Displaying 20 results from an estimated 20000 matches similar to: "Difference in show channels"
2007 Sep 09
1
Maximum retries exceeded on transmission
I have searched this list and others, and see other pepole having this
issue. However, I have not seen how to fix it.
Sep 6 18:52:36 *WARNING*[4620]: *chan_sip.c*:*1835 retrans_pkt*: Maximum
retries exceeded on transmission
778f89593967725f0abe40eb1752504c for seqno 1620 (Critical
Response)
Sep 6 18:52:36 *WARNING*[4620]: *chan_sip.c*:*1835 retrans_pkt*: Hanging up
call
2005 Jan 09
2
TE110P error
Good day all
We got a Wildcard TE110P
I installed linux,zaptel,libpti and asterisk
I coped over my zaptel.conf and zapata.conf from a previous E100P config
But when I try to start asterisk it gives error not bying able to load
zap channles:
== Parsing '/etc/asterisk/zapata.conf': Found
Jan 10 08:17:18 WARNING[-1084595552]: chan_zap.c:9308 setup_zap:
Ignoring switchtype
Jan 10 08:17:18
2009 Sep 27
1
Peers Listed in "sip show channels"
Hi,
I am using Trxibox 2.6 latest ISO install.
Following is the output of : "sip show channels"
[trixbox ~]# /usr/sbin/asterisk -rx "sip show channels"
Peer User/ANR Call ID Seq (Tx/Rx) Format
Hold Last Message
212.53.40.40 0218245 6cfb845d050 09011/00000 0x0 (nothing) No
192.168.1.116 (None) YTc4ZmM3NjV 00101/00006 0x0
2007 Sep 10
5
online active call watching
Dear all
I have asterisk 1.4.11 i am new in asterisk i want to see online call list how it is possible to see how man call currently active is there any command or tool to see online call ?? from --- to
Regards
---------------------------------
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2007 Sep 06
1
Dead SIP channels
I am using a2billing as calling card platform with asterisk 1.2.17.
After running for several days, if I issue 'sip show channels' command, I got a lot of dead sip channels although 'show channels' command only show 5 channels. What cause these dead channels? How can I clean out these dead channels? Will they pose any problem to my * server if left alone? What does this (d) mean?
2007 Jan 22
1
OT: Optimum voice problems.
I'm trying to figure out if I'm the only one with these problems with them.
I recently had a few customers that switched to them because of the
price, of course that means that they have to use FXO ports, but it is
realy cheaper, so customers don't really care.
In any case, there are 2 issues that I can't get solved, and they are
not interested in helping.
1. When they tell you
2008 Jul 22
2
3-way calling for IAX channels
How can I made a 3-way conference betwwen IAX channels?
My current version is: 1.4.21.1
Thanx,
Daniel Arohuanca Lagos
+51 1 3594122
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2009 Nov 07
1
Difference between 'core show channels' and 'sip show channels' ??
vps*CLI> iax2 show channels
Channel Peer Username ID (Lo/Rem) Seq
(Tx/Rx) Lag Jitter JitBuf Format
0 active IAX channels
vps*CLI> core show channels
Channel Location State
Application(Data)
0 active channels
0 active calls
vps*CLI> sip show channels
Peer User/ANR Call ID Seq (Tx/Rx) Format
2007 Sep 07
3
Show Callee name on Display
We have users with Cisco 7900 phones running sip. When user A calls
user B, we want user B's name to appear on user A's phone. It shows the
extension they call, but not the internal name of the called user. Is
this possible? We have some people that used to be on an MGCP based
system and they would get the callee's name popup on their phone when
they called someone. I
2007 Sep 05
7
Can asterisk give half-ring periodically for MWI?
Hi all,
Configuration: Analog phone connected to TDM400p.
I'd like the phone to give a half-ring (chirp) periodically when there
is a message waiting. Can this be done? How is it configured?
The visible "Message waiting" indicator and the stutter dial tone are
working fine, but are not sufficient for me.
Thanks!
2007 Sep 18
6
Limiting Simultaneous calls
Is there a way to limit simultaneous calls. I like to limit
simultaneous outgoing calls as more than few simulataneous calls are
charged by my voip providers. However, I do not want to have any such
restriction for internal calls.
Thanks
Jim
2009 Aug 18
2
Channels don't go away with soft hangup
Hello List,
our setup:
Callcenter
IBM Hardware, 1x TE420, 1x xircom analog switch, 4x different cellular
providers on the xircom analog port, ~60 agents
Debian 5.0.1 (Lenny)
Asterisk 1.4.21.2 Debian Package recompiled with additional app_queue
segfault fix
Zaptel 1.4.11 Debian Package
My Problem is I have two channels (Zap/9-1 and Zap/6-1) which have a
duration of over 4 hours.
I am
2007 Sep 11
4
Installing Asterisk on to CentOS 4
Hi expets,
I have installed Asterisk 1.4.11 on CentOS4 successfully without any error.
But when i am trying to start asterisk with following cmd i am getting unknown command.
[cybercall at ip-208-109-177-212 ~]$ asterisk -vvvvvvc
-bash: asterisk: command not found
[cybercall at ip-208-109-177-212 ~]$
I checked modules and other configuration files which are installed correctly.
Please help me
2007 Sep 26
2
ChanSpy issue
Hello list
I am having an issue with Chanspy/SIP that I?m hoping someone has come
across and resolved in the past.
I am sending calls that come in TDM through T1 ZAP channels and go out to a
SIP trunk.
If I spy on the SIP channel, I can hear the person on the SIP side of the
call just fine, but the person on the ZAP channel fades in and out.
If I spy on the ZAP channel, and can hear
2007 Sep 27
15
Cisco 7940G licensing with asterisk
Hi there,
In Cisco web site
http://www.cisco.com/en/US/products/hw/phones/ps379/products_data_sheet09186a008008884a.html
It says that regardless of the technology used you have to buy a licencse.
Does the license apply to use the phone with asterisk, or, can i just
buy the phone?
Also, the phone does not requiere to use an AC adapter if used with
PoE injectors/switches.
Can non-Cisco PoE
2009 Sep 10
4
Looking for a way to show caller id information on the desktop
Hi there.
My problem, I can't figure out how to ask this question. So,
hopefully someone out here can point me to the FM on this.
I would like to have either a web page or an application that I can
view that whenever a call arrives on the Asterisk server
the application will display the callerid information. I've found
quite a few examples of the reverse of this. To where a
script is
2007 Sep 18
4
Linux limits
Hi all,
Any one know how to increase the Linux limit? I am hiting a wall on 200
calls playing files at the same time. From Asterisk console, I am
getting messages like
Sip_request_call: Unable to build sip pvt data for "asterisk1/700"
Too many open files
Is this a limit of my Linux box? I only have 512MB of ram. Will increase
it to 2G help or I have to change some configuration in
2007 Sep 28
4
. (period): Wildcard match; matches one or more characters
Hi List;
In the outbound, I read in the documents the Wildcard
match "by using the . (period)", but I did not
understand how Wildcard will work (like what)? As I
know that Wildcard is a term used with the Diguim TDM
card (FXO and FXS), so what is the relation between
such cards and the matching in the dial plan?
Any help?
Regards
Bilal
2009 Aug 24
1
disconnection silent channels
Dear,is any way to find silent channels , and disconnect them after 30 secs?
best
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2007 Sep 26
4
Asterisk realtime error
Hi! I am proving Asterisk 1.2.24 in realtime with MySQL 5.0.27 using Idefisk
softphones. I followed the steps of "how to" of voip-org but always have
this error:
Sep 25 20:29:07 WARNING[12000]: res_config_mysql.c:360 update_mysql: MySQL
RealTime: Failed to query database. Check debug for more info.
Sep 25 20:29:07 WARNING[12000]: res_config_mysql.c:360 update_mysql: MySQL
RealTime: