search for: endpoints

Hi, I'm still struggling with dovecot and Unixware: On my main server, I have two problems: 1- imap-login has great difficulties to connect to 'default'. I have traced it to accept in auth_master_listener.c witch fails with errno=71 (EPROTO) weird... 2- It some times works and I can log-in but imap gets killed by signal 6 (abort?) See the logs: Jan 24 12:40:25 dovecot: Dovecot

Dear R - Users I have some problems fitting a linear mixed effects model using the lme function (nlme library). A sample data is as shown at the bottom of this mail. I fit my linear mixed model using the following R code: bmr <-lme (outcome~ -1 + as.factor(endpoint)+ as.factor(endpoint):trt, data=datt, random=~-1 + as.factor(endpoint) + as.factor(endpoint):trt|as.factor(Trial),

Hello to all !!! I have some issues with some users that login to the same account through imap from many devices like: mobile,desktop/tablet !!! Issues like: 1. When a new email is arrived is not appearing to all devices... 2. New email is arriving to some other devices after a long period of time ... Generally issues like that .... Does anyone have an idea of how to solve this ..?

Hello there Larry ! No it is not a client issue... Because i have tried many clients and all the same issue ! Actually the 1st client that refreshes the new email appear all others not ! I have tried RoundCube/Outlook/Thunderbird/Android Email... etc All have the same issue ..? -----Original Message----- From: Larry Rosenman [mailto:larryrtx at gmail.com] Sent: Tuesday, July 18, 2017 8:51 PM

Greetings all, Anyone done any work on these products? I have several (a couple of different models) that are rebranded Ultra in the US. Using explore with usbhid returns some significant quantities of data and I've done some captures with usbsnoop. I've also asked Ultra to get me the HID definition that was used for them, in the hopes I don't have to usbsniff all the setting

...mail.com > <mailto:serov.d.p at gmail.com>> wrote: > > Hello. > > Asterisk 13.2. > I transfer configs from chan_sip to res_pjsip. > In chan_sip i have "match_auth_username=yes" and have nothing in > pjsip. > > I have a lot of endpoints and registrations on same SIP server. > And it's problem in pjsip now. Is not it? > > I requesting to add new value for endpoint option identify_by. The > value 'uri'. > Simple config (cutted): > > [siptrunk] > type=registration > t...

Dear R-users I am relatively new to R, i hope my many novice questions are welcome. I have problems accessing some objects (specifically the random effects, correlation structure and variance function) from an object of class gls and lme. I used the following models: yah <- gls (outcome~ -1 + as.factor(Trial):as.factor(endpoint)+

Currently using PJSIP. First, they want me to get this working with the existing PJSIP configuration, but then setup a second box using chan_sip performing similar work. For PJSIP... I currently have an endpoint configured to a system using IP based authentication. It is configured with a match setting in the endpoint section. All channels coming from that IP address go to this endpoint. They

List, Regarding a problem I'm having that I can't login to our domain, after searching google I checked my smbd log file. I pasted the log file below. Please know that I could see the samba server in my network places just fine before. Now when I double click on the domain I get the error message that the path was not found. I'm running w2k. Thanks, [2007/04/06 08:41:39, 0]

I am running asterisk 13.1.0 In pjsip.conf, the endpoint section has an aors and an auth field. I can name the auth field anything I want. The key is to set the auth=field accordingly. However, when I try this with the aors field, it never works. It seems I have to name the aors=field to match the name of the endpoint section. Is this correct? Would there ever be a need for multiple aors to

Hello, I`m try to make the attended transfer work Dlink DG-104S via FLASH, when somebody calls my phone I pickup and press flash to get a second line to call another extension. When I press flash I hear no dialtone, and only a long and then small beep. When I try to dial digits I hear again those long+short beeps, but the extension dialed is not ringing. If I pres flash again I get back to

...g external media address to 152.X.Y.Z This message is clearly lacking from the debug session where the incorrect media address is sent.  But there's not enough detail in the debugs to see why this decision was not made to use external_media_address By default we use nat settings for all our endpoints, but obviously it's not required here for an ITSP that has trustworthy media ports in the SDP.  Maybe a bandaid is turning off rewrite_contact for this endpoint?  Going to try that as soon as possible. Why is external_media_address not being used all of a sudden?  Has anyone else seen this...

...gcp compat with NCS -- anything I can do to make it compat?? This is what happens - below *CLI> mgcp reload Reloading MGCP == Parsing '/etc/asterisk/mgcp.conf': Found Use EXIT or QUIT to exit the asterisk console == MGCP Listening on 10.1.22.39:2427 == Using TOS bits 0 mgcp show endpoints Gateway '001099008521' at 10.1.22.81 (Static) -- 'aaln/4@001099008521 in 'default' is idle -- 'aaln/3@001099008521 in 'default' is idle -- 'aaln/2@001099008521 in 'default' is idle -- 'aaln/1@001099008521 in 'default' is idle *CLI&...

...gs ! My goal is to get Twilio trunking working, and with TLS/SRTP. I see this concerning message in my log: [Feb 7 16:50:26] ERROR[20596] res_sorcery_config.c: Could not create an object of type 'endpoint' with id ?twilio' from configuration file ?pjsip.conf? Thus, ?pjsip show endpoints? does not show the endpoint for the Twilio trunk. Hoping for a sanity check of my pjsip.conf file, and what could be causing this. A test call form Twilio?s system hits the PBX (over TLS), but always says ?No matching endpoint found? in the asterisk log. pjsip.conf [transport-tls] type =...

...when we purge the sorcery memory cache it starts finding the aors > but then it says it cant find the auths. > > The wired thing is when it says it can't find the aors and auths entries > it does not show it is looking for the values in the aors and auth fields > from the endpoints tables. It keeps putting the value from the endpoints > id > field as the entries it can't find. > > One point of note the tables we used and created for pjsip back when we > setup the 13.12.2 version are not what is currently being created when we > run alembic now.....

Hi, Thank you for the great OpenSSH package. I obtained 1.2.2 and tried it on linux 2.2.14. I had reported earlier that I saw "Transport endpoint is not connected" warning messages were observed on the linux 2.2.14 kernel with earlier release of openssh. The same happens when I used the port-forwarded HTTP connection. But this time, after accessing such http connections, I noticed

Hi, in following of a recent discussion I got to work on MGCP with the Cisco ATA186 again, and got it to work very nicely. However, there is a little thing with transfers I would like to get comments on: Call comes in from PSTN and goes to an ATA186 (MGCP) Call is answered and then, using flash, transferred to another extension If the extension is available, there can be an announcement and

I just realized that you are asking about dynamic AORs, not static Contacts in an AOR. That may be the difference. I have never actually tried giving a dynamic AOR a different name. And you wouldn't want more than one dynamic AOR, you'd just use an AOR that allowed more than 1 contact. On Wed, Apr 1, 2015 at 2:59 PM Trey Hilyard <kctrey at gmail.com> wrote: > I don't know

...06 PM, Dmitriy Serov <serov.d.p at gmail.com> > wrote: > >> Hello. >> >> Asterisk 13.2. >> I transfer configs from chan_sip to res_pjsip. >> In chan_sip i have "match_auth_username=yes" and have nothing in pjsip. >> >> I have a lot of endpoints and registrations on same SIP server. And it's >> problem in pjsip now. Is not it? >> >> I requesting to add new value for endpoint option identify_by. The value >> 'uri'. >> Simple config (cutted): >> >> [siptrunk] >> type=registration &g...

Hello. Asterisk 13.2. I transfer configs from chan_sip to res_pjsip. In chan_sip i have "match_auth_username=yes" and have nothing in pjsip. I have a lot of endpoints and registrations on same SIP server. And it's problem in pjsip now. Is not it? I requesting to add new value for endpoint option identify_by. The value 'uri'. Simple config (cutted): [siptrunk] type=registration transport=udp-transport outbound_auth=siptrunk server_uri=sip:sip.exam...