similar to: Phone manual..

Hi, I tested asterisk server with Xpro program, and all the function working well ( like 3 way calling, transfer.... ). But on the VOIP phone, I don't know press which key for 3 way calling function and transfer function... Can anybody teach me ? thanks

brett@websmyths.com wrote: > Also - an outside chance - make sure Tip and Ring > are correct. You could be getting ground loops - depends on the noise. > I am having noise and slip errors between my TE110P and a legacy PBX T1 card. Could this be the same symptom? The connection is made using a 15 pin serial on the T1 Card side to RJ48 on the TE110P side. I can't determine what the

I have been getting these for awhile now in my log files. Apr 24 11:02:38 asterisk named[1072]: lame server resolving 'pbx_loopback.so' (in'so'?): 205.166.226.38#53 Apr 24 11:02:38 asterisk named[1072]: lame server resolving 'pbx_dundi.so' (in 'so'?): 205.166.226.38#53 Apr 24 11:02:38 asterisk named[1072]: lame server resolving 'pbx_spool.so' (in

We're having a strange problem with our receptionist. She runs an xpro softphone and we're using a queue to handle incoming calls. It seems nearly all of the calls that come in through the queue get dropped. At first we thought it might have been human error (clicking the wrong button in xpro or something) or that the person waiting in the queue just gave up and hungup, however it

I am having a slight issue. I am trying to register 2 asterisk boxes with GNUGK and when I try to add the 2nd it gets denied cause of it saying its a duplicate. How do I change the configs to allow more than one asterisk box register to the same GK? brian This email was scanned by: Mcafee GroupShield ---------------- CONFIDENTIAL DISCLAMER ---------------- All information provided in this

Hi all, The firmware I used is pa168s_iax2_us_151011.bin. My problem is the handset dial before I finished key in all the numbers, no matter how fast I managed to press the keys. It appeared it always dialed immediately, for example "011862", when I actually ment to dial 0118620xxxxxxxx. Thus left the remaining numbers "0xxxxxxxx" unsent. The handset had its dial plan

Hi, I cannot do the following: telnet 127.0.0.1 5038 I get connection refused and this is preventing AMP from installing. I had this working when I was using FC3 but I had to upgrade to FC4 for another application. So I am running PHP5, MYSQL 4 with FC4 and asterisk is running (I had this problem before with FC3 and it turned out asterisk was not running) I am using 1.2.0 beta1 Asterisk

Hello all, One of the buildings I have an asterisk box deployed in is used by two small companies on two floors. They have an agreement between them whereby they'll answer each other's incoming calls and take messages if the office is empty / everyone is on the phone. Each of them has an ISDN BRI delivered to asterisk via zaphfc, then dropped into a context as follows: exten =>

Hi List, I am from Peru, I have installed an asterisk server in my company with digium card E1 TE120P, I am having issues when i make calls, here the error from my server [Oct 18 09:13:50] WARNING[2377]: channel.c:3232 ast_request: No channel type registered for 'Zap' [Oct 18 09:13:50] WARNING[2377]: app_dial.c:1106 dial_exec_full: Unable to create channel of type 'Zap' (cause

Hello All, Anyone here has experience of accepting a ilbc call and sending it on g711 or g729 I am having problem in VOICE , call goes though but there is no voice. Senario: Call is coming in from Machine A to Machine B, sending to Machine C Machine B is an asterisk box, transcoding it from IBLC to G711 and g729. Problem: Voice is not appearing on the sip user sitting on machine A Already

Hello all, I am trying to complete my dial plan and have come up with an interesting situation. My configuration is set up with 12 xlite SIP clients on SUSE linux workstation. They are calling out via 10 analog lines, TE110P->rhino 24 fxo. It all works and dials out great ... but ... this unit was brought in to handle the "global" office. So the help desk support on the Suse

-----Original Message----- I'm crazy here trying to make X-Pro use ONLY g729, and you're struggling to make it not to use it :)... Can you please indicate what's your config for X-Pro and sip.conf? sip.conf: [12345] type=user username=12345 secret=12345 nat=no host=dynamic reinvite=no canreinvite=no disallow=all allow=g729 allow=g729a allow=g723.1 allow=g726 allow=ulaw allow=alaw

Hi, Just downloaded the latest cvs from zaptel on my sparking new Athlon64 Centos4.2 system, but hitting a stumbling block... (sorry for the long post) #make cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -m64 -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -c -o gendigits.o gendigits.c cc -o gendigits gendigits.o -lm ./gendigits cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -m64

All, I'm needing volunteers to help moderate the sessions of the LLVM Developers' Meeting. All you need to do is introduce the speaker, make sure the speaker stays on time, and run Q&A at the end (run a microphone, select people, etc). Its a pretty easy job, but critical for our meeting to run smoothly. If you are interested in moderating, please send me your top 2 session choices.

Does anyone have any views on the best software base SIP client to use that normal users could use with Asterisk without being too techie ? I have tried the X-Lite client with varying success. The first version worked OK but music on hold broke the voice paths and the slightly newer version initiated the call but failed to make the voice connect in both directions. The SJphone client works but

Dear R help I have been trying to conduct a chi square test on two groups of variables to test whether there is any relationship between the two sets of variables chisq.test(oxygen, train) Pearson's Chi-squared test data: oxygen X-squared = 26.6576, df = 128, p-value = 1 > chisq.test(oxygen) Pearson's Chi-squared test data: oxygen X-squared = 26.6576, df = 128,

Olly Betts <olly at survex.com> wrote: > On Fri, Mar 30, 2018 at 05:21:43PM +0000, Eric Wong wrote: > > Hello, is there a way to optimize sorting by certain values > > for queries which return a huge amount of results? > [...] > > $enquire->set_sort_by_value_then_relevance(0, 1); > > If you're just wanting the 200 newest, it'll be faster not to

I'm part of a team working on adding an llvm codegen backend to HHVM (PHP JIT, http://hhvm.com) using MCJIT. We have a code layout problem and I'm looking for opinions on good ways to solve it. The short version is that the memory we emit code into is split into a few different areas, and we'd like a way to control which area each BasicBlock ends up in during codegen. I know this

It probably has something to do with the fact Win 95 doesn't support encrypted passwords... Brett -----Original Message----- From: Cristian Mijea [mailto:cmijea@protvcluj.ro] Sent: Wednesday, May 22, 2002 3:44 AM To: samba@lists.samba.org Subject: [Samba] !!! HELP !!! samba and win95 I'm running a smb server on a redhat 7.2 system. everything works just fine except when I try to

I've been implementing ELF/AArch64 support for lld. I can now successfully link and run a simple "Hello World" app for both dynamic and static linking. I'd like to upstream this implementation, but wanted to get feedback on how people might like to see it. Would people rather see the whole thing in one shot, each file individually or somehow break it down even farther? The