similar to: No subject

FYI, not sure if it's of use to you... but... The digium tc400b is a transcoder card that can offload upto 120 channels of transcoding for g729 <-> ulaw... It's available as PCI only, but, if that's OK, it could be an alternative to replacing your server... G729 licenses are not needed when using that card... There have been posts by some people about having multiple CPU

AGI is executable. =20 Then run 'agi debug' from the asterisk cli, place a call and see what was send and receive from your agi =20 From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of James A. Shigley Sent: April-23-09 12:26 PM To: asterisk-users at lists.digium.com Subject: [asterisk-users] AGI PHP script =20 I have the

What is the problem with SIP retransmits? ----------------------------------------- Sometimes you get messages in the console like these: - "retrans_pkt: Hanging up call XX77yy - no reply to our critical packet." - "retrans_pkt: Cancelling retransmit of OPTIONs" The SIP protocol is based on requests and replies. Both sides send requests and wait for replies.

"We have created an easy and cost effective way to have customized recordings done quickly and with no hassle." I thought this was rather amusing, as: 1. If you want multiple prompts recorded, you need to submit a new order for each, which means that even prompts of a couple of words are still charged at $12. That is NOT cost effective. You could record all your prompts as a single

Happy holidays! I am experiencing an issue when trying to check my mail using IMAP. with Dovecot I have tried checking my mail using a full GUI client (Thunderbird) and telnet. Both times I get disconnected before all of my messages can be downloaded and I see an error in my mail log. Here are the details: [root at cust19-1-prod-domain userqa]# dovecot --version 2.0.9 [root at

built-in; This doesn=92t matter because the moderator would have to use meetmeadmin or the confbridge equivalent to control the other functions. The moderator would either need two phones or a phone and a web = interface. Let=92s say Yves=92 =93special conference=94 is 5555. The moderator = would start using this command Exten =3D> s,1,meetme(5555) The participants would do Exten =3D>

Telco, location, ect?) At X times of day? =20 Ect, ect. =20 It sounds like bleed over, which can be causes by some many things the best place to start is to find a pattern if there is one. =20 James Shigley Monroe Telephone Answering Service =20 From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of David @ULC Sent: Tuesday, May

Class: default File: /var/lib/asterisk/moh//reno_project-system File: /var/lib/asterisk/moh//macroform-robot_dity File: /var/lib/asterisk/moh//manolo_camp-morning_coffee File: /var/lib/asterisk/moh//macroform-cold_day File: /var/lib/asterisk/moh//macroform-the_simplicity Class: none File: /var/lib/asterisk/moh/.nomusic_reserved/silence

"What is CentOS? CentOS is an Enterprise Linux distribution based on the freely available <ftp://ftp.redhat.com/pub/redhat/linux/enterprise/> sources from Red Hat Enterprise Linux. Each CentOS version is supported for 7 years (by means of security updates). A new CentOS version is released every 2 years and each CentOS version is regularly updated (every 6 months) to support newer

Connected to Asterisk 1.4.11 currently running on asterisk (pid =3D = 31999) -- Remote UNIX connection Verbosity is at least 8 -- Executing [00425298582 at numberplan-custom-1:1] Macro("SIP/8001-b7d0bb20", "trunkdial|SIP/trunk_3/0425298582") in new stack -- Executing [s at macro-trunkdial:1] Dial("SIP/8001-b7d0bb20", "SIP/trunk_3/0425298582")

entries) [2002/06/07 12:46:16, 0] smbd/service.c:(614) 1001627gefage (3.61.235.23) Can't change directory to /tmproot (Permission denied) [2002/06/07 12:46:17, 0] rpc_client/cli_netlogon.c:(157) cli_net_auth2: Error NT_STATUS_NO_TRUST_SAM_ACCOUNT [2002/06/07 12:46:17, 0] rpc_client/cli_login.c:(74) cli_nt_setup_creds: auth2 challenge failed [2002/06/07 12:46:17, 0]

asterisk -rx 'core show channels' | grep DAHDI | sort -n Channels with a value of 1-23 are on your primary DS1, channels with a value of 25-47 are on your second DS1. Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 ----- "Mike" <list at virtutel.ca> wrote: > > Hi, I have just recently been using DAHDI, and I wanted to know how to

Connected to Asterisk 1.4.11 currently running on asterisk (pid =3D = 31999) -- Remote UNIX connection Verbosity is at least 8 -- Executing [00425298582 at numberplan-custom-1:1] Macro("SIP/8001-b7d0bb20", "trunkdial|SIP/trunk_3/0425298582") in new stack -- Executing [s at macro-trunkdial:1] Dial("SIP/8001-b7d0bb20", "SIP/trunk_3/0425298582")

1. Does "Wrap-Up-Time" apply to all queue agents/extensions that just rang,= or only the one who actually answered the call (I assume the latter)? 2. Does the "Member Delay" delay the ringing of new calls to agents, or onl= y come into play AFTER the agent answers the ringing call? Any other suggestions for how I can resolve this issue? I am wondering whet= her "Agent

Connected to Asterisk 1.4.11 currently running on asterisk (pid =3D = 31999) -- Remote UNIX connection Verbosity is at least 8 -- Executing [00425298582 at numberplan-custom-1:1] Macro("SIP/8001-b7d0bb20", "trunkdial|SIP/trunk_3/0425298582") in new stack -- Executing [s at macro-trunkdial:1] Dial("SIP/8001-b7d0bb20", "SIP/trunk_3/0425298582")

using Asterisk. =20 Is this all you want Asterisk to do? (eg as an application service rather than provide telephony for the rest of the library as well), or are you looking to have it replace your existing telephony equipment? =20 As a suggestion if you google Trixbox and Nerd Vittles you will find a fairly detailed explanation of how to set your Trixbox server (a version of Asterisk) up to

"Why Siphon doesn't allow to receive a call when it doesn't run Apple doesn't accept (for the moment) an application runs in the background= . So, when Siphon doesn't run, the SIP server of your provider doesn't know= your iPhone." --_000_EC80F07C30CE3E46B2AD6B4407BE086F0C2AAD0248cworksmailcwo_ Content-Type: text/html; charset="us-ascii"

=20 arp | grep "192.168.0.1" =20 substituting the IP address of the SIP device. =20 From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of DHAVAL INDRODIYA Sent: Wednesday, 28 October 2009 4:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] SIP client MAC address. =20 hello, is

=20 Thanks!=20 =20 =20 Gustavo A. Gonz=E1lez Dto. de Infraestructura Despegar.com, Inc. ggonzalez at despegar.com=20 =20 ------=_NextPart_000_003E_01C8C00B.B3A8DA60 Content-Type: text/html; charset="iso-8859-1" Content-Transfer-Encoding: quoted-printable <html xmlns:v=3D"urn:schemas-microsoft-com:vml" = xmlns:o=3D"urn:schemas-microsoft-com:office:office" =