Displaying 20 results from an estimated 7000 matches similar to: "5-7 second connection delay in outgoing FXO calls"
2011 Jan 14
1
5-7 second delay in connecting outgoing FXO calls
I'm running AsteriskNow 1.7.1 with a OpenVox 2FXO/2FXS card,
SPA942 SIP phone and outgoing SIP and IAX routes.
When I dial local PSTN numbers from the SPA942 using the FXO channels I
observe a 5-7 second delay between when the PSTN number answers the call
and when Asterisk connects the call at my end. There's enough delay
time that I hear an additional ring after the PSTN number has
2010 Sep 17
0
5-7 second delay in connecting outgoing FXO calls
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2011 Jan 24
2
Outgoing FXO calls have no audio with callprogress=no
My outgoing FXO calls are answered but have no audio in either direction
if I have callprogress=no in chan_dahdi.conf. If I change to
callprogress=yes then the audio returns. My chan_dahdi.conf file is
listed below. Can anyone point-out why callprogress=no isn't working?
#cat /tmp/a
[trunkgroups]
[channels]
language=en
context=incoming
toneduration=40
;usedistinctiveringdetection=yes
2011 Mar 18
2
Problem routing call to fax machine on DAHDI FXS port
I am running Asterisk 1.6.2.17.2 with a Openvox A400 card with 2FXO/2FXS
modules. I'm trying to set-up things to route analog fax calls from a
FXO port to an analog fax machine on a FXS port on the same card.
Outgoing faxes work just fine. But incoming faces are routed to the
right DAHDI extension, but the call dropped right as the fax machine
rings for the first time. The fax machine
2010 Nov 24
2
SPA942 on speaker phone does not hang up?
Hello all,
I am using Linksys SPA942 in my current installation activity. I see a
peculiar behavior: A call is made and the SPA942 uses its speaker. When the
far end of a call hangs up , the SPA942 stays off hook, and after a time
plays a fast busy. The user then has to press the line presence button to
hang up the phone.
I think I must be missing some sip.conf parameter. My sip.conf is pretty
2007 Jan 11
2
calls to SPA942 disconnect after 15 seconds (chan_sip.c set_destination: can't find address)
Am having a unique problem, calls received on my SPA942 seem to end after 15 seconds, but calls made from this device do not have this problem.
For this device (when receiving calls) I get periodic "chan_sip.c set_destination: can't find address for host"
I have set the "canreinvite=no" in the sip.conf. Does anyone have a sample entry from sip.conf for the Lynksys SPA 942
2006 Oct 15
2
SPA942 quality for a Bank
Before committing to about 50 of the spa942's, I like to take a last
poll from those on the list to identify any negative issues that might
be associated with the audio, functionality, early failures, etc, on the
spa942.
Expecting to deploy these using existing cat5 cabling and both rj45
jacks. Been using three of theme in a short term demo with the customer,
but the demo systems has
2009 Oct 19
3
delay in processing dtmf
Hi,
I'm new to this list
I'm developing asterisk application where users can call and control volume
up and down in music player.
Problem I'm getting is if users press 222228 in fast speed, system will
process all those 2s and then process 8, so there is few seconds ( around
4-5) processing key press 8 , therefore users will feel unresponsiveness in
system.(in other words users will
2009 May 22
3
No response to our critical packet problem
Hi,
I have a strange problem. At a site where there are 20+ phones, there
is one phone that cannot make outbound (to PSTN) calls.
Each call is dropped after 20s with "no response to our critical packet".
Calls to voicemail and internal extensions work fine.
I understand that everything points to a NAT problem, but I don't
understand how it could be because:
1) It does not affect
2009 Nov 11
1
How to control DTMF tone duration on Zap channels?
Hi,
I am using zap channels, and by using sendDTFM application, I can control
the duration between two DTMF digits, but I can't find a way to control the
duration of the digits themself. Did search on the Internet and found out
that I can change it in the asterisk source files and recompile asterisk.
Wiki also says that it can be controlled using toneduration option in
zapata.conf, but it
2008 Aug 11
1
Asterisk Realtime Unregister
Hi,
I'm running asterisk realtime, i had prob when a user does not
unregister properly.
I tested with SPA942 and a PAP2, when phone is registered, i call using
the SPA using x-lite no problem, but when i unplugged the power, it does
not unregister properly, so asterisk think SPA942 is still registered,
when i call using x-lite, asterisk tries to call it.so it gets stuck at
[Aug 11
2007 Jan 08
2
OT:spa942 provisioning
Hello!
Sorry for the OT-thread, but i don't know where else too ask...
Has anyone done http-provisioning of a Linksys SPA942 with client side
ssl-authentication? Where do i get the CA from?
I'm aware of the Sipura mass deployment howto on voip-info.org, but it
doesn't cover the authentification part.
Thanks
Christian
2007 Nov 21
3
Aastra 480i CT - No Incoming Calls
I just bought an Aastra 480i CT for a client who needed cordless
capabilities in their office. I'm trying to set up the base station and
cordless handset in my office first. I'm able to connect the phone to
my Asterisk box and make outgoing calls from either the base station or
the handset - to extensions within my office as well as numbers outside
the network. But I can't
2008 Mar 26
1
Got SIP response 406 "Not Acceptable"
I'm getting "Got SIP response 406 "Not Acceptable" back from 10.0.1.2"
occasionally when try to dial to SPA942 ,
anyone has any idea on this before i consider Firmware upgrade?
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2009 Jun 06
2
Sangoma A200 wt HW echo on 64bit Centos , 4 fxo and crosstalk!
Hi,
Has anyone ever experienced crosstalk with sangoma analogue cards on 64bit?
We have exhausted every test to try and replicate this and find a
solution with Sangoma tech support, but we can not fix it.
We are about to try the card and four *seperate* UK BT lines in a 32bit system.
The current system is a 4gb, dual core cpu with pbx in a flash 1.4,
Zaptel and Asterisk 1.4.21-2
Currently we
2011 Jul 10
2
Thomson ST022 - External Call problems
Hy all of you,
I've successfully installed a freepbx solution with 10 extensions :
- 5 on Linksys SPA922
- 1 on Linksys SPA942
- 1 on Thomson ST022
Everything seems to work fine with all the hardphones excepts last week.
The thomson has a strange behaviour. It can reach french mobile cell
phones but when it reaches "fix" phones, the correspondant can't hear
the caller.
What
2007 Feb 14
1
[Fxo] Digium TDM01B vs. OpenVox A400P01?
Hello
If someone had the opportunity of trying those two analog cards, how do
they compare? Digium's sells for $150 while OpenVox's sells for $95.
Thanks.
2007 Nov 19
4
Help: How to configure SIP domain on SPA942
I'm using a bunch of SPA942's, and I'm trying to provision them mostly
by DHCP (and what I can't set that way, I try to provision via HTTP
interface into the phone).
I changed the domain in my AstLinux config from "astlinux" to redfish-solutions.com, and set
that in my sip.conf file as well:
context=incoming
2008 Oct 25
9
Cheapest 4 port FXO
I need to increase reliability at an office as SIP/Internet provider outages are causing some issues.
What would be the least expensive analogue card that people are using reliably?
Thanks!
jlc
2010 Oct 16
3
Detect incoming fax on PSTN and route to fax machine on DADHI extension?
I'm running an AsteriskNow V1.7.1 with both a PSTN connection and fax
machine. Both are connected to a DAHDI board. I'd like to route
incoming PSTN fax calls to the extension of the fax machine and process
non-fax calls through different dialplan.logic.
What's the best way to go about doing this? I've looked into Fax for
Asterisk, bit I'm not sure that I want it or NVFax