similar to: Arraycom voip phone

Displaying 20 results from an estimated 2000 matches similar to: "Arraycom voip phone"

2003 Jun 23
5
dynamic queue channels
Hi, I'm trying to build a call center application that allows attendants to come in the morning and dial a certain extension to make their extension available. I wouldn't like to use the AgentLogin app because their line would need to stay off-hook (is this correct?) Is there any SET channel status command that would allow me to do something like this? PauloHM -------------- next
2003 Oct 17
2
Beta testers for visual configuration tool f or asterisk
Count me in too. -----Original Message----- From: sip [mailto:sip@intology.com] Sent: Friday, October 17, 2003 1:56 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Beta testers for visual configuration tool for asterisk count me in ----- Original Message ----- From: "Paulo Mannheimer" <paulohm@instant.com.br> To: <asterisk-users@lists.digium.com>
2003 Jul 22
3
busydetect and random hangups
Hi, I'm having random hangup problems with zap channels. If I turn busydetect off in Zapata.conf, * fails completely to detect a user hangup in the middle of a script. On the other hand, if I turn it on, everything works much better, but long calls tend to be hung up without a motive. Any other parameter that I can try? Any #define that I can tweak and recompile? Will callprogress
2003 Dec 11
1
Iax, Iax2 and Iaxcomm
Hi, I'm trying to use iaxcomm. I can place a call from the softphone, but when I place a call to it, when I answer I get ... NOTICE[16401]: File channel.c, Line 1094 (ast_read): Dropping incompatible voice frame on IAX2[paulohm]/3 of format GSM since our native format has changed to ALAW My iax.conf looks like this .. [paulohm] type=friend host=dynamic username=... secret=...
2003 Oct 17
5
Beta testers for visual configuration tool for asterisk
Hi All, We've been developing for a while an IDE for Asterisk, and the time has come to open it for beta testers. You can check at www.instant.com.br/viv.html for a snapshot of the application. Current modules are Dialplan and VoiceMail configuration. As you may see, it is all-visual, with drag and drop support and integrated sound recording, saving and cross-checking, so you dialpland
2003 Dec 10
3
pridump
Hi All, Can anyone tell me what are the <dev1> <dev2> parameters that I should use to run pridump? I took a look at the source code but couldn't figure this one out. Best, PauloHM
2003 Sep 03
2
E1 problems
Hi, I'm testing an E1 with E&M signaling. Some of the problems I'm running into are the following: 1) if I try to configure any channel above channel 15, I start getting a "multiframe alignment error" on my telco test equipment. So I have my zaptel file only configured for 15 channels, like this span=1,1,0,cas,hdb3 e&m=1-15 2) When the test equipment tries to send me
2003 Nov 26
1
Pbx / channel bank install
Hi all, We are about to make our first channel bank install. This will be a one PRI outside connection and up to 70 extensions. As the schedule (and the budget) is pretty tight, I would like to learn a little bit more about general experiences with channel banks, like echo cancellation problems, Caller ID usage, etc. TIA, Paulohm
2004 Feb 06
4
Conference server
Hi, we are setting a 120-channel conference server and would like to learn if someone already did this (hardware, problems, etc...) Best regards, PauloHM
2003 Aug 12
1
new on E100P
Hi, I'm installing my first E100P. My zaptel reads the following: Span=1,0,0,ccs,hdb3,crc4 E&m=1-31 My Zapata.conf reads the following: Signaling = em_w Channel =1-15 Channel =16-31 After starting the zapter service I get: ZT_SPANCONFIG failed on span 1: No such device or address (6) ??? PauloHM -------------- next part -------------- An HTML attachment was scrubbed...
2003 Sep 11
3
SIP busy
Hi, I would like * to treat a SIP extension as a normal extension, when it comes to the busy functionality. In other words, if someone tries to call the SIP phone and there is already an ongoing conversation, the new caller should get a busy message/tone Is there any parameter that I can set? Is this something that should be configured at my softphone? Best, PHM
2003 Oct 29
3
Sip bandwidth usage
Hi All- I'm working on a project that will have remote (internet)access to an * server through SIP phones, either soft or hard ones. Does anyone have any experience to share about which SIP product they are using under similar conditions, as well as which codec is being used and bandwidth usage? TIA! PauloHM
2003 Dec 16
4
broken pipe - * does not respond
Hi, I?m having a serious problem at one customer. After 6 hours answering a PRI line, * stops responding in a very similar situation as described here ... http://lists.digium.com/pipermail/asterisk-users/2003-July/015391.html I took a look at "/proc/first * PID/fd" and there are hundreds of file descriptors; If I try to connect using asterisk -r I get the "broken pipe"
2003 Dec 18
1
AGI and broken pipe
Hi All, I was able to track down what I believe is a bug when using AGI services. This bug may crash your system if your extensions.conf script is intensive in using AGI services. Depending on your system's ulimit, * keeps opening files until it reaches the system limit and then stops responding. Function app_agi/launch_script seems to leave an open and unused file. Can someone confirm this?
2003 Jul 01
1
gotoiftime error
Hi folks, There was a bug with the GotoIfTime built-in command, under certain circumstances a variable contained garbage, screwing up correct time identification. I'm submitting now a patch to Mark so this can be fixed. PauloHM -------------- next part -------------- An HTML attachment was scrubbed... URL:
2003 Jul 30
1
voicemail file access problems
Hi folks, I'm having problems accessing my voicemail files through the web interface. I remember that this was discussed on the list, and it seems to be a permission problem, but I couldn't find any answer by searching the archives. Any hint? PauloHM -------------- next part -------------- An HTML attachment was scrubbed... URL:
2003 Oct 01
2
(still) channel problems
Hi folks, I'm still having the following problem, maybe someone can help me out of it. Two IDENTICAL MACHINES (same motherboard, same RH 7.2, same *) communicate through IAX2. Everything works ok on machine 1. On machine 2, if I try to use 4 fxo's from a TDM400 card, sound gets lousy. If I manually destroy one of the zap channels (e.g. zap destroy channel 4), sound gets good again.
2003 Nov 09
1
Iax2 channel usage
Hi all, In a forthcommming project, I'll have one * server tentatively calling 10 PSTN numbers through IAX2 and an * gateway. Can someone tell me if bandwidth is being used for each of these calls/channels even while my gateway tries to call and connect the destination numbers? Best, PauloHM
2003 Dec 09
1
Strage bip on ISDN/PRI
Hi All, We are just starting to deploy a new PRI IVR system, and the incoming calls sometimes get random short 'bips' while navigating our IVR menu. Any hint on what this can be? Best regards, PauloHM
2011 Mar 16
2
[LLVMdev] Prevent unbounded memory consuption of long lived JIT processes
On Tue, 2011-03-15 at 20:29 -0700, Jakob Stoklund Olesen wrote: > On Mar 15, 2011, at 4:15 PM, jfonseca at vmware.com wrote: > > > This series of patches address several issues causing memory usage to grow > > indefinetely on a long lived process. > > Thanks for working on this. > > Did you measure the performance impact of these changes? I tracked performance