Displaying 20 results from an estimated 26 matches for "sierralta".
2003 Oct 08
4
asterisk & festival problem.
Hi,
I?m trying to get app_festival to work. I got the source from the
Debian woody package of festival-1.4.2 and applied the patch that came
with * sources it applied fine; then I made the debian package and
installed it.
I have this on extensions.conf:
exten => 6700,1,Festival(Hi there how are you doing ?)
When I dial 6700 I hear nothing and then * hangups:
-- Executing
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 Feb 06
8
snom soft phone
Some of you might already know that we are releasing a new phone, snom
360. To make the phone well-known and stable, we have made a soft phone
version out of it and offer it for trial or private use for free (for
more details, see the license conditions).
There are only few limitations to the phone. First of all, the audio
subsystem will work only work with an acceptable quality if you are
using
2003 Oct 14
2
VAD in Asterisk ?
Hi,
Is there is some form of VAD on * for SIP channels, cause I have a
problem with MOH. I made an extension which simply plays MOH, when I
dial that extension with my ATA188 MOH sounds choppy if I talk on the
phone the MOH keeps playing.
I saw the sip channel (show channel SIP/*) and I see no packets going
in/out when I talk then packets shows going in/out.
I don?t have this kind of problem
2006 Jan 18
2
SipAddHeader bug?
Hi,
I'm using the new SipAddHeader application on Asterisk 1.2.1,
here's a snip of my extensions:
exten => _9XXXXXXX,1,SipAddHeader(P-Asserted-Identity: <sip:${CALLERIDNUM}
exten => _9XXXXXXX,2,SipAddHeader(P-Asserted-Identity: tel:${CALLERIDNUM})
exten => _9XXXXXXX,3,Dial(SIP/${EXTEN}@${SIPTRUNK},,tT)
exten => _9XXXXXXX,4,Congestion
The problems is that Asterisk
2006 Jan 28
1
double ringing tone on asterisk 1.2 (workaround)
After reading a description of apparently the same problem by Juan J.
Sierralta more detailed than mine
"tuuu tuuu instead of tuuu" we've solved the problem changing the call
progress tone of sip phones to something not udible.
2003 Dec 05
3
MGCP IADs
Hi,
For MGCP users. Is there any success stories with any MGCP IAD vendor.
I?m trying to find an IAD which works with Asterisk. I?ve tried the
Cisco IAD 2430 without success; but SIP on this IAD works but it?s
limited (no authentication, no notify messages, etc) and with higher
density IAD (16 or more ports) it?s nice to control using MGCP.
Any information will be apreciated !
Thanks.
--
2003 Oct 03
3
Message Waiting on Cisco 7940 does not work
I have a cisco 7940 with the following sip.conf config:
[Desk1.1]
type=friend
secret=******
defaultip=192.168.1.14
insecure=no
mailbox=102
callerid="Desk1.1"
qualify=500
canreinvite=no
context=extensions
host=dynamic
group=2
I do not get message waiting indicator (mwi) on this phone. Is the
another .conf file invilved in configuring this function other than the
mailbox=xxx in the
2003 Oct 18
6
Outgoing call to IVR not being "answered"
I don't know if this is a problem with my cisco sip IP Phones or
asterisk but I thought I would post here in case someone else has
experienced this issue.
When I make a call from my SIP cisco IP Phone to some remote IVRs I
never get the rest of my soft keys, only the "End Call" soft key, and
also DTMF doesn't work... its like the phone is acting like the remote
end hasn't
2004 May 16
6
X100P problem with PSTN from BOLIVIA
Hi
Please help!
I have one X101P and TDM400P in my asterisk Box
When i make a call from * to PSTN, everything goes Ok,
When the PSTN hangups or * hangups, the busy tone is detected and *
disconnects the channel without problems.
The problem occurs when the call comes from PSTN. When * hangups, the
other end (at pstn) does not hangup, it only presents silence.
Please tell me how to solve this
2004 Jun 08
8
New version of DIAX (0.9.8a) available now for free download
Hi all,
A new version of DIAX (0.9.8a) is ready to be downloaded from the following
locations:
http://www.laser.com/dante
or
http://www.geocities.com/tdanro
What's new in 0.9.8a:
- unconditional autoanswer or based on CallerID (user configurable);
- use any Ericsson/SonyEricsson GSM/PCS to control DIAX (feedback on the
phone display) through Bluetooth (or serial cable). You do not even
2003 Oct 15
2
skinny problem
has anyone seen this?
-- Starting Skinny session from 192.168.13.102
-- Starting Skinny session from 192.168.13.102
triton*CLI> Oct 15 13:44:05 WARNING[213019]: File chan_skinny.c, Line 2243 (get_input): Skinny Client sent less data than expected.
Oct 15 13:44:05 NOTICE[213019]: File chan_skinny.c, Line 2301 (skinny_session): Skinny Session returned: Success
Oct 15 13:44:05
2003 Nov 19
1
2 TE410P
Hi,
Is there anybody in this list who had experience with two TE410 cards
on a server ?
I know that the cards can?t share IRQs and I?m seeing to have two cards
on a x335 IBM Xeon server.
TIA
--
Juanjo sin .sig
2003 Dec 18
2
Polycom phones update
Hello,
We have updated the Wiki page for Polycom phones:
http://www.voip-info.org/tiki-index.php?page=Polycom+Phones
We posted several configuration specs as well as a link to an admin guide
for the phone.
We also posted a link on there to two firmware versions for download.
The official Asterisk-Polycom support website should be up and live sometime
in January.
If anyone has anything to add
2003 Oct 17
2
Polycom IP 600 phone
Hello,
I have finally received the details from Polycom to get into the backend
configuration of their SoundPoint IP 600 SIP VOIP phone. The phone is quite
nice looking but the configs are very sparse, not even a place for a
secret(password) field in their SIP registration section.
If anyone else has one of these and needs the passwords to get into the back
end configurations, just send me an
2003 Dec 17
2
Troubles with voicemail and cisco 7905 SIP
I'm deploying a fairly large number of Cisco 7905 with SIP connected to an
asterisk PBX.
The problem:
the 7905 has this nice "feature": You can set a "voice mail number" in
configuration, so that you can listen your voice mail just by hitting the
"messages" key on the phone. It just autodials "8500" or whatever else.
The customer wants this, so
2003 Sep 15
4
Talking to other SIP hosts, wrong IP
As per my problem yesterday with the Cisco 7960 and getting it talking to
Asterisk on a different subnet, I gave up trying and just put the Asterisk
box back on the internal subnet.
However, I made two changes:
- the external IP address is set on an ethernet alias eth0:0
- the main Linux router will change outgoing requests from 10.1.1.2 to the
external IP (rather than the default behaviour of
2003 Dec 03
2
Cisco IAD with MGCP
I repost a message I put a week ago:
I have a Cisco IAD 2431 which has MGCP protocol; I cannot make
it to work againts Asterisk; at least there is some MGCP conversation
between them but when I offhook a phone attached to IAD I get no tone at
all.
As anybody managed to get working Asterisk against an MGCP Cisco
gateway ?
Which MGCP version should I use ?
Also I recently
2003 Nov 19
3
RTP timing in a SIP only world (choppy MOH)
I have an * setup with sip phones and sip fxo gateway.
When a sip phone places a sip/fxo call on hold, MOH is very choppy.
It looks like RTP has a real problem with timing if it is not receiving
RTP packets. If the outside call that is placed on hold is not generating
any audio, the sip/fxo gateway does not send * RTP packets.
Is this valid?
Is this a problem with the sip/fxo gateway or a problem
2004 Dec 01
8
Interrupt latency problems
I'm debugging a TxFax problem whereby the fax transmission fails. I
suspect interrupt latency--some interrupt routine is holding its
interrupt too long. I have all unnecessary services switched off and X
is not running when I perform these tests. Some transmission are
successful while others fail at random points.
I've noticed that after I boot Linux, load zaptel, wcfxo, and wcfxs,