similar to: strange caller display

Hi all, I have installed X-lite client on a windowsXP machine and asterisk on an enterprise linux m/c. The client is sending a registration message to asterisk server. It is able to process the message and sends 200 OK back. But later it says "Scheduling destruction of sip dialog xxxx ". Then it says "Really destroying sip dialog xxxx". What to do for this problem??? I

Hi, Recently we got a new feature request from our customer, they want a report to list the duration that agents putting customer on hold, they want to base on this to measure the agents performance. I cannot find any events in cdr, message logs, or manager interface, only when I enable sip debug, then I can see the ReInvite Event in the cli , some thing like the attached logs, is there any

Hi, I've been doing some testing on moving from 1.2 to 1.4 and one issue I've encountered is re-transmits whenever an INVITE is cancelled. I have a stateless SIP proxy in fron of my asterisk servers (all it does is direct requests to one asteisk server or another) and the re-transmits do not occur on 1.2.17 which is the current verion I have in use on my production servers. The

I have a new asterisk install (1.0.7) - and in case it's relevant I'm not using autoload option in modules.conf. Generally all is working well. However, when I make a call from my softphone and try to leave a message, the message is cutoff after a few seconds (whenever I pause for 1 second between words). Strangely, when I use an analog phone connected to my ATA, I can record as long as

I set up my * server using its publc IP address. Now that i switch over to using the domain name, X-Lite can't log in. =========With Domain Name (doesnt work)============ Transmitting (NAT) to 85.250.206.46:6007 <http://85.250.206.46:6007>: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 85.250.179.93 <http://85.250.179.93>

Hi; I am running asterisk 1.6 from Fonality (Trixbox PRO). I am trying to initiate a call FROM a softphone client to asterisk (either an internal 4 digit extension call) or an outside line via a SIP trunk. In both cases, asterisk rejects the call with a 420. In this case, it?s a call from x3992 to x4415 Does this require a change on the softphone for x-call-detail? <--- SIP read

I have a SIP account that I can successfully register with XTEN and a Sipura-2000. I have yet to be able to get it to authorize with *. My XTEN looks like: Username: 001234 Password: xxxx Authorization Username: 001234 Domain: domain.net Register with domain:

The SIP trunking service that I am trying to set up keeps saying that my registration from Asterisk is invalid. Asterisk registration: REGISTER sip:{registration_ip} SIP/2.0 Via: SIP/2.0/UDP {asterisk_ip}:5060;branch=z9hG4bK5c2eb10c;rport Max-Forwards: 70 From: <sip:{registration_user}@{registration_ip}>;tag=as5579cc0c To: <sip:{registration_user}@{registration_ip}> Call-ID:

Hi list, How do I make Asterisk 13 (using PJSIP channel) to handle PUBLISH sent from the phones? The trace looks like: ## PHONE -> ASTERISK ## PUBLISH sip:1001 at example.com SIP/2.0 Via: SIP/2.0/UDP 172.31.19.250:2048;branch=z9hG4bK-w2orn21sre9u;rport From: "1001" <sip:1001 at example.com>;tag=98slbhbn16 To: "1001" <sip:1001 at example.com> Call-ID:

REPOSTED with MORE Info and Modified Subject Line: -------------------------------------------------------- I am using one of the Minute Provider to dial out USA numbers. Now in one of my process, we need to Dial IVR and the enter DTMF digit and then it connects to the automated IVR. When I dial out the IVR directly using Xlite and VOIP Mins provider , it works perfectly. but when In try from

Thanks, chdir dir /data error seems go away, but at 12:00 o'clock the following error appear: Feb 20 11:45:03 brulx01 vsftpd: Mon Feb 20 11:45:03 2006 [pid 12273] [job260] OK DOWNLOAD: Client "10.90.1.1", "/data/mde/DATA/.././ACTUAL/mdeact260.msg", 12481 bytes, 174.76Kbyte/sec Feb 20 11:56:10 brulx01 smbd[12410]: [2006/02/20 11:56:10, 0]

On 09/04/16 11:55, Lists wrote: > nop, the same message. > > I made all change, the tests are success but the same message. > > ----- Αρχικό μήνυμα ----- > Από: "Rowland penny" <rpenny at samba.org> > Προς: "samba" <samba at lists.samba.org> > Απεσταλμένα: Σάββατο, Απρίλιος 9, 2016 1:40:50 μ.μ. > Θέμα: Re: [Samba] kinit succeeded but

On 09/04/16 11:30, Lists wrote: > Ok I am sorry was a test I send you again the smb.conf > > # Global parameters > [global] > netbios name = SOLFS > security = ADS > workgroup = SOLAE > realm = SOLAE.LOCAL > > log file = /var/log/samba/%m.log > log level = 1 > > dedicated keytab file = /etc/krb5.keytab

Taking this back on list where it belongs: On 09/04/16 10:31, Lists wrote: >> Are you 100% sure it is off, even better would be to remove it (or >> change '.local' to something else) > yes I am. > > systemctl list-unit-files | grep avahi > avahi-daemon.service disabled > avahi-daemon.socket disabled > OK

IVR Number :17275691533 When I try it from xlite configuring my provider directly, it works perfectly. When I try to dial out from dialer , it doesnt work. [sip8] type=peer username=user fromuser=user authuser=user secret=password host=8.14.146.111 nat=no canreinvite=yes insecure=very disallow=all allow=g729 allow=ulaw context=default dtmfmode=rfc2833 What cld be the reason ? --------------

On 09/04/16 11:09, Lists wrote: > First at all the ip of Samba AD DC is 10.0.0.22 and the smb.conf of this ad server is the following: > [global] > workgroup = SOLAE > realm = SOLAE.LOCAL > #security = ads > # Use password server option only with security = server > #password server = solad.solae.local > netbios name = SOLAD > server role = active directory domain

Hi, I can access our server with \\10.0.0.25 \\lampadas \\lampadas.stjamesthegreat.org All three ways will show me a list of shares. The first two will allow me to access the shares but the last fully qualified domain name results in "\\lampadas.stjamesthegreat.org\WPKG is not accessible. You might not have permission to use this network resource. Contact the administrator of this