere < Jeroen.Dobbelaere at

At a high level, EarlyCSE should be intersecting the metadata of instructions that it combines. If it doesn't, and also doesn't drop the metadata, that seems like a bug, regardless of anything else. On 1/22/20 9:30 AM, Jeroen Dobbelaere wrote: Hi Neil, Hall, - as far as 'C' is concerned, this is input code is valid, as the pointers are not used to modify objects. - as far as

Hello, I am trying to run exim 4.68 and dovecot 1.0.13 on Solaris 10 x86 5/08 using dovecot lda and sieve with virtual users and domains, tls and ssl. At the moment certificates are from my internal CA Exim and Dovecot, dovecot lda and dovecot sieve were downloaded and installed from Blastwave. Mail delivery (ie from Exim to dovecot via dovecot lda) is working correctly but when I try

I've got a bunch of Win XP Pro machines, and I setup domain logins to the samba server so I'd have roaming profiles, etc. Alas, I've now discovered that windows doesn't let you use fast user switching when you do domain logins. I really want fast user switching -- is there a way to configure samba / winXP to fake some the domain login features? I don't care about password sync

I'm thinking of starting a partyline, where people call in and talk to other people. For record keeping and billing purposes, I'd like to go by the callers telephone number. This method works fine as long as the caller doesn't have callerid blocked, but what are my options if they do block their number? I know there must be a way to report it, because there is a service provider here

On Thu Mar 28 2019 17:04:37 GMT-0400 (Eastern Standard Time), Patrick Mahan via dovecot <dovecot at dovecot.org> wrote: > Hmm, actually it is set - > > root at ns:/usr/local/etc/dovecot # dovecot -a | grep auth_username_format > auth_username_format = %Ln Use doveconf, not dovecot (although they may do the same thing). doveconf -a just shows you ALL settings, regardless of

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

On Mon, Feb 10, 2020 at 11:56:06AM +0800, Jason Wang wrote: > +/** > + * vdpa_register_device - register a vDPA device > + * Callers must have a succeed call of vdpa_init_device() before. > + * @vdev: the vdpa device to be registered to vDPA bus > + * > + * Returns an error when fail to add to vDPA bus > + */ > +int vdpa_register_device(struct vdpa_device *vdev) > +{

On Mon, Feb 10, 2020 at 11:56:06AM +0800, Jason Wang wrote: > +/** > + * vdpa_register_device - register a vDPA device > + * Callers must have a succeed call of vdpa_init_device() before. > + * @vdev: the vdpa device to be registered to vDPA bus > + * > + * Returns an error when fail to add to vDPA bus > + */ > +int vdpa_register_device(struct vdpa_device *vdev) > +{

Hi, I've just got a TE110P card and installed at Asterisk. I configured zapata.conf, according to www.digium.com/index.php?menu=configuration, but the following error is happening: ... ... ... [chan_phone.so] => (Linux Telephony API Support) == Parsing '/etc/asterisk/phone.conf': Found == Registered channel type 'Phone' (Standard Linux Telephony API Driver)

I am new to Asterisk and have a question about a problem that really confuses me. I am running Asterisk 1.4.15 at site A and 1.4.10 (from ubuntu repository) Both are NATted. I set up a IAX2 connection between two asterisk boxes (A and B). During my tests I used a laptop with zoiper and a username of laptop (on site B) with a (IAX2) connection to the remote asterisk box A. I do not know whether

I can change the sip port to any number, and when I unload and reload chan_sip.so, I always get == Registered channel type 'SIP' (Session Initiation Protocol (SIP)) == Parsing '/etc/asterisk/sip.conf': Found == SIP Listening on 64.1.16.172:5060 == Using TOS bits 4 == Parsing '/etc/asterisk/sip_notify.conf': Found == Registered application 'SIPDtmfMode' Is

I'm experiencing a "Failed to authenticate on INVITE" error (see output below) whenever I try to MAKE a call through the Broadvoice account. I noticed some others had the same problem but it went away when they rebuilt Asteris w/ a new version. N such luck for me! I'd be grateful for any assitance. Here's what I've done so far: 1) I downloaded the latest stable

So, tweaking configs, rebuilding this and that... restarting, twiddling, it works (yeah!), but fails on re-boot to work at all. Consistently, though. I believe it comes down to this: I can call out only *after* I've received a call. So, cold boot. Then: modprobe dahdi modprobe wctc4xxp modprobe wcfxo dahdi: Telephony Interface Registered on major 196 dahdi: Version: 2.1.0.3