similar to: Problem logging in using domain

Folks, I've been trying to handle the problem where blocked callerids appear as coming from asterisk <asterisk> on the email notification, and the message envelope simply doesn't say anything (does not actually play the vm-unknown message). So, following the tip provided by several previous posters, I tried putting this in my extensions.conf (the xx's are my DID,

Dear r-helpers, Here I would like to have your kind helps on generating Venn diagram. There are some packages within R on this task, like venneuler, VennDiagram, vennerable. But, vennerable can not be installed on my Mac book. It seems VennDiagram can not work on my data. And, venneuler may have generated a wrong Venn diagram to me. Do you have any experience/expertise on those Venn diagram?

Hi, I am trying to get the following two timeseries (these are small subsets of the whole thing) into R so I can merge them using zoo. Timeseries 1=[ Date Count 9/28/2003 1505 10/5/2003 1535

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 all, I am using asterisk1.2.14,realtime and I find there is a strange case in the receiver's display. I have a dial plan to route a call to the destination. I haven't set the callerid(num) for the caller. In the receive ends, it's display shows "asterisk" when I make a call to the receiver. I wonder why "asterisk" shows in the display as I haven't set

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 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:

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

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

Alan Hourihane (1): avoid checking for fds_bits on mingw Colin Harrison (2): Use winsock2.h Use Sleep() instead of sleep() on windows James Cloos (2): Fix typo in XF86Keysym.h Add more dead key syms Jeremy Huddleston (3): Apple: Cleaned up some Apple definitions Apple: Define _DARWIN_C_SOURCE otherwise _XOPEN_SOURCE, _POSIX_SOURCE, or _POSIX_C_SOURCE