On 10/26/07, Steve Totaro [EMAIL PROTECTED] wrote:
Does realtime work reliably on Asterisk 1.2.24?
Are there any definitive guides, I can only find bits and pieces here
and there. Any accurate howtos would be of great help.
I am missing func_realtime.so. Where does this file come from?
RB == Remco Barendse [EMAIL PROTECTED] writes:
RB Hi list! Is anyone using the Kirk IP600/3 with SIP firmware
RB connected to Asterisk?
Yes.
RB Any experiences / caveats?
Make sure you keep the firmware updated. It improves rapidly.
RB If anyone would be willing to share the dump of their
On 10/26/07, Kashif Naeem [EMAIL PROTECTED] wrote:
Hello All
Has anyone integrated ARI with Asterisk 1.4 ? Is there any manual or steps
available ? Also let me know if someone know about any other similar
software.
Hi
Look at Trixbox its already integrated along with asterisk, and ISO
On Thu, Oct 25, 2007 at 04:08:11PM -0700, Dominic Son wrote:
Hi. I'm still a bit of a newb to linux, I see this in my messages log:
Asterisk init: Id ax respawning too fast: disabled for 5 minutes
What does this mean?
and how severe is it?
This is something to do with your distribution oior
David Kennedy wrote:
Hi
While I have fixed the problem from this post, I do have another
problem, and you have asked for a debug output here, so I'll go
against my better instinct and reply here :)
-- Making new call for cr 32774
-- Requested transfer capability: 0x00 - SPEECH
[
On Fri, Oct 26, 2007 at 10:46:31AM +0500, Kashif Naeem wrote:
Hello All
Has anyone integrated ARI with Asterisk 1.4 ? Is there any manual or steps
available ? Also let me know if someone know about any other similar
software.
Anybody actively maintaining ARI?
Are there bugfixes added to
On Fri, Oct 26, 2007 at 01:53:18AM -0500, Alejandro Kauffmann wrote:
1. I missed something. (modprobe zaptel, modprobe (card driver), or ztcfg)
All the above commands may change the current configuration.
Try:
cat /proc/zaptel/*
On very recent Zaptel:
zaptel_hardware
lszaptel
--
Could you please call it and confirm with me it's not working for you
either? I should probably transfer my DID number anyways, if I could only
get them to respond! Does anyone have a suggestion as to where to go in this
situation? Possibly a place with high capacity concurrent incoming calls...
I think the group numbers was just a mismatch inbetween my fiddling to
try and make it work.
The reason the channel is 6 is because once a channel is tried, it
seems to got locked somehow. Channels end up stuck in a Resetting
state.
I tried altering some settings - not using CRC4 etc - and
Dear Marc;
Thanks a lot for your kindly help.
My output of the command cat /proc/cpuinfo is:
[EMAIL PROTECTED] /]# cat /proc/cpuinfo
processor : 0
vendor_id : GenuineIntel
cpu family : 15
model : 4
model name : Intel(R) Pentium(R) 4 CPU 3.40GHz
stepping:
Firstly can I ask when the documentation site will be online again? I'm
struggling here without it.
Further to my recent post I have tried to simplify things a little.
I have used a VoiceXML app to simple call an asterisk extension. EG:
form id=transfer
block
call
Dominic Son wrote:
Could you please call it and confirm with me it's not working for you
either? I should probably transfer my DID number anyways, if I could
only get them to respond! Does anyone have a suggestion as to where to
go in this situation? Possibly a place with high capacity
Hi everybody!
Can you give me a hint, how can I start a call from asterisk with some
(php, bash, etc) script?
I need to start two calls and bride it together.
Thank you.
--
Suich
___
--Bandwidth and Colocation Provided by
Some additional debug output:
--- SIP read from 10.0.2.136:5060 ---
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 10.0.2.136:5060;branch=z9hG4bKb690d74021ef56
From: sip:[EMAIL PROTECTED];tag=C1431100-8D08-67CF-A5B0-EDB815EEBF60
To: sip:[EMAIL PROTECTED]
Max-Forwards: 70
CSeq: 1
Hint: look into asterisk call files. Create call files with PHP or
whatever and drop them into asterisk's outbound calls directory. There's
plenty of samples on voip-info.org
Regards
Rennes
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Suity
Zsolt
On Thursday 25 October 2007 18:59:52 Steve Totaro wrote:
Does realtime work reliably on Asterisk 1.2.24?
Are there any definitive guides, I can only find bits and pieces here
and there. Any accurate howtos would be of great help.
doc/extconfig.txt
doc/realtime.txt
I am missing
I've been trying to setup AddQueueMember() as a replacement
for the deprecated AgentCallbackLogin(), but I get _tree_
Queue()'s.
Massaged extensions.conf (can provide the original if need be):
- s n i p -
[default]
include = agent-loginout
include = local
; --
[agent-loginout]
On Fri, Oct 26, 2007 at 02:25:42PM +0200, Suity Zsolt wrote:
Hi everybody!
Can you give me a hint, how can I start a call from asterisk with some
(php, bash, etc) script?
I need to start two calls and bride it together.
A call file, a manager Originate event or the 'originate' CLI
Suity Zsolt wrote:
Can you give me a hint, how can I start a call from asterisk with some
(php, bash, etc) script?
I need to start two calls and bride it together.
http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out
___
Has anyone had any luck getting an asterisk box to talk to a Nortel
C15K softswitch? Or any Nortel sip products? I've been playing with
it for several days and can't seem to pass calls either direction. I
know that whike the Nortel says the C15K speaks SIP, it really speaks
nortel's
I would stay with DECT, the battery in WIFI devices only lasts a couple
of hours. (Unless you want to take the phone with you and use it on
public hotspots etc)
Zoa
Luis Antonio Prata Barbosa wrote:
Some days ago, I was looking for some mobility solutions...
My conclusion is Wi-Fi
Hello All,
I have a setup of ABE on rPath linux,Sangoma A101D, and a T-1 line (Not PRI)
which is all happily coexisting and all lights are green.
The T-1 comes in from the world into a Shark Box which splits the T into
384K data and 6 channels voice. The data side is working great. The voice
Hi List;
I established an SIP IP Trunk between Asterisk and
another softswitch (asterisk registered on the
softswitch successfully) and I saw this on the
softswitch.
From firefly softphone, I was need to do a call to be
via this softswitch (ofcourse, the softphone will send
for asterisk and
I think you can use the 'ngrep' command to see the sip packets coming in
using the sip listening port. I dont know the exact command though, you will
have to lookit up urself. you will see the sip packets coming into ur system
and in those packets you can see the response code.
On 10/25/07,
Dear All;
The codec problem resolved, I used the correct codec
should be codec_g729a_v32_pentium4m.tar.gz as the
output of the command cat /proc/cpuinfo giving:
Pentium(R) 4 CPU and my confusion was that I cared for
the output of the command core show version which
gives:
Asterisk
David Kennedy wrote:
Is there some part of the debug output I need to tell the telco about?
When I was on to them earlier today, the engineer only seemed to know
how to turn bits of their network on and off, not much about settings
I need my end etc.
Just tell them when you try to make a
Are there any asterisk users/developers who have been working with or
trialing installations of Red5?
It's an open source version of Adobe Flash Media Server -
(http://osflash.org/red5)
If so please email/call to discuss.
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
mailto:[EMAIL
Rizwan Hisham wrote:
I think you can use the 'ngrep' command to see the sip packets coming in
using the sip listening port. I dont know the exact command though, you will
have to lookit up urself. you will see the sip packets coming into ur system
and in those packets you can see the response
On Fri, Oct 26, 2007 at 06:55:12AM -0700, bilal ghayyad wrote:
Hi List;
Ip address to destination?
Unable to create channel of type SIP (cause 3 - No
route to destination)
i think you have the wrong ip information
I established an SIP IP Trunk between Asterisk and
another softswitch
David Kennedy wrote:
I think the group numbers was just a mismatch inbetween my fiddling to
try and make it work.
The reason the channel is 6 is because once a channel is tried, it
seems to got locked somehow. Channels end up stuck in a Resetting
state.
I tried altering some settings -
Rony Ron wrote:
Hi, in meantime if you have another type of digium pri
card you can plug it into your box to confirm that it's not related to
that card!
Better eliminate any doubt about that card... it made me suffer !
Well, if signalling didn't work on the D-channel, that might be a more
Dear all, I have Asterisk 1.4.13 and I need to use encryption among
Asterisk and my SIP users, and with the RTP data interchanged among
users. I prefer the use of ZRTP/SRTP because we use Twinkle and
X-Lite/Zfone as our voip clients and they support these encryption
mechanism.
My question is: do
We have a CS2K and don't have problems with Asterisk communicating at all.
The NGSS/SSTK on the CS2K doesn't have (as far as I know) support for
user and password, but IP authentication is working fine. I don't know
about the CS15K.
We did run into some issues with getting phone calls to work
I know that you can set it up to where a user hits 0 from their mailbox
and goes to an operator, but can you set up other options as well?
Could I have 0 for an operator and 1 to go to another extension? I know
you can do this by building an AA, but I don't want to have to do that
for every
On Fri, Oct 26, 2007 at 02:25:42PM +0200, Suity Zsolt wrote:
Hi everybody!
Can you give me a hint, how can I start a call from asterisk with some
(php, bash, etc) script?
I need to start two calls and bride it together.
A call file, a manager Originate event or the 'originate' CLI command
On 10/26/07, Matthew Fredrickson [EMAIL PROTECTED] wrote:
Is there some part of the debug output I need to tell the telco about?
When I was on to them earlier today, the engineer only seemed to know
how to turn bits of their network on and off, not much about settings
I need my end etc.
On Fri, 26 Oct 2007, Zoa wrote:
I would stay with DECT, the battery in WIFI devices only lasts a couple
of hours. (Unless you want to take the phone with you and use it on
public hotspots etc)
The battery in my UT Starcom F1000G lasts several days, as does the one in
my Nokia E90.
However,
Hi Dean,
The BlindSide project is using Red5. I'm not affiliated with them at all,
but I think the project looks great. It's intended to be an open source web
conferencing/webcasting platform. See: http://code.google.com/p/blindside
Cheers,
AR
On 10/26/07, Dean Collins [EMAIL PROTECTED] wrote:
Eric ManxPower Wieling wrote:
On 10/25/07, Douglas Garstang [EMAIL PROTECTED] wrote:
I'd like to grab the SIP response code that comes back from an INVITE. The
HANGUPCAUSE gives the converted ISDN cause code. Anyone know of a way to get
the SIP response code instead?
There is an RFC for
Have a look at this.
http://www.voip-info.org/wiki/index.php?page=Asterisk+phone+cisco+79xx
Glenn
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul Lacatus
Sent: Friday, October 26, 2007 12:32 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users]
Mojo with Horan Company, LLC wrote:
And it makes *clear* calls assuming you're within allowable range.
Speakerphone seems to work well too.
I meant to mention that the DTMF tones and dialtone sound like they're
played at such a high volume that they clip through the handset's
speaker. DTMF
I'm not sure if you can specify more than one mailbox= line in sip.conf,
but I had a suggestion. We have a similar AA/IVR, and we prefix info to
the callerid string to say what they chose at the menu. This might work
for you if you were going to have one SIP user check them all anyway.
On 10/26/07, Peder @ NetworkOblivion [EMAIL PROTECTED] wrote:
I know that you can set it up to where a user hits 0 from their mailbox
and goes to an operator, but can you set up other options as well?
Could I have 0 for an operator and 1 to go to another extension? I know
you can do this by
Hi everybody,
I have some 7911G to be used on an asterisk . Right now they are using sccp
protocol. I am trying to load a SIP firmware but until now the dealer failed
to do so.
Is there some howto files to use them on asterisk skinny channel ? I
understand that all configurations will be done
Hi,
On my asterisk I have a voicemail for my extension and my Soft SIP phone
tells me when there is mail for me.
I also have other voicemail boxes which are not tied to any specific
extension - rather they are for incoming callers when they interact with
an AA, such as to register for the
Mojo with Horan Company, LLC wrote:
Mojo with Horan Company, LLC wrote:
And it makes *clear* calls assuming you're within allowable range.
Speakerphone seems to work well too.
I meant to mention that the DTMF tones and dialtone sound like they're
played at such a high volume
Tzafrir Cohen wrote:
On Thu, Oct 25, 2007 at 04:08:11PM -0700, Dominic Son wrote:
Hi. I'm still a bit of a newb to linux, I see this in my messages log:
Asterisk init: Id ax respawning too fast: disabled for 5 minutes
What does this mean?
and how severe is it?
This is something
Your signalling is wrong.
The channels as programming in * should fxsks (use ks instead of ls) and
not fxols.
At Verizon's end, they use fxo and you grab it via fxs emulation in *.
Lyle
John Millican wrote:
Hello All,
I have a setup of ABE on rPath linux,Sangoma A101D, and a T-1 line (Not
Yes you can just add the second or third box after a coma on the mailbox
line like:
mailbox=3113, 8000
and the phone will alert you to messages in either box
==
Andrew Ott Email: [EMAIL PROTECTED] or [EMAIL
On Friday 26 October 2007 12:35:24 Mojo with Horan Company, LLC wrote:
lol I replied to my message in my Sent box to add some more detail...
and the reply came through to the list before the initial review...
Sorry to be confusing. There's more along soon.
If it hasn't made it through yet,
I need to know how fast a sip device needs to respond
to an INVITE sip message from asterisk before asterisk
retransmits the INVITE message again.
Thanks
__
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam protection around
The initial review is re-posted at the bottom:
Mojo with Horan Company, LLC wrote:
Mojo with Horan Company, LLC wrote:
And it makes *clear* calls assuming you're within allowable range.
Speakerphone seems to work well too.
I meant to mention that the DTMF tones and dialtone
Hi Pablo;
How the IP address will be wrong, and asterisk able to
do registeration on the destination?
If the IP address wrong, so I will not be able to
register on that IP address.
Regards
Bilal
Hi List;
Ip address to destination?
Unable to create channel of type SIP (cause 3 - No
route
On Fri, 2007-10-26 at 11:12 -0700, John Riek wrote:
I need to know how fast a sip device needs to respond
to an INVITE sip message from asterisk before asterisk
retransmits the INVITE message again.
Thanks
Snip ---
7.2.1 INVITE received
When an INVITE request is received by the
Digium said the issue was resolved in the newest zaptel. They also
suggested they would replace cards with the newer versions if there
was an issue. Might want to call Digium and hope for the best in this
case.
On 10/23/07, Joseph Begumisa [EMAIL PROTECTED] wrote:
Has anyone had any
On Fri, Oct 26, 2007 at 09:36:52AM -0800, Mojo with Horan Company, LLC wrote:
Tzafrir Cohen wrote:
On Thu, Oct 25, 2007 at 04:08:11PM -0700, Dominic Son wrote:
Hi. I'm still a bit of a newb to linux, I see this in my messages log:
Asterisk init: Id ax respawning too fast: disabled for
Alejandro Cabrera Obed wrote:
Dear all, I have Asterisk 1.4.13 and I need to use encryption among
Asterisk and my SIP users, and with the RTP data interchanged among
users. I prefer the use of ZRTP/SRTP because we use Twinkle and
X-Lite/Zfone as our voip clients and they support these
I have a new Sangoma A102 and I'm trying to get it running in asterisk. A
look through the dmesg log shows the card is detected and the various
channels created. However, when I start asterisk I get the error below.
Any ideas?
My zapata.conf is below.
Thanks,
MD
== Registered custom
Tzafrir Cohen wrote:
On Thu, Oct 25, 2007 at 11:31:37AM -0500, Tilghman Lesher wrote:
Well, if it's the same format, you can use #include.
Doesn't this break voicemail password changing?
I think Steve Murphy recent fixed the configuration handler to be able to handle
this situation.
...
On Fri, Oct 26, 2007 at 04:00:30PM -0400, Michelle Dupuis wrote:
I have a new Sangoma A102 and I'm trying to get it running in asterisk. A
look through the dmesg log shows the card is detected and the various
channels created. However, when I start asterisk I get the error below.
Any ideas?
I'm connecting a T1 PCI card to a Nortel Option 61 switch T1 card. My
Sangoma A102D shipped with 2 T1 cables - which I assume are straight
through. Do I need to make crossover cables for this scenario?
Thanks
___
--Bandwidth and Colocation Provided
In what amount of time does 100 Trying message have
to be sent to asterisk? I see asterisk retransmitting
the INVITE message multiple times before receiving the
100 Trying message.
--- David Boyd [EMAIL PROTECTED] wrote:
On Fri, 2007-10-26 at 11:12 -0700, John Riek wrote:
I need to know how
I have a new asterisk system with a T1 card. It appears that running ztcfg
-vv is required in order for asterisk to start properly.
Is this correct? Are people adding this command to the asterisk startup
script?
Thanks
___
--Bandwidth and
On 10/26/07, Michelle Dupuis [EMAIL PROTECTED] wrote:
I'm connecting a T1 PCI card to a Nortel Option 61 switch T1 card. My
Sangoma A102D shipped with 2 T1 cables - which I assume are straight
through. Do I need to make crossover cables for this scenario?
Yes - a crossover *is* needed in
Thanks. I am quite familiar with ngrep. I was asking how I could get the SIP
response code from the dial plan.
Doug.
- Original Message
From: Rizwan Hisham [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday,
For pinout info, check out: http://www.asteriskdocs.org/cables/
John Beaman
Telecom Specialist II
Voice Telecommunications Services Department.
Good Samaritan National Campus
605-362-3331
[EMAIL PROTECTED] 10/26/2007 4:01:29 PM
Michelle Dupuis wrote:
I'm connecting a T1 PCI card to a
I'm tying a Nortel option 61 to asterisk via T1. I don't want to split each
of the t1 channels out into individual lines (tied to a specific extension)
- so a trunk in and out.
Assuming PRI over T1 signaling, how would I pass the CALLED and CALLER info
across the channels so each side knows
I chased this down to ztcfg not being run (output of command you suggested
showed all channels as unsetup).
Thanks...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Tzafrir Cohen
Sent: Friday, October 26, 2007 4:10 PM
To: Asterisk Users Mailing
Michelle Dupuis wrote:
I'm connecting a T1 PCI card to a Nortel Option 61 switch T1 card. My
Sangoma A102D shipped with 2 T1 cables - which I assume are straight
through. Do I need to make crossover cables for this scenario?
Thanks
dear fellows,
for some reason when snom 360 send and receive notify/subscription
data these messages appear on asterisk console:
failed to extend from 0 to 159
failed to extend from 0 to 41
failed to extend from 0 to 49
failed to extend from 0 to 48
failed to extend from 0 to 44
etc.
(I
Hello,
I'm currently looking at FXO options to provide a POTS line to Asterisk to
trunk calls with.
Does anyone have any experience using the Linksys Sipura 3201 as an FXO device
for Asterisk?
Thanks
___
--Bandwidth and Colocation Provided by
On Fri, 2007-10-26 at 16:35 -0400, Michelle Dupuis wrote:
I have a new asterisk system with a T1 card. It appears that running
ztcfg -vv is required in order for asterisk to start properly.
Is this correct? Are people adding this command to the asterisk
startup script?
This
Michelle Dupuis wrote:
I'm tying a Nortel option 61 to asterisk via T1. I don't want to
split each of the t1 channels out into individual lines (tied to a
specific extension) - so a trunk in and out.
Assuming PRI over T1 signaling, how would I pass the CALLED and CALLER
info across the
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Conall O'Brien wrote:
Hello,
I'm currently looking at FXO options to provide a POTS line to Asterisk to
trunk calls with.
Does anyone have any experience using the Linksys Sipura 3201 as an FXO
device for Asterisk?
I just did this
Hi:
Iam using an asterisk server with astcc ,iam facing a problem with astcc that
when the call is hangup sometimes astcc doesnt calculate the call cost and the
call time and without writing the call status on cdrs table .
I tried to run this command realtime mysql status on the asterisk
Tzafrir Cohen wrote:
In Debian/Ubuntu/etc. you're actually going to need to try
Ubuntu. Not Debian. And will upstart give error messages from init
about respawn?
Thank you, I sent that message way too quick :) I should try hitting
the send button with my foot so I know it's not in
Andrew Ott wrote:
Yes you can just add the second or third box after a coma on the mailbox
line like:
mailbox=3113, 8000
and the phone will alert you to messages in either box
Thanks, I thought I'd read that you could do this somewhere before but I
couldn't find the reference. I've set
On Fri, Oct 26, 2007 at 04:35:22PM -0400, Michelle Dupuis wrote:
I have a new asterisk system with a T1 card. It appears that running ztcfg
-vv is required in order for asterisk to start properly.
Is this correct? Are people adding this command to the asterisk startup
script?
It is
There is a bug in Zaptel 1.4.5.1 that prevents other Echo Can's from being
selected.
http://bugs.digium.com/view.php?id=10555
Use 1.4.6 or 1.4.4 or edit the source yourself.
-Original Message-
From: marcotasto [mailto:[EMAIL PROTECTED]
Sent: Wednesday, October 24, 2007 9:08 AM
To:
On Friday 26 October 2007 16:13:11 Douglas Garstang wrote:
Thanks. I am quite familiar with ngrep. I was asking how I could get the
SIP response code from the dial plan.
You cannot. And you shouldn't have to. The dialplan should be generic
to all protocols, not customized to SIP.
--
Tilghman Lesher wrote:
On Friday 26 October 2007 16:13:11 Douglas Garstang wrote:
Thanks. I am quite familiar with ngrep. I was asking how I could get the
SIP response code from the dial plan.
You cannot. And you shouldn't have to. The dialplan should be generic
to all protocols,
I don't have T1 but it seems that the first time I run ztcfg (or in
fact, the zaptel startup script runs it for me) it fails. Then I need
to run it again for it to actually configure things right. So, my
(redhat-style) /etc/rc.d/rc.local contains
modprobe wctdm
ztcfg -vv
asterisk
Michelle
Zaptel creates a startup script. You just need to make sure it run/loads
fully before Asterisk starts in your bootup scripts.
This gets into tweeking your system and that varies based on the exact
OS/distro you are running.
Lyle
Mojo with Horan Company, LLC wrote:
I don't have T1 but it seems
On Friday 26 October 2007 19:47:27 SIP wrote:
Tilghman Lesher wrote:
On Friday 26 October 2007 16:13:11 Douglas Garstang wrote:
Thanks. I am quite familiar with ngrep. I was asking how I could get the
SIP response code from the dial plan.
You cannot. And you shouldn't have to. The
SIP wrote:
Tilghman Lesher wrote:
On Friday 26 October 2007 16:13:11 Douglas Garstang wrote:
I was asking how I could get the
SIP response code from the dial plan.
You cannot. And you shouldn't have to. The dialplan should be generic
to all protocols, not customized to SIP.
That's
Hi,
Have you tried Callweaver http://www.callweaver.org
___
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