Thank you, Joshua!
--
sergio.
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https
I have pjsip endpoint with callerid= context=localpeers which
looks follow:
[localpeers]
exten => _.@_./sergio,1,Dial(Local/${EXTEN}@somecontext)
exten =>_.@_.,1,NoOp()
And this works fine:
== Setting global variable 'SIPDOMAIN' to 'DOMAIN'
-- Executing [EXTEN@localp
On 16/10/2020 10:11, Michael Maier wrote:
Sometimes, linphone shows missed calls as missed.
You could try to reproduce it
I can't reproduce it, it happens less than once a month.
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Sometimes, linphone shows missed calls as missed. Look like asterisk
replies with cause=487 that time, but I can't understand why.
Grandstream always shows calls as missed ones.
How should I investigate this?
--
sergio
> The CANCEL message has a Reason header with Q.850 protocol and cause
0, which doesn't mean call has been missed (should be 487).
Is this my dialplan / setup or an Asterisk issue? How can I get Asterisk
to send cause=487?
--
ser
We could extend that to runtime as well.
Would be nice!
--
sergio.
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.
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sergio.
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On 30/09/2020 14:59, Joshua C. Colp wrote:
latest version of 16 on Ubuntu
16.12.0~dfsg-1 ?
--
sergio.
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sergio.
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.
There is a from_domain option that is specified in my pjsip.conf, but
I've not found any analogue for to header.
But I've found same (unanswered) question on community.asterisk.org:
https://community.asterisk.org/t/asterisk-pjsip-how-to-force-using-domain-name-instead-of-ip-address-for-to-header/71898
--
sergio
Hello,
Open the url in the browser "https://xxx:8089/ws " address, you accept
the ssl certificate and retry the sip registry.
With that in principle ought to be solved
Best regards
Sergio Virviescas.
Telf: +34 722557601
Email: developersavs...@gmail.com
SIP/181-000a and SIP/182-000b
== Spawn extension (default, 182, 1) exited non-zero on 'SIP/181-000a'
Seems like extension 182 (called ext) is getting call and passing them
another time to me 181 (origin call)
I've try it with siemens pbx and works as expected
cheers!
Sergio
work with DIALSTATUS variable and with Dial
application I don't know how to limit to only one incoming call.
Are there any way to solve this?
Any help would be appreciated.
regards,
Sergio
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connect with unixODBC to postgresql
database
I hope this help some one.
Regards,
On Mon, 2012-02-27 at 13:49 -0600, Sergio Basurto wrote:
Thank you Jonathan,
I already do the steps you mention, my configuration is:
in res_odbc.conf
enabled = yes
dsn = asterisk-connector
pre-connect = yes
and post your extconfig.conf. I'm guessing that the reason you are
able to post CDRs in spite of not having the Connected status show up in your
ODBC show is because you are connecting with res_pgsql.conf instead of odbc.
- Original Message -
From: Sergio Basurto sbasu...@soft
Hello,
I install asterisk an postgresql 9.1 in gentoo, I already did the
configuration in both asterisk and postgresql, in fact If I make a call
and asterisk log it to CDR table, my question is:
how can I make a function like the ones in func_odbc.conf for
postgresql, if I am using
On Wed, 2012-02-22 at 06:48 -0600, Sergio Basurto wrote:
Hello,
I install asterisk an postgresql 9.1 in gentoo, I already did the
configuration in both asterisk and postgresql, in fact If I make a
call and asterisk log it to CDR table, my question is:
I make a typo mistake I mean If I make
Hello there!
I'm working on some modifications on Asterisk to adapt it to our needs
considering some particular demandings of the infraestructure we want to
provide.
Two of these modifications are:
1- A proprietary configuration driver that will communicate with a
server that will be the
this variable, please let me know.
Thanks and best regards,
Mauro.
Mauro Sergio Ferreira Brasil escreveu:
Hello there!
I'm working on some modifications on Asterisk to adapt it to our needs
considering some particular demandings of the infraestructure we want to
provide.
Two
Sorry guys.
My bad!
As you can see, the command on prior message is incorret.
I've changed to:
Dial(SIP/${EXTEN}|20|RtTL(30:6:2))
and it's working now.
Thanks and best regards,
Mauro.
Mauro Sergio Ferreira Brasil escreveu:
Hello there!
I'm testing Dial call limit option
Hello there!
The only available way to control call duration is using the RTCC patch
(discussed here https://issues.asterisk.org/view.php?id=6335; and
mainteined here http://ast.varna.net/;) ?
The purpouse is to have a way to monitor (probably on a per-minute
basis) and hangup costly calls
Hello there!
I'm testing Dial call limit option on Asterisk version 1.4.26, but
it's not working.
The issued command is: Dial(SIP/${EXTEN}|20|RtT|L(30:6:2)).
Am I missing something ?
Does it only work with Asterisk version 1.6.X ?
Thanks and best regards,
--
__At.,
Thanks a lot Faheem for you help.
I totaly understand now the approach you've used.
It's very interesting and inventive for sure.
I didn't know that I could append IP:Port info on user when using the
Dial command and that this will make calling to two different devices
registered using same
Thank you very much for all your help, Muhammad! (please let me know if
I should call you Faheem, instead).
I'll make some tests with this script on my premises as soon as possible.
Having a look on it, I couldn't realize how it really works in
conjunction with Asterisk.
I mean, it seems that
varchars(30)
Please adjust the table fields appropriately.
Hope this code block will solve you problems.
Muhammad Faheem
Software Engineer
AxVoice Inc.
307,Y Commercial,
DHA Lahore, Pakistan
+92-333-4793314
http://www.axvoice.com
--- On *Fri, 8/28/09, Mauro Sergio Ferreira Brasil
...@10.0.0.150:6060
The complete script is attached.
Muhammad Faheem
Software Engineer
AxVoice Inc.
307,Y Commercial,
DHA Lahore, Pakistan
+92-333-4793314
http://www.axvoice.com http://advcomm.net/
--- On *Wed, 8/26/09, Mauro Sergio Ferreira Brasil
/mauro.bra...@tqi.com.br/* wrote
Thanks Atis, its working pretty fine now.
Best regards,
Mauro.
Atis Lezdins escreveu:
On Wed, Aug 26, 2009 at 12:11 AM, Mauro Sergio Ferreira
Brasilmauro.bra...@tqi.com.br wrote:
Hello there!
Problem found.
For some reason, the update statement below is generated with an invalid
Hello there!
We are planning to use Asterisk on our VoIP platform, and we are
spending some brains on a way to provide the following facility: let
some SIP user (extension) registrate with more than one client (ATA,
SoftPhone, VoipCelular, etc) - what isn't a problem at all -, initiate
calls
Hi Elliot, and thanks for the reply.
I'm not completely sure you've considered that the SIP users registered
on all devices are the same.
Have you ?
I mean...
How will I use Dial command with a sequence of same devices, like:
Dial(SIP/101SIP/101SIP/101), for example ?
That's why we are
Hi Barry, and thanks for the reply!
This was the first question I've made on meeting yesterday to decide
about this facility.
Having me here today making this question should give you an idea of the
level of acceptance of my suggestion :-).
Anyway, the idea is really try to make it work with
: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mauro Sergio
Ferreira Brasil
Sent: Wednesday, August 26, 2009 12:19 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Multiple user registration
Thanks again Barry for the help and attention.
Thanks for wishing me lucky as well... If we insist on this road I'll
need it for sure :-).
I can't agree more with your position, and I'll try to be sure our
commercial demands can't be acchieved with normal approaches before
adventuring on such
Hello there!
I was testing Asterisk for the last two weeks using the Realtime driver
for MySQL, and leaving rtcachefriends=yes configured to enable MWI.
Today I started making additional tests with rtcachefriends=no because
we will probably need to use Asterisk without this cache.
For some
any idea ?
Thanks and best regards,
Mauro.
Mauro Sergio Ferreira Brasil escreveu:
Hello there!
I was testing Asterisk for the last two weeks using the Realtime
driver for MySQL, and leaving rtcachefriends=yes configured to
enable MWI.
Today I started making additional tests
Hello there!
I need some help to configure two Asterix boxes to route calls using SIP.
I followed the instructions present at this site:
http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/connecting_two_asterisk.html;,
but I couldn't get it working so far.
The only
Hi guys!
The problem was solved by the use of same password for registration
users of both boxes.
Is there no way to indicate different password for registration user of
Box1 and registration user of Box2 ?
Thanks and best regards,
Mauro.
Mauro Sergio Ferreira Brasil escreveu:
Hello
Hello there!
During some research on Internet I found the following comparison on
site Voip-Info (see, http://www.voip-info.org/wiki/view/OpenPBX.org+FAQ;):
The main points listed on Asterisk's CONS that concerned me were:
* Conferencing on Asterisk depends on Zaptel hardware and/or kernel
going to use Asterisk for. Sounds like it is
for conferencing. Would you care to elaborate?
CS
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mauro Sergio
Ferreira Brasil
Sent: Tuesday, August 18, 2009
A similar issue happens to us.
Make sure that, for inbound AND outbound calls rtp packets are reaching the
other endpoint.
If a NAT device(s) is between the endpoints make sure that the device NATs the
traffic on BOTH ways (inbound AND outbound).
Regards
On Saturday 27 September 2008 23:54:37
asterisk is on a public ip address, maybe that helps.
Take care,
--
Sergio Fabian Veltri
Director
Business IT
Of: +54-11-5217-1297 Ext. 2201
Cell: +54-911-5977-0977
http://www.businessit.biz
IT Service Management and Control Best Practices
--
Message: 5
Date: Sun
Hi,
Do you know how see the peers statuses like: sip show peers but when sip
peers are configured by Relatime method.
Thanks
0xception escribió:
yes you can use the type friend
On 5/3/07, *Forrest Beck* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
I setup sip realtime. Is it
Check this out HYPERLINK
javascript:ol('http://www.voip-info.org/wiki-Asterisk+cisco+FXO');http://w
ww.voip-info.org/wiki-Asterisk+cisco+FXO
_
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Joesph
Enviado el: Martes, 03 de Abril de 2007 02:53 p.m.
Para: Asterisk Users
How can i see if snmp is running ok on mi * box ?
Thanks in advance
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Forrest Beck
Enviado el: Domingo, 25 de Febrero de 2007 06:14 p.m.
Para: Asterisk Users List
Asunto: [asterisk-users] Re: Marks SNMP HowTo
busydetect, but no solution. Any
ideas?
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Sergio de los Santos
ssantos
Using tdm400. While transfering a call from outside to another
extensions, while this outside call is waiting with music, the
another extension call hangs up suddenly, and the call is back to the
outside call suddenly.
Wathcing logs:
Jan 15 13:32:44 DEBUG[30148] res_musiconhold.c: Read 462
firewall? i dont think so because sometimes the phone can register ok
and sudendly the appears unregistered
Leonardo Silva [EMAIL PROTECTED] ha escrito:
2006/10/31, Jon Farmer [EMAIL PROTECTED]:
Sergio R. D'Ippolito wrote:
Hi all, i have an * version: Asterisk SVN-branch-1.2-r45691, I
Hi all, i have an * version: Asterisk
SVN-branch-1.2-r45691, I need to register a linksys 922 phone thru internet and
when I make sip debug command i see this debug information:
-- SIP read from x.x.x.x:1024:
REGISTER sip:mysipserver.com
SIP/2.0
Via: SIP/2.0/UDP
How can I do to select
the channel to spy ?
thanks
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Ralph Liebessohn
Enviado el: Miércoles, 18 de
Octubre de 2006 09:29 a.m.
Para: Asterisk
Users Mailing List - Non-Commercial Discussion
Asunto: Re: [asterisk-users]
Hi list!
I have asterisk 1.2.12 installed and i need that the
users can make a logon and logoff whit theirs phones on my asterisk pbx.
Anybody know how can I do this ?
Thanks in advance.
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You have to leave a message in the voicemail, then listen it and the error
will not apear again.
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Doug Lytle
Enviado el: Miércoles, 13 de Septiembre de 2006 08:45 a.m.
Para: Asterisk Users Mailing List -
hi,
I´ need some help to implement the Grandstream GXV-3000 in my *
platform. Someone know the state of H.264 Video Codec for Asterrisk??
Thanks!!!
p.D.: appreciate any help
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Hi list!
Im using Tycho software to see my voicemail, y
can see de detail from the message but i cant hear de message.
Somebody use that software any time ? have you the
same problem ?
Thanks
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Sergio R. D'Ippolito [EMAIL PROTECTED]
wrote in message news:[EMAIL PROTECTED]...
I can login on the web http://myasterisk.com/cgi-bin/vmail.cgi
without problems but i cant see the messages on any
folder.
Thanks, Sergio.
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Try changing the configuration on your PAP2 linksys, more precisly the part
where is the NAT parameters, try changing the options from NO to YES.
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de andrutto
Enviado el: Miércoles, 23 de Agosto de 2006 03:41 p.m.
I had the same problem.
The problem was another sip extensions whit the same ip.
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Rich Adamson
Enviado el: Martes, 22 de Agosto de 2006 11:21 p.m.
Para: [EMAIL PROTECTED]; Asterisk Users Mailing List -
I can login on the web http://myasterisk.com/cgi-bin/vmail.cgi without
problems but i cant see the messages on any folder.
Thanks, Sergio.
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Hi, The link in this mail is my blog about asterisk and voip technology: http://skalog.blogspot.com Thanks! Chiacchiera con i tuoi amici in tempo reale! http://it.yahoo.com/mail_it/foot/*http://it.messenger.yahoo.com ___
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Dear guys:We are expanding our voip unit and currently looking for an Asterisk technician that can be part of our company here in Buenos Aires. If you know anyone who lives here and knows Linux and Asterisk, please contact me asap.
Best regards,Sergio Veltriwww.pointhorizon.comSuipacha 119 Primer
How about using LVS?
http://www.ultramonkey.org/3/topologies/lb-overview.html
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas Garstang
Sent: lunes, 24 de abril de 2006 17:12
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
[EMAIL PROTECTED] ha scritto:
On Wed, Apr 12, 2006 at 09:32:12PM +0200, Sergio Chersovani wrote:
[EMAIL PROTECTED] ha scritto:
context = from-sccp-intenal
I guess intenal is not the righe context :-)
Sergio
The from-sccp-internal is almost an exact copy of my from
[EMAIL PROTECTED] ha scritto:
context = from-sccp-intenal
I guess intenal is not the righe context :-)
Sergio
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WARNING[29878]: chan_capi.c:3334
show_capi_conf_error: ISDN3: conf_error 0x300b PLCI=0x103
I guess you have to set the old echo facility number in your capi.conf
Sergio
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When I was in Telefonica I+D I developed an software for windows that allows
sending sms throw an ISDN line. It was more than 3 years ago and I don't recall
to many details but we had to implement ETSI ES 201 912 and
make an V28 modem emulation over ISDN.
-Original Message-
From:
Hello:After configuring FXS and FXO channels of a TDM11B card, I can make calls from the telephone attached to the TDM11B card to the outsite (the PSTN, analog line), The problem is when I try to dial from the PSTN to my asterisk box (it has to make ring the handset on the TDM card), but I got
[0 online , 0 offline]
asterisk2006*CLI sip show users
Username
Secret Accountcode Def.Context ACL NAT
asterisk2006*CLI
What is the problem?
Thanks and Regards,
Sergio
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?
Regards,
Sergio
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I have an [EMAIL PROTECTED] installed and then upgraded to 1.2.4 with bristuff
and i have the same problem.
If you see the logs the INSERT trace has wrong values before the comand is
executed.
By the way, everyone of us that have this problem use HFC cards?
-Mensaje original-
De:
/*
cd asterisk-1.2.4
make clean
make upgrade
asterisk -r
stop now
safe_asterisk
that's all
Sergio
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Andy Webster ha scritto:
hi,
I'm trying to get the latest chan_sccp. The links from
http://chan-sccp.berlios.de are all dead. Is it just me? Does anyone
know an alternate source to get chan_sccp?
Just tested, all the links work
Sergio
Mark Johnson ha scritto:
Feb 1 09:10:33 WARNING[5327]: channel.c:784 channel_find_locked:
Avoided deadlock for '0xbf002d10', 10 retries!
Yes, the chan_sccp could lock the asterisk channel.
To fix it I need a sccp debug 10 log of the call that is locking the channel
Sergio
to a different
phone, the procedure stays the same.
You can change the softkeys order editing the sccp_protcol.h from line 1060
Sergio
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:31:15 WARNING[19727]: loader.c:391 load_modules: Loading module
chan_sccp.so failed!
you have to load the module res_features.so
Sergio
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phone that
Telefonica sells here in Spain, so i would like to make asterisk work with
that model.
Greetings
Sergio
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This is entirely SIP
The behavior is only SIP to SIP...SIP to PSTN or PSTN to SIP
= OK When one or both use speaker phone, the behavior is present.
Both Handset or Headset = OK.
How about trying with different codecs?
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was failing.
I sent all the debug logs to the draytek dev team, but they were slow on
updates to I bought a new and different brand router.
Hope they fixed that issue in the new firmwares
Good luck
Sergio
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instead chan_skinny
You have to play a bit with the configuration.
Sergio
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[EMAIL PROTECTED] ha scritto:
is it possible to use the cid of a isdn-phone as well to identify multiple
devices behind one line ?
I did not understand the question, what you mean?
Sergio
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this one works
exten = 1, hint, Zap/1
this one does not work
exten = 1, hint, ZAP/1
this one does not work
exten = 1, hint, zap/1
Sergio
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Kristian Kielhofner ha scritto:
Or you can keep using the phones with SIP and use sip_notify. I think
Ciscos support it.
In my last try it was not doing it on cisco sip phones.
Sergio
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René Enskat [Teamware GmbH] ha scritto:
-- Executing Set(SCCP/1000131-0006, Language()=de)
edit your sccp.conf and in the general section set
language=de; Default language setting
Sergio Chersovani
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.
You have to run nslookup from the machine A to understand if the record
was updated.
Set the /etc/resolv.conf to point to a ISP dns server.
Sergio
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José Luis Gómez ha scritto:
Thanks, I will try thats.
There was an issue in the ast_sip_ouraddrfor function. When the dns is
down it fails to get the right address, you can easy patch it looking to
the new code
Sergio
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-info.org/wiki-Asterisk+phone+cisco+79xx
Sergio
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at $180.00 ...
buy CON-SW-VPKG1 59 euros in europe
Pff i can bye a new equipment with this price hihihi
yep that is cisco
i can't guest the latest firmware, for me i thinks that the solution
are buy
new voip phone and put the 7910 in Dead
Yes you are right
Sergio
get a new firmware ?
You need a cisco contract.
http://froogle.google.com/froogle?q=CON-SNT-CP7910
Sergio
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fax card
solution with hylafax as software fax machine
Maybe the t38 stuff will help in the future
Sergio
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/isdn4linux/CVS-Snapshots/isdn4k-utils-CVS-2005-10-28.tar.bz2
tar xvjf isdn4k*bz2
cd isdn4*
./configure
make
make install
that's all
Sergio
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install
that's all
Sergio
Great, that's clear for me now.
Maybe a good idea to add this to the wiki page.
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http
in the newer versions
Sergio
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and of course the device are not able to establish a right
2way audio session
Let me know
Sergio
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is the one on
the top of the button 1
element in SEPmac_address.cnf.xml, I also put it into sccp.conf, but
pushing Message always dials 8500.
vmnum = 123456
in the line section
Sergio
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way to do this with SIP and the 7960? I've seen the 7914
but then I'd have to use SCCP and I'm not sure if it is stable enough
for production use.
Well give it a chance :-)
http://chan-sccp.berlios.de
Sergio
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at the site
http://chan-sccp.org/
Sergio
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The log could be more verbose than this.
Set debug = 10 in your sccp.conf
or in the console
sccp debug 10
You should see what is happening with your audio stream
Sergio
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port) the rtp packets
What firmware are you running?
Sergio
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work
Do you know if I can get it to work with both my Cisco 12 SP+ and my
ATA-186?
Well you just need to change the default tcp port
you can use chan_sccp on port 2000 and chan_skinny on port 2001
Sergio
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my console:
You may want to try the http://chan-sccp.berlios.de code
Sergio
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-Snapshots/isdn4k-utils-CVS-2005-10-28.tar.bz2
tar xjf isdn4k-utils-CVS-2005-10-28.tar.bz2
cd isdn4k*
./configure
make clean
make install
ldconfig
Sergio
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Tryt the latest chan_sccp release
http://chan-sccp.berlios.de
nr k ha scritto:
I have configured Asterisk call manager and i conneted
2 cisco ata 186 (SCCP).I make call between the ata's
through Asterisk.the phones are perfectly registered
with asterisk i am able to make calls but the call not
Chris Bagnall ha scritto:
lower soft buttons hae labels like Pnbsp;, and apart from the single
This is a old firmware issue, upgrading the phone firmware everything is
working ok with the 7960
Sergio
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