Displaying 20 results from an estimated 8000 matches similar to: "SIP Registrations broken on 1.4.22.1?"
2009 Jan 28
1
asterisk-users Digest, Vol 54, Issue 94
> Date: Wed, 28 Jan 2009 13:11:19 -0600
> From: "Danny Nicholas" <danny at debsinc.com>
> Subject: Re: [asterisk-users] SIP Registrations broken on 1.4.22.1?
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
> <asterisk-users at lists.digium.com>
> Message-ID: <D32AD473FC574B41AE6A842E46549174 at db0005>
>
2009 Jun 04
6
Phones dropping registration, but asterisk thinks phones are still registered
Hi,
I have a serious problem with Asterisk 1.4.18.
Every so often, usually after Asterisk has been running for a few days
consistently, phones start dropping registrations.
However, when this happens, doing a "sip show peer" on those
extensions shows them as "OK".
Therefore, I have no way to tell this problem is happening until
customers start calling.
The only way to fix it is
2012 Oct 31
2
Asterisk and OpenLDAP
Hello guys,
i would like to implement authentication for my sip extension with an
openldap server.
Following this guide
http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/ExternalServices_id291590.html
i see a template named [sip] to map the information of sip peers into ldap.
But i'm not interested to create a template, i would only authenticate
sip extensions using username
2008 Apr 28
2
PRI hangup certain outgoing calls
I have a problem calling a certain number from our PRI line. Calling the
number from a separate PSTN phone works fine.
The remote number seems to have some funny call redivert setup, when you
call it, it answers immediately, makes some kind of beep and then starts
to ring.
Our PRI is in the UK from Telewest/NTL/Virgin Media and most outgoing
calls work without a problem. The server is
2006 Oct 11
3
asterisk 1.2.12 lost phone registrations today... why?
I lost my internet connection today for a short time.
During that time 1.2.12.1 stopped talking to my phones.
Asterisk was still working as I got 2 voicemails. I have TDM analog
cards for incoming calls.
Anyway my cisco phones had X's (lost registration) and my uniden phones
said "Registration error".
Why would phones loose registration to asterisk when the internet
connection
2009 Apr 03
1
conference calling
Greetings listers.
I'm running asterisk 1.4.21.2 on SUSE 11.0 using
Polycom 501 phones. My outgoing connections are Zapata using a TDM401P.
For the most part I can make and receive calls fine except for these 3
issues:
1. When I call an external conference, the call never bridges and
hangs up after 60-90 seconds.
2. When I call another number there is a
2008 Aug 15
1
Problem with Aastra 480ci and qualify=yes
Hi,
We have a few Aastra 480ci phones and we've noticed that in order to
get the phone to receive a call, qualify must be = no.
Apparently the Aastras do not respond to the qualify message (or
respond in a way Asterisk doesn't understand) and Asterisk thinks the
phone is unreachable.
However, this now prevents MWI from working properly on the phones.
Does anyone know how to get MWI
2008 Feb 26
1
How do I tell if T.38 was used?
I am running Trixbox 2.4 which has Asterisk 1.4.18-1
I have kind of followed:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20T.38
I added to sip_general_custom.conf
;NEEDED!!!
t38pt_udptl = yes
I did not add this to the actual SIP extension, as I assumed this being
general it applies to all sip extensions, and doing a sip show peer ext#
did indeed come up with t38pt_udptl = yes
2007 Oct 06
9
Unusable performance over WAN (part 2)
Hi all,
Disregard my previous posts, I've consolidated everything here.
I'm having terrible performance issues with samba over a WAN
(point-to-point T1 link).
Doing a copy of a 2MB file from a samba server to a linux client
running smbclient takes over 5 minutes.
SCPing the same file takes seconds.
The server is running samba version 3.0.25c with kernel 2.6.16.18.
I've put up a set
2011 May 02
3
out of the blue one way audio
Greetings List.
we're facing a strange case with my system where in the middle of the call .. after like 7 minutes (not necessarily ) the callee is unable to hear the caller however the caller is able to hear the called party. the scenario is the following.
1- 15 computers running Windows XP SP3 joining a Windows Domain Controller with DHCP , DNS, ISA Internet Acceleration Server.
2- Internet
2010 Jun 25
2
Call drops on group paging asterisk - 1.4.22.1
Hi All,
We are using group paging and our asterisk version: 1.4.22.1, but when ever
any one page to the whole group (28 extensions), the calls which are on hold
on those extensions will be dropped, is there any other way to have this
feature or to go with Overhead paging. Currently this has become a serious
problem, can anyone through some light on this group paging senario?
Thank you very much
2008 Jan 16
3
HDLC errors
I'm running Asterisk 1.2.26.1 svn rev 79171 on Trixbox 2.2. libpri
1.2.7 and zaptel 1.2.22.1. The hardware is a HP dl360 single cpu with a
TE220B. The system load is below 0.10.
I moved the server into production, with one PRI, on Friday. On that
day we handled a couple thousand calls and I only saw one HDLC abort
message. On Saturday half the calls and two abort messages an hour
2009 May 08
2
Override sip.conf settings in extensions.conf? Possible?
Hi all...
Does anyone know if it is possible to override sip.conf settings in extensions.conf
(for example: session-minse=90) without needing to create an overarching peer in sip.conf
and selecting it specifically in the dial plan?
I'm on the 1.4 stable code base and looking to implement session-timers on certain call
flows in a modular dial plan.
Thanks,
Josh Fuller josh.fuller at
2009 Jun 26
4
T38 Fax Gateway for Asterisk 1.6
Hi,
I remember seeing a T38 Gateway application for Asterisk 1.6 floating
around, but I can't seem to find it again.
Does anyone have any pointers to it? I really want to be able to send
an incoming T38 connection directly to the PSTN.
Thanks.
-- James
2008 Jun 06
2
Bad ringback tone on zap channel
Hi,
I've noticed that sometimes instead of getting a regular ring tone
when calling out on a Zap channel, I get this obnoxious loud noise
which forces me to hang up.
Is this a problem in the Zaptel driver? I seem to recall that ringback
tones are generated by zaptel when dialing out from a SIP phone over a
Zap trunk.
Thanks.
2010 Mar 27
4
Cisco 7960 become UNREACHABLE behind pix firewall
Hi,
I have about 10 Cisco 7960s behind a PIX 506E (IOS v6.3) firewall.
After some period of time, asterisk says that some of them are
unreachable, and the phones lose their registration.
The only way to make the phones recover is to clear the NAT
translation tables for the phones on the PIX (clear xlate...)
Does anyone know how to fix this? As you can imagine, it is quite
annoying. And it does not
2014 Nov 25
2
High resident memory with 11.14.0 ?
On Tue, Nov 25, 2014 at 8:14 AM, Matthew Jordan <mjordan at digium.com> wrote:
> On Mon, Nov 24, 2014 at 2:12 PM, James Lamanna <jlamanna at gmail.com> wrote:
> > Also, how big does the cache in frame.c grow to?
> > I've recompiled with MALLOC_DEBUG on that server:
> >
> > asterisk -rx "memory show summary"
> >
> > ....
> >
2008 Oct 22
7
Sonicwall potentially causing long ping times to SIP phones
Hi,
I'm having an issue where some phones behind a sonicwall are auto-congesting.
The status on "sip show peer" shows ping times anywhere from 80ms all
the way up to 1100ms.
PCs behind the same firewall have a ping time of about 30ms to the PBX itself.
Does anyone know if the sonicwall is inserting delay into the SIP
signaling path and lagging the OPTIONS messages for qualify?
2011 Dec 30
1
Asterisk 1.4.42 NOTIFY replies ignore NAT setting
Hi,
I've been trying to fix NOTIFY replies (specifically keep-alives) in 1.4.42
(I can't upgrade to 1.8.x at the moment for various reasons).
I've noticed for user agents that have a VIA header with a different
port than the port the NOTIFY was sent from,
the NOTIFY reply will always be sent back to that port, which is incorrect.
(Sonicwalls and other routers love to do this, even
2004 Aug 05
3
Avaya/Lucent Definity -> Asterisk interop question
Calling all Definity admins,
Got a few questions about Definity -> Asterisk interoperability.
1) What are the options for integration? Can I hand off extensions from the
Definity and vice versa?
2) Anybody have any working configs they would like to post?
I've found and read the legacy integration on the wiki about the two
systems. I've also googled and found a few threads that were