Displaying 20 results from an estimated 500 matches similar to: "Cirpack KeepAlive packets causing SIP errors"
2008 Feb 11
1
SIP Bad request protocol Packet on Asterisk 1.4.18
Hi all!!
I have a really weird problem. I upgraded 2 Asterisk 1.2 boxes to Asterisk
1.4.18. Both are home PBX's and both boxes register to a SIP DID at
exactly same provider. One box runs without errors on the console, the
other box keeps repeating :
[Feb 11 23:40:29] WARNING[11292]: chan_sip.c:6705
determine_firstline_parts: Bad request protocol Packet
When i set debug on, it seems to
2008 Feb 13
2
MWI problem with Siemens Gigaset S675 IP
Hi list,
Before purchasing a number of Siemens DECT SIP phones, the Gigaset
S675 IP, I read that the problems with MWI had been fixed with the
latest firmware version (see
http://www.voip-info.org/wiki/view/Siemens+Gigaset+S675IP). Now I'm
not so sure that's the case.
After setting up a network mailbox for one of these phones, as well as
an Asterisk voicemail account (ext.
2007 Dec 28
2
Problems with zaptel and HFC-S PCI card
Hi list,
Now that I've got my Asterisk server to recognize my HFC-PCI card, I've run
into some serious problems. The first thing I noticed was this message
that would show up every five seconds on the CLI:
Dec 27 15:46:42 WARNING[12484]: chan_zap.c:2512 pri_find_dchan: No
D-channels available! Using Primary channel 3 as D-channel anyway!
== Primary D-Channel on span 1 down
2008 Feb 27
1
SPA3102 registration problem
Hi list,
After failing to get a Sipura/Linksys SPA3000, which I've configured
as a PSTN gateway, to pass on the Caller ID, I decided to try my luck
with a Linksys SPA3102 after hearing some promising stories.
Unfortunately, I've run into a completely new problem: it seems
Asterisk won't let this device register.
I went about configuring the SPA3102 in much the same way as I
2008 Apr 02
2
Howto connect to Cirpack softswitch with Asterisk ?
Hi,
has anyone connected Asterisk to Cirpack softswitch sucessfully ? Any howto
or more info about needed Asterisk SW and setup ?
Thanks in advance,
regards,
Rob.
2007 Oct 17
1
Portscans and Asterisk
Anything to do about portscans? Is there any way (should I) to see
if the connection is a legit (only SIP currently) connection BEFORE
my * answers?
[2007-10-17 19:23:46] WARNING[4191]: chan_sip.c:6624 determine_firstline_parts: Bad request protocol 01@<ASTERISK_IP> SIP/2.0
-- Executing [s at default:1] Answer("SIP/sip.jmg.se-081dd730", "") in new stack
[2007-10-17
2013 Mar 19
3
SIP account registration fails after upgrade to 1.8
Hi folks,
Following an upgrade from Debian squeeze to wheezy, and Asterisk 1.6.2.9
to 1.8.13, my server is no longer able to register a connection to a SIP
account at my ISP (XS4ALL in the Netherlands). At the same time, it is
still able to register a different account with another SIP provider, so
it must be that they no longer have the same basic requirements.
The relevant part of my
2009 Sep 02
1
Voipbuster not ringing, other SIP peers are ringing...
Does anybody else see the same behavior for VoipBuster connections?
When I trace one of the other SIP peers, I see it sends this message:
----------------------------------------------------------------------
<--- SIP read from 82.101.62.99:5060 --->
SIP/2.0 180 Ringing
Allow: INVITE,ACK,BYE,CANCEL,PRACK,SUBSCRIBE,NOTIFY,UPDATE
Call-ID: 740540ee64fa957513ce89f03ef5e3f2 at sip.xs4all.nl
2007 Dec 30
1
Zap channels for HFC-S PCI card not responding
Hi list,
After upgrading from Asterisk v1.2 to v1.4.14, all kinds Zaptel error
messages related to my HFC-S PCI card disappeared, but now I can't
access the card's resources because it always seems to be busy. Any
idea why?
Thanks,
Jaap
PS -- Below is some info regarding my configuration.
===========================
Zaptel version: 1.4.7 (incl. firmware and modules).
OS: Debian
2005 Feb 23
0
Digium TE405P and Cirpack Switch
Hi,
I've got an * box with 4 E1 (TE405P) and a Class 4 Cirpack switch
(www.cirpack.com).
<IP Network>--<*>--<Cirpack>--<Public PSTN Network>
ISDN interco is EuroIsdn, ports are configured ccs, hdb3, crc4. Cirpack
is Network, * is Terminal/User.
As I encountered some pb with Sip to Zap transcoding (* to Cirpack way
poor quality, the other way fine), I tryed to
2023 Oct 16
2
creating a time series
Hello everyone,
? had 15 minutes of data from 2017-11-02 13:30:00 to 2022-11-26 23:45:00 and number of data is 177647
? would like to ask why my time series are less then my expectation.
baslangic <- as.POSIXct("2017-11-02 13:30:00", tz = "CET")
bitis <- as.POSIXct("2022-11-26 23:45:00", tz = "CET") #
zaman_seti <- seq.POSIXt(from = baslangic,
2005 Feb 28
0
Pb DTMF with Asterisk vs Cirpack Transit Node
Hi all,
I've a problem in DTMF dialog between * and a Cicpack Transit Node Class 4-5.
The call initiat by a mgcp phone pass by the cirpack and arrive in SIP on *,
everything is ok (negociation and phone call) but when we try to use the
voicemail, Asterisk don't understand DTMF.
Here are some logs (SIP debug on) on a DTMF '2' receive :
2005 Feb 28
0
Pb DTMF with Asterisk vs Cirpack Transit, Node
Salut Guy,
I have the same problem with a Cirpack (B3G carrier)
What I see is that you use sip info to detect DTMF.
The problem is that there is no normalisation on the content of the sip
info frame for dtmf detection.
First, asterisk try to detect the header "application/dtmf-relay"
and you have the header "application/dtmf"
see line 6069 of /channels/chan_sip.c function
2005 Mar 01
0
RE: Pb DTMF with Asterisk vs Cirpack Transit, , Node
Salut Guy,
I have the same problem with a Cirpack (B3G carrier)
What I see is that you use sip info to detect DTMF.
The problem is that there is no normalisation on the content of the sip
info frame for dtmf detection.
First, asterisk try to detect the header "application/dtmf-relay"
and you have the header "application/dtmf"
see line 6069 of /channels/chan_sip.c function
2008 Jan 03
2
HFC-S zap channels always busy
Hi list,
Attempting to get an ISDN-BRI line connected using an HFC-S PCI card
together with Asterisk v1.4.14 and Zaptel 1.4.7 on a Debian etch
system, I find that I can't access the card's resources because the
channels are always be busy. An attempt to call out results in the
following CLI output:
== Primary D-Channel on span 1 down
== Primary D-Channel on span 2 down
2023 Oct 16
1
creating a time series
Why did you expect to have 177647 elements ?
I found that 177642 is the correct number:
Marc
baslangic <- as.POSIXct("2017-11-02 13:30:00", tz = "CET")
bitis <- as.POSIXct("2022-11-26 23:45:00", tz = "CET")? #
zaman_seti <- seq.POSIXt(from = baslangic, to = bitis, by = 60 * 15)
y2017_11_02 <- seq(from=as.POSIXct("2017-11-02
2023 Oct 16
1
Ynt: creating a time series
hello,
because ? have data between these times and it has 177647 elements
________________________________
G?nderen: Marc Girondot via R-help <r-help at r-project.org> ad?na R-help <r-help-bounces at r-project.org>
G?nderildi: 16 Ekim 2023 Pazartesi 13:43
Kime: r-help at r-project.org <r-help at r-project.org>
Konu: Re: [R] creating a time series
Why did you expect to have
2004 Oct 05
2
SIP multipart mime messages
I was messing about integration of a Cirpack softswitch with Asterisk
and banged my head against a problem previously noted on the list.
http://lists.digium.com/pipermail/asterisk-users/2003-November/026436.ht
ml
What is the status of this problem? Has it been fixed? I scrambled
through chan_sip.c, but couldn't find ay reference to "multipart".
Regards,
Jesper Dalberg
2003 Nov 06
6
Error in Incoming SIP call
When I get a SIP call, I get the following error:
*CLI> NOTICE[1133718080]: File chan_sip.c, Line 1768 (process_sdp): Content is
'multipart/mixed;boundary="unique-boundary-1"', not 'application/sdp'
WARNING[1225991360]: File pbx.c, Line 1154 (pbx_extension_helper): No
application '' for extension (incoming, 5147771111, 1)
== Spawn extension (incoming,
2008 Feb 07
3
Need good voicemail documentation
Hi list,
After wrestling with the voicemail system for a while (Asterisk
1.4.14, Debian etch), I got it to work, but I still have lots of
questions, like:
* Why can't I delete any voicemail messages?
(Response: "Message undeleted.")
* Why can't I listen to the messages in the Old folder?
* Why can't I use the advanced options?
(Response: