Displaying 20 results from an estimated 5000 matches similar to: "slight echo via sip provider"
2006 Apr 05
2
SIP Asterisk Polycom Reinvite
Wondering if anyone has experienced an intermittent one way audio
(called party can not hear) problem in these conditions;
Several IP501 phones local, same subnet.
Remote asterisk
No NAT anywhere
Polycom IP501 ulaw only, canreinvite=yes
Asterisk
Call termination path is to a sonus GSX operated by the upstream
carrier, ulaw only, canreinvite=no
The idea is that if the Polycoms are
2007 Jul 24
2
SIP IP Trunk, between Asterisk and Softswitch
Dear List;
I am trying to create a link between Asterisk and My
softswitch, the link to be SIP Trunk.
I did the below configuration and I do not know if any
one can help me and advise me to have better
configuration to be sure that link is fine. But I do
not know how to determine the SIP username to be sent
for my softswitch as sometimes the softswitch need to
check it.
Also, does asterisk
2015 May 21
2
libvirt and VMWare Workstation Shared Server mode (of GSX history)
Hi everyone,
I searched previous postings and I couldn't find a definitive answer on
this..
I run a small lab of RHEL/Centos Based servers on which there's VMWare
Workstation running on a non-standard port but still manageable by tools
like VMrun (and the Fusion of Workstation GUI, of course).
I'm trying to use virsh with this setup and getting the following error
from both
2004 Dec 24
3
Preventing Asterisk from sending 'h' across to SIP Provider
Hi,
I want to prevent Asterisk from sending the h extension across to the SIP
provider or to prevent it from hitting the script at all. The SIP Provider
does not know what to do with the h extensions once it receives it. My SIP
Provider takes all digits and forwards them off to a softswitch for
processing. Everytime a call hangs up, it complains about running AGI scripts
on hungup
2006 Mar 15
1
dropping voice frame ulaw - slin?
Mar 15 12:54:01 NOTICE[24269] channel.c: Dropping incompatible voice
frame on Local/[removed number]@context-5c3e,2 of format ulaw since our
native format has changed to slin
Can anyone provide an English translation of what this means?
The extension is a Polycom IP 501
The only allowed formats are g.711u
MOH is MP3 files (obvious)
All prompts have been re-recorded in .ul uLaw
2007 Oct 23
2
register => to let Asterisk register to another softswitch via SIP
Dear Alex;
Thanks alot for your nice help.
This is if I need to let Asterisk register with
another softswitch (so I used register =>), what if I
need asterisk to send call for the softswitch without
register to it (directly)? If I removed the register
=> then how it will distiguish the IP address in the
"host" at the [sip_trunk] is the IP address of the
softswitch that need to
2007 Sep 19
2
what is softswitch
Dear all
what is softswitch what is difference between asterisk and softswitch ??
regards
satish patel
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2005 Mar 03
3
Asterisk not relaying back the SIP response messages
HI all,
I have the following setup running:
EP<--->Calling Asterisk<--->Relaying Asterisk<--->Softswitch<---> PSTN
The Endpoint EP is registered with the Calling Asterisk. Calls are
forwarded from this machine to
Relaying Asterisk which in turn forwards it to the Softswitch. In
addition, this machine also
relays back responses from the Softswitch to the Calling
2003 Nov 07
2
Softswitch
Pardon my ignorance, but I was hoping someone could clear up something for
me.
- For a few POTS lines, digium has a single port card for that, or a T1 card
to a channel bank.
- For 10 or more lines, digium has a T1 or E1 card for that too based on PRI
channels
- For 100's to 1000's of lines, I suspect a soft-switch is in order???
A traditional phone company will sell:
- POTS lines for
2007 Dec 02
2
Softswitch digim
Hello averybody,
I'm looking the softswitch in digium website, anyone test the softswitch?
Best Regards
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2007 Jun 28
2
Linking Asterisk with another SIP PBX (or SIP Softswitch)
Hi List;
If I need to do a trunk between Asterisk and another
SIP softswitch (so Asterisk will send a SIP calls to
that softswitch), then I have to configure this on the
sip.conf file or where exactly? And is it the same
when I configure iax trunk?
Should I determine the context in this case for this
SIP trunk?
Regards
Bilal
2007 Jan 30
1
Strange problem
Hi guys.
I'm working on a VOIP service provider.
We have two customers running asterisk. Customer A and B.
When A call to B everything is ok.
When B call to A the call ring but sip messages didn't arrive on
asterisk A. In my softswitch i see the invite sip message sended to A.
When every other numbers(TDM and SIP) call do A everything is ok.
Have any issue in asterisk that can resolve
2003 Aug 06
2
Semi-newbie question "Softswitch" and Asterisk - Is there a difference?
I've been working in the VoIP industry for just a bit over a year now...
Mostly taking care of the underlying systems. I've now reached the
point where I'm being drawn more and more into the call processing side
of things. My background is in computer and "classic" telephony systems
(DMS250/MTX, DSC 400, T1, channel banks. telabs analog echo supressor
modules and
2010 May 26
4
Help with IP Routing
Hello,
?
I'm in a bit of a fix. We have a particular Windows based softswitch which is has its SIP and H323 ports hardcoded to listen on a particular IP address. The problem is that the ISP is having major issues and we can no longer depend on them for service. The softswitch will not listen on any other IP address and this can not be fixed. I was thinking of creating a NAT network wherein we
2009 Jan 15
2
How to transfer a call from one Asterisk Server to another
Can anyone tell me how I can completely move an established call off of one
Asterisk server to another?
In our case we have a server with our IVR. Depending upon digits entered,
the call can be transferred to any of our other servers depending where the
extension or queue reside.
We would like to completely move the call off of the first box so we don't
tie up resources on it.
In our lab
2003 Oct 24
8
SS7 signaling/Softswitch
I'm confused a bit about the following and was hoping to get some answers on
this group - What is exactly implied when we say asterisk can connect to a PSTN.
Does it mean connecting to the PSTN via PRI/T1/E1? If yes, then I assume
asterisk does not need to do any SS7 signaling and all it does (playing the role
of a PBX) is to connect to a Class 5 Switch at the CO. Is this a correct
statement?
2005 May 27
2
Grandstream GSX-2000 - dead :-(
I have a Grandstream GSX-2000 with ..
Software Version: Program-- 1.0.0.3 Bootloader-- 1.0.0.3
I tried to do an HTTP update from the Grand Stream web site...
After half an hour, I recycled power and now its dead... LED's come on
and stay on, screen and buttons are dead. Connectivity to
Grandstream.com was always good - whenever I checked (I downloaded the
"User Manual" in a
2005 Aug 10
4
GrandStream GSX-2000 strangeness
I have a really baffling problem.
A couple of months ago I purchased a pair of GrandStream GSX-2000 phones for
use with Asterisk.
At first all was well. But recently I've noticed terrible sound quality
problems. Basically the sound will "glitch" or stutter randomly from time to
time.
Now, what is interesting is that this happens even with the phone totally
disconnected from any
2015 Aug 14
2
chan_sip.c: Retransmission timeout reached on transmission
Hello friends:
I am facing cutoffs randomly when negotiating calls.
The PBX dials the destination, the provider (softswitch) receives the
request *[1]* and sudenly the PBX hangs up the call* [2]* while the
provider is still dialing it, as a consequence the remote peer receives a
ghost call. Along the atempt I could see six times a messages regarding NAT
isuues *[3]*
I hope anyone can give me an
2005 Sep 02
0
Semi-OT: An idea for New Orleans temporary communications infrastructure
The national guard and/or army routinely implements VoIP over wireless
in situations where comm is lost, I did see an news release that the
Guard started this project in the south the day after the disaster hit.
The key is not the VoIP infrastructure, that is the easy part (one ss7
Sonus softswitch and a DS3!), the key is distributing IP over a wide
area, which is best done on the quick with WiFI