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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

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>

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.

"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

"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

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

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

=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

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

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")

=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

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")

eeds to be added to the database or coding of the program. Is this right? I= sthere something else I need to do?? Ray Garza System Analyst II Materials Management (956) 381-2163 --_000_60EAA3B47B099544A16557729563A1D608920F4FCCINFDBMBX4dsut_ Content-Type: text/html; charset="us-ascii" Content-Transfer-Encoding: quoted-printable <html

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")

to use the table asterisk.cdr but I can't find it anywhere. ------_=_NextPart_001_01C92A7E.C6B88024 Content-Type: text/html; charset="us-ascii" Content-Transfer-Encoding: quoted-printable <html xmlns:v=3D"urn:schemas-microsoft-com:vml" = xmlns:o=3D"urn:schemas-microsoft-com:office:office" = xmlns:w=3D"urn:schemas-microsoft-com:office:word" =

=========== process 5496: arguments to dbus_connection_ref() were incorrect, assertion "connection != NULL" failed in file dbus-connection.c line 2497. This is normally a bug in some application using the D-Bus library. org.freedesktop.DBus.Error.Spawn.ExecFailed: dbus-launch failed to autolaunch D-Bus session: PuTTY X11 proxy: wrong authentication protocol attemptedAutolaunch error: