[asterisk-users] DECT with handover

at Polycom's wireless servers but I couldn't figure out exactly how they work. Does the KIRK Wireless Server 300 support handover in a asterisk compatible way? Best regards, Paulo Santos -- _ -- Bandwidth and Colocation Provided

Re: [asterisk-users] maximizing sound quality in 10.0

On Ter, 2011-12-27 at 13:34 -0600, Danny Nicholas wrote: Hi list, I have a set of 300 or so WAV files I was combining and playing using playback/background in 1.4.X. Now that I have moved on to the 10.0 set, I understand that I can replace my 8 Khz mono files with virtually

Re: [asterisk-users] Use different local IP for each SIP trunk

it. In my case, the operator installed a gateway with a dedicated line and it's connected to the local network, but instead of being 192.168.0.0 it's on 10.0.0.0. So I use this 2 networks in the same NIC in the asterisk machine. Best regards, Paulo Santos

[asterisk-users] Asterisk 1.4.x segfaulting daily

, including the BRI card (B400P, OpenVox). Also tried different versions of Asterisk, Dahdi and mISDN. I'm stuck with 1.4 Asterisk branch and mISDN v1. Best regards, Paulo Santos Core was generated by `/usr/sbin/asterisk'. Program terminated with signal 11, Segmentation fault. [New process 21726

Re: [asterisk-users] Asterisk 1.4.x segfaulting daily

recent versions with tons of bugs fixed. Unfortunately I can't do that, at least not now. I will, however, migrate it eventually to either mISDN v2 or Dahdi, depending on the state of Dahdi then. P.S.: Attached the log. Best regards, Paulo Santos [Dec 12 16:38:36] DEBUG[22160] chan_sip.c

Re: [asterisk-users] ISDN - Busy signal on 3rd call

Paulo Santos wrote: Hello, Following my first mail about this issue [1], I think I know now what the problem is. When I have both lines being used and a third call comes in, the person calling doesn't get a busy tone, he gets something like line unavailable. I've been debugging mISDN

[asterisk-users] Phones don't stop ringing

the first 180 Ringing _after_ the call is answered. This can be a network issue or a buggy firmware on the phones, but either way, shouldn't Asterisk send a CANCEL to an INVITE even if the phone didn't send 180 Ringing? Thanks in advance. Best regards, Paulo Santos

Re: [asterisk-users] Module reload

dahdi? No, but there's probably something wrong with your chan_dahdi.conf. Maybe in the previews line. It would help if you could show us your chan_dahdi.conf Best regards, Paulo Santos -- _ -- Bandwidth and Colocation Provided

Re: [asterisk-users] ISDN - Busy signal on 3rd call

much have no idea what they're talking about I'll call them again and confirm those features are actually disabled. Best regards, Paulo Santos -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New

Re: [asterisk-users] Module reload

Hello, Flavio Miranda wrote: Asterisk:/var/log/asterisk# pico /etc/asterisk/chan_dahdi.conf [...] You're missing the context [channels] at the start. Best regards, Paulo Santos -- _ -- Bandwidth and Colocation Provided

[asterisk-users] ISDN - Busy signal on 3rd call

. Anyone knows how can I force asterisk to send cause 16 or 17 in this situation? Thanks in advance. Best regards, Paulo Santos misdn.conf: http://pastebin.com/FmgECqkU misdn debug: http://pastebin.com/Tg6wPKBD [1] http://www.mail-archive.com/asterisk-users@lists.digium.com/msg244330.html

[asterisk-users] ISDN - Busy signal on 3rd call

83 39 31 36 33 39 31 37 34 32 70 03 c1 38 34 I've changed misdn.conf so it sends a release cause as 17 (user busy), but I get the same behaviour - cause:0 ocause:0. Anyone knows how can I force asterisk to send cause 16 or 17 in this situation? Thanks in advance. Best regards, Paulo

[asterisk-users] BRI line issue on third call

anyone have any idea what can be causing this? Thanks in advance, Best regards, Paulo Santos -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar

[asterisk-users] Can't place 2nd call to provider

is not an issue. I have another Asterisk running with the same configurations on another place and with the same provider and I don't have this issue. What can be the problem? Best regards, Paulo Santos ___ -- Bandwidth and Colocation Provided by http://www.api

[asterisk-users] Sound quality issue

have QoS/ToS/whatever for me to test it. Plus, the phones are 3 switches away from the other phones. Thanks in advance. Best regards, Paulo Santos ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15

[asterisk-users] Sound through NAT issue

. The problem is that I can't ear anything nor can the phones inside the network phone the outside phone. Is there any port I'm forgetting to forward? Best regards, Paulo Santos ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com

[asterisk-users] Asterisk as a gateway

. I'm using alaw codec on every call. Does anyone have any idea what this problem could be? Thanks in advance, Best regards, Paulo Santos ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list

Re: [asterisk-users] Asterisk on AVR32

? Or does it do some more operations? Thanks everyone, Best regards, Paulo Santos ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman

[asterisk-users] Asterisk on AVR32

for that matter. Does any one have any idea on how to do this? Thanks in advance, Best regards, Paulo Santos [1] menuselect/menuselect --check-deps menuselect.makeopts /bin/bash: menuselect/menuselect: cannot execute binary file make[1]: *** [menuselect.makeopts] Error 126 make[1]: Leaving directory

[asterisk-users] Digits timeout (ISDN)

, Best regards, Paulo Santos ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Digits timeout (ISDN)

Paulo Santos wrote: I was wondering if something like Set(TIMEOUT(digit)=5) would work in this situation. Found out Waitfordigits is needed in these situations. To make it available on asterisk I just downloaded app_bundle [1] and a simple make make install did the trick. Best regards

Re: [asterisk-users] A400P + Intel D201GLY2(A) motherboard?

Gilles wrote: Hello I'd like to build myself an Asterisk server for SOHO use. Intel's D201GLY2 motherboard (http://tinyurl.com/ddarzp) looks like a very good deal, but I'm concerned about two things: 1. Will an A400P (from OpenVox, but supposed to be Digium-compatible

Re: [asterisk-users] A400P + Intel D201GLY2(A) motherboard?

Paulo Santos wrote: I managed to do 10 calls per second, lasting 5 seconds each. 10 or 5, I can't remember... ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options

Re: [asterisk-users] BRI/ISDN, misdn.conf/misdn-init.conf, OpenVOX B100P and Etisalat in Dubai

. Regards, Paulo Santos -- HTML e-mail is evil. Go plain text. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk

Re: [asterisk-users] CDR - Asterisk-Stat and PHP5

Tiago Durante wrote: Hi all, I don't know if its the right place to ask, but... Does any one have the asterisk-stat-v2 running with PHP5? Tks! # php --version PHP 5.2.0-8+etch13 (cli) (built: Oct 2 2008 08:26:18) Copyright (c) 1997-2006 The PHP Group Zend Engine v2.2.0, Copyright

Re: [asterisk-users] CDR - Asterisk-Stat and PHP5

Marco Signorini wrote: Hi Tiago. I've it working on PHP 5.2.6 but only after having modified the php.ini default configuration keys: zend.ze1_compatibility_mode = Off short_open_tag = Off Though my zend.ze1_compatibility_mode is set to Off, short_open_tag is set to On and it is working.

Re: [asterisk-users] DTMF tones mid conversation

a Zap channel, but I'll run some more tests. asterisk 1.4.17 / addons 1.4.7 / zaptel 1.4.12.1 / mISDN 1.1.8 Paulo Santos ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update