Set(CALLERID(number)=XX) works here.
Also check with your VoIP provider what format they want for the number. (I
believe) most accept a 10-digit number, but I seem to remember reading
about the odd provider that wanted a leading 1.
On Thu, Jun 18, 2015 at 11:47 AM, D'Arcy J.M. Cain
So do I need 7 contexts, one for each ip address?
sean
Yes, if what you call a context is your peer definition in sip.conf. CC
routes calls through a varying number of SBCs, all (also) resolving to
callcentric.com, but each having their own name, typically
alphaxy.callcenctric.com.
I think
=no
qualify=no
disallow=all
allow=ulaw
[alpha11](callcentric-template)
host=alpha11.callcentric.com
[alpha12](callcentric-template)
host=alpha12.callcentric.com
[...]
On Tue, Apr 15, 2014 at 4:29 PM, Kai-Uwe Jensen kujen...@gmail.com wrote:
So do I need 7 contexts, one for each ip address
Great info, thanks for sharing! I was resigned to the fact that the same
account couldn't be logged into Gtalk simultaneously. (Never had issues
with being logged into Gmail, though.)
On Wed, Mar 20, 2013 at 6:57 AM, Chris Gentle gent...@gmail.com wrote:
I just wanted to send out some
File,
thanks for that quick fix! Using it now.
-- kuj
On Fri, Jan 11, 2013 at 4:09 PM, Joshua Colp jc...@digium.com wrote:
Hey everyone,
I just put in a fix for the underlying issue that was causing this to
occur. It will be out in a future Asterisk 11 release. If you want the
change now
On Thu, Jan 10, 2013 at 5:52 AM, Joshua Colp jc...@digium.com wrote:
Joshua Colp wrote:
If any of you can file an issue I'll get this sorted as soon as possible
(probably over the weekend, or maybe even sooner!).
https://issues.asterisk.org/jira/browse/ASTERISK-20916
--
Saw the same behavior earlier today. Did play around with it some more, and
things came back to normal after i removed the transport=google-v1
statement from motif.conf and restarted asterisk. (The transport statement
was required with an earlier 11.x build, at least for me.) Running 11.1.2
right
On Wed, Jan 9, 2013 at 5:38 PM, Roy Abshire r...@coopvr.com wrote:
I have the transport=google-v1 too but restarting Asterisk always solves
my problem for a day...so how do you know that fixed it?
I don't.
--
_
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Using a Gigaset C610IP here, and am very happy with the features. The base
station can handle two concurrent SIP calls, and another internal one at
that. It does it with a single SIP registration to each server. You can
setup multiple servers if you want to and define dial patterns/plans that
Yes they are. However, product numbers/bundling is a little bit
different over here. They typically sell the base bundled with a
handset already. For the current US offering, check out
http://gigaset.com/us/en//hq/en/cms/PageInternetVoIPNextGPhones.html.
I believe the equivalent of what you're
And now they've gone back and reactivated that protocol change, breaking
chan_gtalk again. Applying the small patch from
https://issues.asterisk.org/jira/browse/ASTERISK-18301 got things
operational again. Let's see how long this one goes before they go back to
the old protocol version.
--
On Mon, Aug 22, 2011 at 1:42 PM, Paul Belanger pabelan...@digium.comwrote:
I've just reverted this patch, it seems google is still making changes to
the protocol.
Yep, looks like they also reverted their protocol change. Going back to
pre-patch chan_gtalk.c makes things work again.
Wish
Yep, looks like Google changed something. Try this:
https://issues.asterisk.org/jira/browse/ASTERISK-18301
Fixed it for me.
On Fri, Aug 19, 2011 at 11:09 PM, Jim Boykin boykin...@gmail.com wrote:
We have been using gtalk channel from a long time now. It was working
fine so far but from
That patch to 1.8 was a very simple change: modify one line, add another
line. Should be easy and straight-forward to replicate on 1.4.42. (Not using
1.4 anymore over here, otherwise I would've provided the patch.)
--
_
--
Did this fix make it into 1.8.4? Getting registration errors on Cisco 79XX
in 1.8.4, going back to 1.8.3.3 everything works. I did open
https://issues.asterisk.org/view.php?id=19264 and included a SIP trace.
On Fri, May 6, 2011 at 12:24 PM, Julian Lyndon-Smith aster...@dotr.comwrote:
It was my
Currently most every phone works well, if the patch for Cisco
subscriptions gets tested then I would say any non-skype phone.
Andrew, do you have any details on this patch? Is it in the bug tracker? I'd
be happy to try it out.
I've taken a patch from
According to chapter 7 (Outside Connectivity) of the excellent Asterisk:
The Definitive Guide (review version online at
http://ofps.oreilly.com/titles/9780596517342/index.html), the following
enables secure signaling and media paths:
exten = 1234,1,Set(CHANNEL(secure_bridge_signaling)=1)
same
I've got the following in my phone.cfg:
reginfo
msg msg.bypassInstantMessage=1
mwi msg.mwi.1.callBack=*97 msg.mwi.1.callBackMode=contact
msg.mwi.1.subscribe= /mwi
/reginfo
The actual config looks good, but the structure of the XML is off. Here's
what I use (and it works):
phone1
msg
Lots of good info and pointers so far. But do keep in mind that not all
phones will automatically reboot just because you sent it a check-sync or
resync event with the sip notify command.
I vaguely remember that for e.g. the Polycoms some other condition had to be
true: either the phone's config
I've never worked with Aastras, so don't have any additional data over
what's been said by others. Also, I've never sent the SIP check-sync
notify to a phone that wasn't already registered with the asterisk server
the SIP notify was sent from. My best *guess* would be that actual behavior
of the
I use MeetMe(,Ms) in the Dialplan and if a Conference Room does't exist
Asterisk play (conf-invalid.slin)
If i use MeetMe(${room},Ms) (value from DTMF Read) and the Conference
Room doesn't exist Asterisk don't play (conf-invalid.slin) and Asterisk
Hangup the Call.
Use the i extension
Yes, I can hear the tones as I'm pressing the *1, it has to do with the
timing, pressing the buttons with two finger is quicker than with one.
Can someone provide where is the timing Francesco in previous post
mentioned.
It's featuredigittimeout (in milli-seconds). Try
featuredigittimeout =
During a call, I get the animated arrows. When I put a
call on hold, I get the flashing phone with the handset upside down. When
I
try to retreive the call, I get the animated arrow for a second,
This is normal and expected behavior so far, at least on my Polycom 500,
asterisk 1.4.27.
Here's what Philipp Kempgen wrote on this topic back in January. Nice
summary, I believe.
Klaus Darilion schrieb:
http://www.voip-info.org/tiki-index.php?page=Asterisk+sound+files+international#
German
lists a plenty of sound files for German.
Can someone recommend one for Asterisk 1.4 (any
On Sun, May 17, 2009 at 2:35 PM, David @ULC ucoms2...@gmail.com wrote:
All my calls are getting DECLINED when I am trying from xlite :
Codecs.
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There's a boat-load of articles on the web with step-by-step guidance. The
first I became aware of was
http://ronaldlewis.com/asterisk-pbx-on-amazon-ec2-how-to-guide-almost-complete/,
another good one is
http://voxilla.com/2009/2/13/asterisk-amazon-ec2-1178
Google is your friend.
On Sat, Apr 25,
I'm using
exten = s,42,SIPAddHeader(Alert-Info: info=Bellcore-r2)
and it works just fine.
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Kindly note if I run the script from shell using sudo everything looks
greate as below:
sudo /var/lib/asterisk/agi-bin/dial.pl
sudo: unable to execute /var/lib/asterisk/agi-bin/dial.pl: Success
Please read that command output again. It may not be as great as you think.
unable to execute.
I would also double-check the Sipura settings. On the Voice / PSTN-Line tab,
you have a section labeled PSTN Disconnect Detection (almost at the
bottom). Make sure that - at least for debugging this situation - the
Detect PSTN Long Silence and Detect VoIP Long Silence are set to no.
Otherwise, the
For me, the Polycom loses its subscription when asterisk is restarted.
However, as long as the phone is restarted after asterisk, everything works
fine. Worth a look. (I'm running a Polycom 500, so my firmware is older than
yours.)
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I am successfully using this in my dialplan for a number of Sipuras
(modified to fit your dialplan):
exten = 700,1,SIPAddHeader(Alert-Info: info=Bellcore-r2)
Not saying there's no other way to get it accomplished, but this is known to
work (1.4.21.2).
On Box C, you define your IAX peers A and B in iax.conf. For your peer Box
A, no special config is needed, it will use the default port 4569. For the
configuration entry for Box B, you add a line to the peer definition that
reads port=4570. Box C will then try to reach Box B via port 4570.
On
Not here. No such delay.
On Wed, May 21, 2008 at 6:39 PM, Jerry Geis [EMAIL PROTECTED] wrote:
There seems to be a 2 second delay after issueing the command
/usr/sbin/asterisk -rx sip show peers
in 1.4.20
Is there a reason why the delay? It wasnt there before.
Matter of fact I just jumped
On Fri, Apr 18, 2008 at 1:17 PM, Rob Schall [EMAIL PROTECTED] wrote:
If not, is there a way to configure this range on the phones? (They are
polycom 501s).
You're right, how could the phone read your asterisk rtp.conf setup?
Download the SIP Administrator Guide for the 501 from the Polycom
An app to invoke the Cepstral text-to-speech engine.
On Tue, Apr 15, 2008 at 11:46 AM, Zoa [EMAIL PROTECTED] wrote:
What is app_swift ?
Zoa
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When setting a forward on the phone, the phone will upload to your ftp
server a modified macaddr-phone.cfg XML file that (amongst other
locally made changes) contains an OVERRIDE statement similar to this:
OVERRIDE reg.1.fwdContact= reg.1.fwdStatus=1 ... /
Change the .fwdStatus attribute to
I misread then. Even though your original message said you wanted to
un-forward a phone. That can be done with the recipe BJ and I
outlined.
I am not aware of any way to disable the forward function, i.e.
prevent a user from forwarding in the first place.
Great info. Could you specify which version of IPCommunicator you got
to work like this?
Thanks!
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Thank You!
On Nov 9, 2007 9:33 AM, Anciso, Roy [EMAIL PROTECTED] wrote:
The version I have is 2.1.2.0. It makes for a really nice software sip
phone:) The other thing I should note is that you only need the
SEPXXX.cnf.xml file and dialplan.xml file in your tftp directory.
If this is on a RedHat-type system (EL, Fedora, but also CentOS), make
sure you have a symlink in place for libltdl.so. Even though I also
had the libtool-ltdl package installed, it only provided libs and
links for /usr/lib/libltdl.so..3.1.4 and libltdl.so.3. It did not
create a symlink to a
There have been a number of instances where recent changes in the *
code have led to a degradation of TTS in the 1.4 releases. I have no
idea whether this is relevant to ABE in general or the version you're
running. However, for a number of us the fix was to edit app_swift.c
(version 2.0rc1 from
Sure. Sorry to be unclear about it. I was using app_swift-2.0rc1, from
http://www.mezzo.net/asterisk/app_swift.html. Part of that package is
app_swift.c. At line 68, I changed the declaration
const int framesize = 160*4;
to
const int framesize = 20;
That fixed things here. As it seems, that
How are you playing the voice? Do you use something like app_swift
or app_cepstral? Just fixed app_swift for my own installation by
changing the framesize constant definition from 160*4 to 20,
after googling for a similar issue. Works like a charm now. It only
broke recently, i.e. not with the
On 8/31/07, Anthony Francis [EMAIL PROTECTED] wrote:
Mindfully wanting to use a + instead of knowing the international access
code seems like willful ignorance to me.
I beg to differ. Consider cell phones as an example. They all provide
+ keys. And it is considered a best practice to store
On 7/26/07, Patrick Buller [EMAIL PROTECTED] wrote:
This is what callwithus is supposed to forward the call to:
IAX/iaxin:[EMAIL PROTECTED]/[EMAIL PROTECTED]
Does that need to be IAX2/iaxin:[EMAIL PROTECTED]/[EMAIL PROTECTED] ?
Notice the 2? It used to be that IAX referred to v1 of the IAX
Howdy!
I'm still trying to make it onto the 1.4 releases. Almost ready to
make the switch, but here's one last thing that doesn't seem to work:
Server A calls Server B over IAX, i.e Dial(IAX2/SrvB/${EXTEN},60).
Both machines are set up in iax.conf to use RSA certificate-based
authentication.
Concur with Steve: OpenVPN is your friend. At one time, I used VPN on
Demand-type functionality in my dial plan to trunk a certain subset
of calls to a different * server via OpenVPN. This is what that
dialplan looked like:
[trunkfreecallsviaoffsite]
exten = _X.,1,NoOp
exten =
On 5/2/07, Steve Totaro [EMAIL PROTECTED] wrote:
That is really a cool idea to add it on demand in the dialplan. Was the
wait(10) required to get the VPN up or could you set it to a lower
number? It seems OpenVPN connects pretty darn quickly. Did you ever
run into issues where wait(10) was
On 5/2/07, Salvatore Giudice [EMAIL PROTECTED] wrote:
If you run it on the fly, doesn't that mean that the Asterisk user will have
permissions to configure VPN's? Nobody sees a problem with that? I thinking
that if you knock over the Asterisk service and get shell execution rights
as Asterisk,
I haven't been using Asterisk for long, but I have not yet experienced
any DNS-related oddities.
Then keep using it, and you will.
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On 4/13/07, Mike [EMAIL PROTECTED] wrote:
True, but that being said 1.6.7 did re-register every 30 seconds with no
issues.
Did the phone loss in performance after the upgrade?
Not for me. Did not observe any lesser performance.
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On 4/13/07, Mike [EMAIL PROTECTED] wrote:
reg.1.server.1.expires=30
My understanding is that the above causes the phone to re-register
with its server (asterisk) every 30 seconds. I would expect a
registration to be a heavy operation, so that does explain the high
CPU load, possibly leading to
I have a working 1.2.17 installation, where calls are successfully
passed from a slave
server to a master server via IAX2. Authentication between the
slave and master server
is set up with RSA certificates (inkeys and outkey, auth=rsa). When
invoking the dialplan
There's also an app_swift available at http://www.loopfree.net/app_swift/
--
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voice service voip
sip
session transport tcp
Last I checked, asterisk doesn't support TCP SIP signaling (or RTP
over TCP). See what happens if you change it back to the UDP default.
On 12/7/06, FaberK [EMAIL PROTECTED] wrote:
Hi to all,
I got a Cisco 2651XM wired to an E1 PRI.
What I want to
- Netem (http://linux-net.osdl.org/index.php/Netem)
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Unless there's a problem with your cut paste, you did not make the
change I proposed. Verified and working here:
exten = 8600,50,Set(CALLERID(num),2000)
exten = 8600,51,VoicemailMain(${CALLERID(num)}|s)
Notice how VoiceMailMain also uses ${CALLERID(num}, not ${CALLERIDNUM}
exten = 8600,51,VoicemailMain(${CALLERIDNUM}|s)
exten = 8600,51,VoicemailMain(${CALLERID(num)}|s)
AFAIK, CALLERIDNUM is no more.
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Sounds like the same issue as reported in bug 7982
(http://bugs.digium.com/view.php?id=7982)
May want to add your data and observations to the bugtracker.
On 9/21/06, Florian Hars [EMAIL PROTECTED] wrote:
For testing purposes, I have a Billion USB adapter connected to our PBX (P2P)
and a
Did you enable the presence feature in sip.xml? I think the phone may
need it for buddy watch (subscription). feature
feature.1.name=presence feature.1.enabled=1 ...
Default for enabled was 0.
-- kuj
On 6/9/06, Douglas Garstang [EMAIL PROTECTED] wrote:
Somewhat off topic.
We upgraded a
On 8/24/05, Eric Wieling aka ManxPower [EMAIL PROTECTED] wrote:
Try using IP addresses instead of hostnames in sip.conf. Asterisk's DNS
support is supposed to be improved in CVS-HEAD, but you should still try it.
However, using an IP address instread of a hostname in your host= line
could
On 8/16/05, Tony Hoyle [EMAIL PROTECTED] wrote:
Don Fanning wrote:
CALLED NUMBER : 1516308
Is that a valid number? AFAIK all voipbuster numbers have to start with
0 as there's no local dialing.
Assuming that number is a US number, area code 516, it should be
dialed as
I'm still having some trouble and have a few questions.
1) When defining the bmp file name...
IDLE_DISPLAY ind.anim.IP_500.38.frame.1.bitmap=astlogo
should the filename be exact or is there an assumed extension of .bmp?
It is an assumed extension of .bmp. So just put the basename into
Has anyone got this to work? Under Idle Display Animation, the
administrators guide says For example, a company logo could be
displayed..
In the ipmid.cfg file, I enabled 'ind.idleDisplay.enabled' (ie changed
it to 1), and under the IP 500 section, I added an entry for the bitmap
that I
[...]
exten = 6,4,Dial(${PHONE1}${DECT1}${DECT2}),25,tm)
Where DECT1 DECT2 are srings for SCCP/phonenr
If i specify DECT2 first and DECT1 second then DECT1 doesn't ring.
I did not see this behaviour on a non-bristuffed install of asterisk, both
phones worked as expected
On Wed, 12 Jan 2005, Paul Rodan wrote:
Yeah, it's a way for numbers to get sent faster, so you don't have to
wait
for the 3 second timeout before it gets transmitted to Asterisk. It's
similar to the dial-plan in the Sipura devices.
I don't know where it's mentioned in their documentation,
Quick question for the experts: I'm seeing stuck SIP channel(s)
scheduled for destruction stay open. This appears to happen
after (apparently successfully) registering with a SIP peer.
Any ideas where to start digging into this? I'm running today's
CVS, however the problem existed before and does
Add insecure=very to your FWD peer context in sip.conf.
The in-between fix really wasn't a fix. Chan_sip was modified some time
ago
(5/24 or so) to require authentication for inbound calls also. To turn this
required authentication off, you need to add insecure=very to your peer
definition.
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