similar to: Asterisk localization

Displaying 20 results from an estimated 10000 matches similar to: "Asterisk localization"

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 May 20
0
FW: SIP-firewall problem?
Thanks for the firewall hint. Actually, most of my UDP and TCP port are open. I believe the problem is related to NAT. Here is the reasoning: - I call from SJPhone using a real IP address (e.g. 200.22.33.44) - The call reaches my router (with a real IP address of 200.33.44.55), which routes the packets to my * server at the fake IP 192.168.0.200 - My * server places a call through my TDM
2003 Dec 11
0
FW: Iax, Iax2 and Iaxcomm
Talking to myself ... ;-) Solved this by ... disallow=all allow=gsm ;allow=ulaw ;allow=alaw -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Paulo Mannheimer Sent: quinta-feira, 11 de dezembro de 2003 09:02 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Iax, Iax2 and Iaxcomm Hi, I'm trying
2003 Dec 09
2
Erratic DTMF on E1/PRI (continuation of Strage bip on ISDN/PRI)
At the same site, DTMF recognition is functioning badly, sometimes duplicating digits and sometimes totally missing others. We have checked already /proc/interrups, there is no interrupt being shared. Our zaptel has .. span=1,1,0,ccs,hdb3 On zapata we have ... switchtype=euroisdn signalling=pri_cpe relaxdtmf=no (yes doesn't seem to help) -----Original Message----- From:
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 May 07
1
SIP-firewall problem?
Hi, I have asterisk working well on my intranet, using SJPhone to make SIP calls. Everything works fine except when I try to connect from outside my intranet. I've opened the 5060 UDP port on my D-link firewall/router and, although I can dial, complete calls and hear the other party, my voice gets lost (the other party does not hear me). I've took a look on the >sip debug output
2003 Jul 22
0
new voicemail messages
Hi, I'm localizing the voicemail messages to Portuguese. To make it possible for another person to translate it, I've set up a couple of extensions that call the following macro for each message on the system. After recording, I can perfectly hear each message using Playback. When I try to play the new recorded message using VoiceMailMain, I can't hear the new message (line goes
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 Jun 05
1
answering calls with SIP phones
Hi, I have an incoming call that I would like answered every time by a different SIP phone (out of 50). Also, some of the phone may not be available (may be turned off and thus unregistered with Asterisk). Any way of doing this? Paulo H. Mannheimer -------------- next part -------------- An HTML attachment was scrubbed... URL:
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 Jul 30
4
SCO/Linux concerns
Hello Since I am getting a bit concerned about the SCO vs IBM issue, I was wondering if can I can setup Asterisk on FreeBSD is it supported ? Are drivers for Digium cards available on FreeBSD ? Thanks Ajit ----- Original Message ----- From: <asterisk-users-request@lists.digium.com> To: <asterisk-users@lists.digium.com> Sent: Wednesday, July 30, 2003 3:05 PM Subject: Asterisk-Users
2003 May 29
0
Large Asterisk installations
Hi, Could anyone elaborate on using * on medium/large sized installations (more than 50 channels)? I'm interested in learning more about performance, stability, scaleability, fault tolerance, and what hardware are you using. Thanks! Paulo H. Mannheimer Instant Solutions +55 21 2512.7999 +55 21 9225.9395 -------------- next part -------------- An HTML attachment was scrubbed...
2003 Jun 02
0
ArrayVox problems
Hi All, I'm having problems with my dialplan and an ArrayVox SIP phone. The phone is working OK, but when I try to call him from another extension it seems that Asterisk enters a recursive state and tries to call it three times at the same time. The phone then stops working, and I have to turn it off and on to make it work again. It doesn't seem to be a problem with my
2003 Jun 30
0
stuck channel
I'm getting this intermittent problem, sometimes a zap channel gets stuck after a call. Below is a snapshot of the channel. Any ideas what can be happening? Name: Zap/1-1 Type: Zap UniqueID: 1056988772.10 Caller ID: (N/A) DNID Digits: (N/A) State: Up (6) Rings: 0 NativeFormat: 68 WriteFormat: 4 ReadFormat: 4 1st File Descriptor:
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 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 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 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