similar to: strange SIP authentication/authorization behaviour

Hello, is there a way to disable call waiting in sip? I`m using grandstream 101 and even when the phone is in use I hear ringing in the headset. It is pretty annoying , is there some way to disable this? I cant find anything like it in the grandstream docs. Thanks -- Anton Yurchenko<phila@dg.net.ua> Digital Generation

hello, I have a test setup with 2 asterisk servers, each having a one snom 100 via sip using it. I`m experimenting on how trunking between them would work. I have them setup for RSA authentication which I plan to use in the future. So I`ve setup the keys and servers seem authenticate to each other. One is named phila and other hurricane. Here is what I see on phila: -- Registered

Hello, Anybody has it working with asterisk? Could you share your experience ( good/bad) Thank you -- Anton Yurchenko<phila@dg.net.ua> Digital Generation

Hello, Right now I have app queue working with incominglimit=1, there is no call waiting signal, but after a while( like couple of hours) some phones randomly get stuck. The * thinks that they are in use and doesnt ring them, when they are infact not in use. sip show inuse, shows that they are inuse. typing reload on the console resets this and they are again available for working. anybody

Hello, I`m trying to make a call from the snom 100( SIP mode) but whatever number I dial I get a 404 error from Asterisk. Here are my configs and a dump from "sip debug" . But if I make a call from a Zap line (see extension 2382031), it rings the snom phone sip.conf: ------------------------------------------------------------------------------ ; ; SIP Configuration for Asterisk

I just downloaded the cvs version CVS-07/28/03-14:45:19 and now I cannot recieve the the calls from the zaptel interface which is a E100P with pri signaling. That is something with asterisk becouse rolling back to version from 06/23/03 using the new libpri and zaptel fixes the problem. Here is an exept from the config: [macro-stdexten]; ; ; Standard extension macro: ; ${ARG1} - Extension

Hello, Anyone aware of any CRM products projects that intagrete with *? Or that integrate with any telephony products? Is there some open API for such integration, or are they all proprietory? Thanks -- Anton Yurchenko<phila@dg.net.ua> Digital Generation

Hello, Has anybody been able to get the Mute button work on grandstream? it simply does nothing. Only Hold is avalable, which is not that good. Thanks -- Anton Yurchenko<phila@dg.net.ua> Digital Generation

Recently I wrote about this problem, but it still exist and I can't dial my Xlite SIP Phone So here is the Notice Jul 5 17:14:07 NOTICE[65541]: chan_sip.c:6731 handle_request: Registration from 'Damian Minkov <sip:damian@10.1.1.2>' failed for '10.1.1.11' The * box(10.1.1.2) and the PC(10.1.1.11) on which is the XLite are in the same network Here is part from sip

I added the playback line to my [macro-stdexten] context but when I dail an extension I don't get the "please hold while I try that extension" message. It just dials the extexsion. Do I have a syntax problem somewhere ? exten => 8005,1,Macro(stdexten,8005,Zap/2) exten => 8006,1,Macro(stdexten,8006,Sip/8006) [macro-stdexten] ; ; Standard extension macro: ; ${ARG1} -

Hello, Can such thing be done through dialplan , that say I transfer a call to an extension but it is busy, so that this call returns back to me. Thanks -- Anton Yurchenko<phila@dg.net.ua> Digital Generation

Hi folks, I want to fix hole in my asterisk set up. I use Vocal as my sip proxy and * for voice mail and the g/w to PSTN, Iconnect, fwd etc. So from Vocal I redirect sip requests which needs to go 'other' places. This senario works fine. Now the issue is someone else running a vocal or another SIP proxy can redirect his calls to my * as well. Those calls two will come through general

Hello, I have an E100P, and in the zaptel.conf I have: span=1,1,0,ccs,hdb4,crc4,yellow fxsks=1-10 the light on the card is green( BTW what do all those states of the card that zttool reports YELLO, RED, BLUE ..., is there a doc for zttool?, or for the card?) in the asterisks` zapata.conf I have: [channels] context=default switchtype=euroisdn signalling=fxs_ks usecallerid=yes hidecallerid=no

I have a simple barplot of 4 mean values, each mean value has an associated 95% confidence interval drawn on the plot as an error bar. I want to make a legend on the plot that uses the error bar symbol, and explains "95% C.I." How do I show the error bar symbol in the legend? I could not find any "pch" values that are appropriate Thanks [[alternative HTML version

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

Anybody has figured out why asterisk + snom have such bad quality using GSM? When I use GSM I see such messages dumped on asterisk console: WARNING[14351]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 2 frames WARNING[14351]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 2 frames WARNING[14351]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect

********************************************************************** This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. ********************************************************************** Hi Just installing R and some

I have read the R manual and help archives, sorry but I'm still stuck. How would I do a t-test with an adjusted p-value? Suppose that I use t.test ( ) , with the function argument alternative = "two.sided", and data such that degrees of freedom = 20. The function calculates a t-statistic of 2.086, and p-value =0.05 How do I then adjust the p-value? My thought is to do

I have a PolyCom Soundpoint 500 sip phone. I'm tring to get the phone registered on an asterisk box but am having no luck. I get the following errors 192.168.22.196 being the phone and 22.254 being the asterisk box.. Apr 23 11:41:33 NOTICE[1133742896]: chan_sip.c:5623 handle_request: Registration from '"110" <sip:192.168.22.196@192.168.22.254>' failed for

Martin Devera wrote: > If you read the manual, the algorithm will not work correctly > with {,c}burst < MTU ... > devik > I just tried to change {,c}burst to 1600, or leaving them by default but no visible result. here is the latest tc -s -d class show dev eth0 class htb 1:101 parent 1:1 prio 0 rate 40Kbit ceil 40Kbit burst 1599b/8 mpu 0b cburst 1599b/8 mpu 0b quantum 512 level