Displaying 20 results from an estimated 1000 matches similar to: "SMS Gateway"
2006 Jan 28
2
RoadRunner
I use SIP over VPN with RR from TWC no problem, connect via WiFi.
According to http://www.speakeasy.net/speedtest/ I am getting 3.5Mbps
down and 353Kbps up at this time (6:15pm Saturday). My laptop currently
has an X-Lite (free version) softphone with GN Netcom USB professional
contact center headsets (GN8110 USB XP adapter). We have found that the
headset makes a major difference in the quality
2005 Aug 10
2
Help with TNT and Asterisk
Im having some problems with connecting a TNT to asterisk. The problem
is
when the call is sent to asterisk and signaling is done the RTP syncs
however
no audio is produced. Can someone give me some idea of how to
accomplish this?
I am using the standard configs and g711 and 729 do the same. No audio.
Public IPs on both ends. No nat. Any ideas would be appreciated.
2005 Jun 30
5
Logrotate
I created some scripts to logrotate. I am having a problem. After I do
it, I am sending kill -HUP to the process
its not using the newly created messages file again. Could someone help
me out with how I can rotate asterisk's
log's without killing the process?
..o-------------------------------------------------------o.
Brian Fertig
NOC/Network Engineer
Planet Telecom, Inc.
Tampa, FL
2005 Feb 25
1
SIP Errors
Can someone explain what this error is?
-- Got SIP response 500 "Server Internal Error - Invalid CSEQ number"
back from 209.xxx.xxx.xxx
How do I fix this?
.o-------------------------------------------------------o.
Brian Fertig
NOC/Network Engineer
Planet Telecom, Inc.
Tampa, FL Office
2005 Jun 30
1
Cisco Voip Question
Does anyone in here know how to setup auto negotiation between g729 and
g711ulaw on
a cisco 5400? I would imagine it would be the same on a 3660.
The problem I am having is natively the call is setup for g729 however
when the call is transferred
to voicemail it uses ULAW so when the cisco tries to connect to the
voice mail I get a SIP error
that the codec couldn't be negotiated. I need
2005 Sep 30
1
No ringback tone generated by Asterisk with OH323connections
are you giving answer()?
..o-------------------------------------------------------o..
Brian Fertig
Network/Systems Engineer
IT Administrator
Planet Telecom, Inc.
Tampa,FL Office
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Juan Jose
Comellas
Sent: Friday, September 30, 2005 10:32 AM
To: Asterisk Users
2005 Jul 01
1
Got SIP response 481 "Invalid CSeq Number" backfrom X.X.X.X
I had the same problem and I believe it was the payload size of the
codec. What code are you using?
..o-------------------------------------------------------o.
Brian Fertig
NOC/Network Engineer
Planet Telecom, Inc.
Tampa, FL Office
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Federico
Alves
Sent: Friday,
2004 Sep 28
1
Looking for whoever wrote cdr_mysql
I don't completely understand this.. Lemme try it out..
[default]
exten => 1112223333,1,Macro(happy-did)
[macro-happy-did]
exten => s,1,Goto(${MACRO_EXTEN},1)
exten => _XXXXXXXXXX,1,NoOp(Normal "s" exten stuff here)
So when this is ran it will cut the cdr and the s will show the actual
DID not the s correct? But then the NoOp would be something like:
....
2005 Aug 16
1
DTMF, Asterisk, External PSTN gateway, and PAP2 (was: RE: Issue with DTMF Tones - CodecIssues)
I run a bunch of the Linksys ATA's.. I always use rfc2833 for DTMF.
works very well and have never had a problem with it.
..o-------------------------------------------------------o.
Brian Fertig
NOC/Network Engineer
Planet Telecom, Inc.
Tampa, FL Office
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of
2005 Jan 14
2
Passing PIN Numbers
To All
If anyone can shed any light on this it would be greatly appreciated.
My phones are unable to enter pins numbers correctly when required by the party they are calling.
For example I was given an outside number to attend conference bridge. After the call was connected it required me to enter a 4 digit PIN. Now here is the problem whenever I enter a pin it is received twice. For example if
2005 Jan 28
2
Fwd and Tollfree
Hallo all
do any of you know if the toll free access to the Netherlands is still working via FWD or Iaxtel?
thanks
liaan
---------------------------------
Do you Yahoo!?
Yahoo! Search presents - Jib Jab's 'Second Term'
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2005 Jul 10
2
SMS Handler in Asterisk
Hello all,
Recently I migrated all telephony in my house to asterisk thanks to the
Asterisk, QuadBRI which works wonderfully well. Some small tweaks to
make but that's on the long list.
On the short list is the ability to reliable send and receive SMS.
For SMS I already built a script email2sms, but sometimes the SMS
doesn't get send from some reason, the sms log then reports something
2004 Oct 04
2
300 extensions on Asterisk?
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hello
I am running an * box with just 8 extensions connected to our old Alcatel BCN
5200 PABX.
The requirement is that we now scale it up to handle about 300 lines and get
rid of our old PABX. Is there a way of hooking up 300 phones to asterisk
without going via the PABX. I am more of a network person than a telecomms
one so i may not fully
2004 Dec 07
1
Monitoring a call in an Call Center Environment
How can I monitor calls in a call center environment real time? Is this
possible? If so could someone show
and example of how this is accomplished?
.o-------------------------------------------------------o.
Brian Fertig
Network Engineer
Planet Telecom, Inc.
Tampa, FL Office
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2004 Oct 05
1
OT: Can I use a SIPURA with Packet8?
I have packet8 and I have spent many hours on the phone with them.
If someone has found away around there DTA configuration I would like to
know so I can bring it in house to my * box. But as far as your
question is concerned. No. Not that I know of. They wouldn't give me
any information about the configs.
.o-------------------------------------------------------o.
Brian Fertig
Network
2005 Sep 19
2
kill a .call file
Any means of killing a .call file that is in progress?
James Taylor
MetroTel
3505 Summerhill Road
Suite 11
Texarkana, Tx 75503
903-793-1956
2006 May 17
2
Diverse servers
I currently have a single server with a few SIP and IAX upstreams for origination and termination with IAX clients. I am adding a second server that will have a much higher capacity and will be handling a larger call volume. However, this second server is not going to be geographically near the first. It will largely share the same upstreams. I would like for this to be an integrated system
2005 Jan 28
2
Nortel --> Asterisk-------->Asterisk
I am looking at setting up the following configuration and any
help/input/comments before signing the PRI contracts will be greatly
appreciated.
PRI Tampa PRI
Sarasota PRI
<--> Nortel BCM-->Asterisk<------------------>Asterisk<--->
I would like to link the Nortel BCM to * using the a digital trunk card.
The BCM will
2004 Nov 24
2
call forwarding to gsm phones
Hii,
I want to forward calls from an asterisk server to a local gsm network.
I have read the wiki pages on various forums.
But the thing i want is to receive the call(Voip) from an asterisk
server then it should be forwarded to a gsm network & again to either
a gsm/ PSTN from the gsm network itself.
Please post a help.
Thanx in advance.
--
Day by Day in Every Way I'm Getting Better
2005 Aug 23
2
YAACID isn't working
Hello, I'm trying YAACID ( http://www.shatterit.com/opensource/yaacid/ )
for incomming call notification on PC (and open url with callerid), but
it does not display/pop anything :-(
my config is very simple...
(yaacid is successfully registered as manager in asterisk)
thanks
PJ
* dialplan:
'953' => 1. NoOp(${CALLERID})
[pbx_config]