Displaying 20 results from an estimated 1000 matches similar to: "No subject"
2007 Jul 12
0
No subject
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Thanks!=20
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Gustavo A. Gonz=E1lez
Dto. de Infraestructura
Despegar.com, Inc.
ggonzalez at despegar.com=20
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2007 Jul 12
0
No subject
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12:53:41.358166 IP (tos 0xb8, ttl 127, id 0, offset 0, flags [none], =
proto:
UDP (17), length: 856) 189.8.113.170.5060 > 189.8.126.177.5060: SIP, =
length:
828
INVITE sip:7002 at 189.8.126.177:5060;user=3Dphone SIP/2.0
Via: SIP/2.0/UDP
189.8.113.170:5060;branch=3Dz9hG4bKba4h2m2070fhnc4q20k1.1
Call-ID: d6dc25017b171144f35fb9e1c9c393a3 at 10.0.0.10
2009 Jan 16
0
No subject
Telco, location, ect?)
At X times of day?
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Ect, ect.
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It sounds like bleed over, which can be causes by some many things the
best place to start is to find a pattern if there is one.
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James Shigley
Monroe Telephone Answering Service
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From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of David @ULC
Sent: Tuesday, May
2009 Jul 20
0
No subject
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arp | grep "192.168.0.1"
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substituting the IP address of the SIP device.
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From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of DHAVAL
INDRODIYA
Sent: Wednesday, 28 October 2009 4:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] SIP client MAC address.
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hello,
is
2009 Jul 20
0
No subject
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arp | grep "192.168.0.1"
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substituting the IP address of the SIP device.
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From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of DHAVAL
INDRODIYA
Sent: Wednesday, 28 October 2009 4:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] SIP client MAC address.
=20
hello,
is
2007 Jul 12
0
No subject
"We have created an easy and cost effective way to have customized
recordings done quickly and with no hassle."
I thought this was rather amusing, as:
1. If you want multiple prompts recorded, you need to submit a new order for
each, which means that even prompts of a couple of words are still charged
at $12. That is NOT cost effective. You could record all your prompts as a
single
2007 Jul 12
0
No subject
What is the problem with SIP retransmits?
-----------------------------------------
Sometimes you get messages in the console like these:
- "retrans_pkt: Hanging up call XX77yy - no reply to our critical packet."
- "retrans_pkt: Cancelling retransmit of OPTIONs"
The SIP protocol is based on requests and replies. Both sides send
requests and wait for replies.
2010 Jul 03
0
No subject
eeds to be added to the database or coding of the program. Is this right? I=
sthere something else I need to do??
Ray Garza
System Analyst II
Materials Management
(956) 381-2163
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2007 Jul 12
0
No subject
to use the table asterisk.cdr but I can't find it anywhere.
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2009 Jul 20
0
No subject
[2010-03-08 14:03:49 CST] WARNING[26890] loader.c: Error loading module
'chan_dahdi.so': libpri.so.1.4: cannot open shared object file: No such
file or directory
[2010-03-08 14:03:49 CST] WARNING[26890] loader.c: Module
'chan_dahdi.so' could not be loaded.
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I am using on CentOS 5.4 64 bit.
Asterisk 1.6.0.25
Asterisk-addons
2009 Apr 23
2
CDR issue
Hello! I?ve an issue whit CDR using asterisk 1.4.23.1. I?ve configured mysql
to store cdr information, but, while I put into cdr_mysql.conf the field
?userfield=1? and doing a query I found that this field is empty in the cdr
table. On the other hand I can?t find records in the cdr table that show me
calls generated through AMI using Originate Action, that?s calls are not
stored in the CDR, but I
2007 Jul 12
0
No subject
so I
would check with him.<o:p></o:p></span></font></p>
<p class=3DMsoNormal><font size=3D2 color=3Dblack face=3DVerdana><span
style=3D'font-size:10.0pt;font-family:Verdana;color:black'><o:p> </o=
:p></span></font></p>
<div>
<p class=3DMsoNormal =
2008 May 26
5
Skype Howto
Hello all! Does anyone have a good howto to setup Asterisk and Skype.
Thanks
Gustavo A. Gonz?lez
Dto. de Infraestructura
Despegar.com, Inc.
ggonzalez at despegar.com
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2008 Aug 21
2
Asterisk and Huawei SoftX3000
Hi folks! I have a problem with our Sip provider that have a Softswitch
Huawei SoftX3000 to send us SIP calls to our Asterisk PBX, we are working
with G711 with them. They start sending calls to our pbx, some time after
they start to receive 408 messages from asterisk and some time after this
they start to complete calls normally, I don?t know what can be wrong.
Someone has configured asterisk to
2009 Jan 09
1
Web Softphone
Hi all! Im looking for 1ezphone to use as a web softphone but I?cant access
to 1ezphone.com. Anyone knows what happened with this site?. Thanks!
Gustavo A. Gonz?lez
Dto. de Infraestructura
Despegar.com, Inc.
ggonzalez at despegar.com
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2008 Jul 23
1
Broadsoft Sip provider
I am looking for a sample sip configuration of a SIP provider that runs
Broadsoft VoIP switch. My sip provider is Conecta from Brasil, that only
give me a SIP IP address to configure my asterisk box, when I call them for
support or authentication data to load on my sip.conf, they say me that I
don?t need such data, so, anyone knows how I would configure my Asterisk box
against a Broadsoft peer?
2008 Jul 01
4
Fax Between IAX Trunks
Hello! I need to send Faxes from an Asterisk box to an Asterisk + Iaxmodem +
Hylafax installed on other box. I have setup IAX trunks between this boxes,
all works fine but can?t send faxes from one to other, Im trying with or
without NVFaxDetect application but does not work. Is there a way to get it
working?. If I connect a fax machine directly to Asterisk with Iaxmodem and
Hylafax, I have no
2008 May 23
0
Asterisk chan Skype
Hello! Iam configuring chan Skype on my asterisk box, doing some test calls
I saw that asterisk answer the calls but hungs up before the call are
stablished. Is this a license problem?
Gustavo A. Gonz?lez
Dto. de Infraestructura
Despegar.com, Inc.
ggonzalez at despegar.com
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2009 Apr 15
1
pickupexten *8
Hello all!, I?ve running asterisk 1.4.23.1 and I need to get working pick up
from feature.conf. It does no work, only I can connect but cant send audio
over the phone. Is there a bug with this feature?. Thanks for any response!
Cheers!
Gustavo A. Gonz?lez
Dto. de Infraestructura
Despegar.com, Inc.
ggonzalez at despegar.com
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2008 Dec 16
0
CDR and Agents Call recording
Hello, I am running asterisk 1.4.22 and Iam recording calls in agents.conf
with the following configuration:
recordagentcalls=yes
recordformat=wav
createlink=yes
The calls are being recorded , but no entry appears in mysql cdr, and, on
the other hand I have other pbx running asterisk 1.2 that do it with the
same configuration. In cdr_mysql.conf I have:
userfield=1
accountcode=1
Is there a