On Tue, 8 Jan 2008, CSB wrote:
We are experiencing slightly distorted audio with playing of recordings on
our Asterisk server when the call comes in over our Eicon Diva Server BRI
card. An example is an incoming call to IVR and playing some of the standard
Asterisk voice prompts. Note that
Hello again,
Just to close this I have found the problem to be related to 1.4.10. For
some unknown reason the sip debug showed
Found description format PCMU for ID 0
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4
(ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec
If I let modules.conf autoload chan_h323.so then when
I try to stop asterisk, it *does* stop (files in
/var/run/asterisk/ are removed and connection via -vr
from another console is not possible) but the
asterisk process stays alive and stalled. In other
words, a 'ps -ae | grep asterisk' show that
On 8 Jan 2008, at 08:17, Armin Schindler wrote:
On Tue, 8 Jan 2008, CSB wrote:
We are experiencing slightly distorted audio with playing of
recordings on
our Asterisk server when the call comes in over our Eicon Diva
Server BRI
card. An example is an incoming call to IVR and playing
I've connected some analogic phone to some fxs modules on an analogic card.
I want to disable by default the call waiting sound.
I know that dialing *70 before to call the call waiting is disabled
until the next call, but isn't there a setting or a dialplan command
to set up this automatically?
hi
i am new to asterisk, kindly give me an idea that how can i relay message
sms messages from asterisk.
what do i required to relay sms messages from my asterisk box, and how i
setup the sms relaying,
is their any gateway used, or any specific SMSC. i want to make a testing
envirement having
Hi,
I have a standard E1 line, but want to receive only 10 calls
simultaneously. I want to give engaged tone to the 11th caller
onwards. Can I configure E1 to do this?
raj
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Hi!
Is there another way to prevent asterisk from rebuilding the DTMF
tones than this http://astrecipes.net/index.php?n=248 ?
I would prefer not the patch the source and rebuild asterisk.
--
Morten Isaksen
http://www.misak.dk/blog/
___
--Bandwidth
That's going to be pretty phone-specific. How about asking your phone
supplier to fix their phone so that it responds to OPTIONS correctly?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Olivier
Sent: 08 January 2008 12:50
Hi,
I've two wifi-phones
1. Nokia e65
2. HP Ipaq
I've configure two sip exten in my asterisk and using these exten in my
phones. But my Nokia phone is keep on loosing the connectivity very soon
life 1-2 min the qualify packet will be double of my HP. So, when I try to
call my Nokia SIP exten
Thanks to all who replied privately as well! ..mike..
At 03:41 PM 1/7/2008, you wrote:
Mike Trest - Personal wrote:
Hi,
Can someone point me to a zapata.conf example that will create a
single DIAL OUT group including all 4 spans on a TE4XXP?
Try:
group=0,1
channel = 1-15,17-31
2008/1/7, Kevin P. Fleming [EMAIL PROTECTED]:
Olivier wrote:
Is there way for an Asterisk server to check if a sip phone is forwarded
without bothering phone's user ?
No.
I was thinking of some Alert-Info option that would let the phone reply
with a 302 Moved Temporarily or 182
http://www.taylortelephone.com/asterisk/
_
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Klaverstyn,
David C
Sent: Monday, January 07, 2008 11:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] app_rxfax.c and app_txxfax.c
I have a standard E1 line, but want to receive only 10 calls
simultaneously. I want to give engaged tone to the 11th caller
onwards. Can I configure E1 to do this?
Yes - that can be done on the carrier side. Lines can be configured to be
outgoing or incoming only.
Christian
On Jan 7, 2008, at 6:19 PM, Matt Riddell wrote:
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Norman Franke wrote:
Greetings!
We have a somewhat noisy background in our call center, and I'd
like to
reduce this. Obviously, we could plaster the walls with sound
absorbing
material,
I've connected some analogic phone to some fxs modules on an analogic card.
I want to disable by default the call waiting sound.
I know that dialing *70 before to call the call waiting is disabled
until the next call, but isn't there a setting or a dialplan command
to set up this
Good Day All,
I am facing a serious problem since I started to use asterisk. I dont know if
it is a bug or some one already solved this.
Currently I am running 120 VoIP SIP channels on my asterisk server but each day
2, 3 calls got hanged in asterisk, and on asterisk CLI show channels
When similar problem occurred, I traced the issue to remote GSM
gateway with poor protocol stack.
The asterisk was doing exactly what it was supposed to do.
The IMMEDIATE work around we used was to put maximum call timer into
extensions.conf
exten = s, 6,Set(TIMEOUT(absolute)=3660)
2008/1/8, Steve Langstaff [EMAIL PROTECTED]:
That's going to be pretty phone-specific. How about asking your phone
supplier to fix their phone so that it responds to OPTIONS correctly?
Yes, you're right but RFC3261 doesn't specify such 302 replies.
So I'm very pessimistic about my phone
Section 11.2 of RFC 3261 details the Processing of OPTIONS Request
The response to an OPTIONS is constructed using the standard rules
for a SIP response as discussed in Section 8.2.6. The response code
chosen MUST be the same that would have been chosen had the request
been an
From: nik600 on Tuesday, January 08, 2008 6:02 AM
I've connected some analogic phone to some fxs modules on an
analogic card.
I want to disable by default the call waiting sound.
In zapata.conf
Callwaiting = no
Don Pobanz
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nhadie,
meetme requires a zaptel timing device... ztdummy is unreliable when
using meetme conferencing... I suggest you spend time elsewhere in *
until you get a digium tdm400 w/ or w/o any daughter modules... you
just need the board for the timing device you don't actually need any
Is this also the case with FC7? I have heard multiple times that FC7 has a
different/better timing method. I wonder if this will help with ztdummy.
Thanks,
Steve Totaro
On 1/8/08, dave cantera [EMAIL PROTECTED] wrote:
nhadie,
meetme requires a zaptel timing device... ztdummy is unreliable
8 jan 2008 kl. 07.41 skrev Mayur:
Hi,
I have asterisk 1.4.16 behind a NAT-FW which is using a hosted
SIP trunk for PSTN calling. Asterisk is configured to support nat
with nat=yes in sip.conf. Now the hosted PSTN Gateway supports
symmetric RTP and early media using 183 Session
glenn,
what an interesting way to use GotoIf() and 9. didn't know you
could do that in GotoIf()!
you could have used (broken out) the individual services
[trunklocal]
[trunkld]
[trunktollfree]
and just included the above individual context in with the groups that
you allowed a particular
hi dave thank you for the reply. i have loaded zap and using only
ztdummy but still can't hear anything when i dial ti my conference, i
think this explains it already. will a sangoma card do?
dave cantera wrote:
nhadie,
meetme requires a zaptel timing device... ztdummy is unreliable when
dave cantera wrote:
nhadie,
meetme requires a zaptel timing device... ztdummy is unreliable when
using meetme conferencing.
On Wed, 9 Jan 2008, Nhadie wrote:
hi dave thank you for the reply. i have loaded zap and using only
ztdummy but still can't hear anything when i dial ti my
Sorry for slow response, been away.
Stefan, thankyou. I've made the changes you suggested to my sip.conf -
and all is back to normal.
Thanks to everyone else for your suggestions.
Phil
-Original Message-
From: Stefan Guenther [mailto:[EMAIL PROTECTED]
Sent: 03 January 2008 16:28
To:
Hello
Since TDM cards are known for being particular when it comes
to motherboards (PCI 2.2, etc.), I was wondering if there is a utility
that can check that the Zaptel driver works OK and can tell if the TDM
card is compatible?
That way, if an FXO module is not reporting an incoming
Currently I am running 120 VoIP SIP channels on my asterisk server
but each day 2, 3 calls got hanged in asterisk, and on asterisk CLI
show channels showing us as call UP but in real there is no call.
When asterisk restarted the hanged calls removed from CLI with very
high duration
On Tue, Jan 08, 2008 at 07:06:17PM +0100, Vincent wrote:
Hello
Since TDM cards are known for being particular when it comes
to motherboards (PCI 2.2, etc.), I was wondering if there is a utility
that can check that the Zaptel driver works OK and can tell if the TDM
card is compatible?
On Tue, 2008-01-08 at 19:06 +0100, Vincent wrote:
Are dmesg, lspci -v, ztcfg -vv and zttool the only tools
available to investigate this issue?
I always find that looking at the files that are generated
under /proc/zaptel is very enlightening as far as showing what the
zaptel drivers are
I have several servers using ztdummy as the timing source, some CentOS
4.x, some CentOS 5.x, some Asterisk 1.2.x, some Asterisk 1.4.x.
zap show status differs between the servers:
ZTDUMMY/1 (source: Linux26) 1UNCONFIGUR 0 0 0
ZTDUMMY/1 (source: RTC) 1
We are not using any GSM Gateway for call carriers we have
Asterisk TELES(iSWITCH) --- MCI
As Teles is world class telecoms product it should not make poor protocol stack.
In my AGI script already i am using
TIMEOUT(absolute)to limit the call according to registrar balance.
I am
Sounds very similar to an issue I was having.
Are you using mISDN?
No. Incidentally, what's the benefit of using mISDN?
Regards
Cameron
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asterisk-users mailing list
To
On Fri, Jan 04, 2008 at 09:57:02PM +0100, Lars Bensmann wrote:
I can see if I can install a vanilla 1.4 off-hours and just test the
SIP-phones. Although I don't know when I will be able to do so.
OK. I tested this today it it behaved exactly like before. Hints work
for incoming calls but
Hi,
I succesfully install spandsp chan_misdn and digium card. the rxfax works
fine and I get the fax result by email.
I would like to do the same using a Patton gw + zaptel but I can't receive
fax anymore,
the call comes in from ISDN in the Patton gw, patton sends it to asterisk,
asterisk run a
Sorry i forget to give my extentions config.
[clientsG]
exten = _x.,1,Set(UserN=${CALLERID(all)})
exten = _x.,2,Set(CalledNum=${EXTEN})
exten = _x.,3,Set(Stime=${DATETIME})
exten = _x.,4,Set(CID=${CALLERID})
exten = _x.,5,Set(HCA=${HANGUPCAUSE})
exten = _x.,6,Set(Cun=${UNIQUEID})
exten =
We're doing callback here. Asterisk dials a number, waits for an answer, plays
a prompt, dials a second number, and bridges the channels together.
Calls are initiated from the AMI.
No problems there. Easy stuff.
However, I'd like to know if it's possible to have Asterisk dial the same two
Lars Bensmann [EMAIL PROTECTED] writes:
Does this really mean it just doesn't work? Nobody has working hints for
outgoing calls? I thought this should be a rather common setup.
I would have imagined so too.
Should I file a bug report for this?
I think it would be great if you did.
/Benny
On Tue, 8 Jan 2008, Abdul wrote:
Sorry i forget to give my extentions config.
[clientsG]
exten = _x.,1,Set(UserN=${CALLERID(all)})
exten = _x.,2,Set(CalledNum=${EXTEN})
exten = _x.,3,Set(Stime=${DATETIME})
exten = _x.,4,Set(CID=${CALLERID})
exten = _x.,5,Set(HCA=${HANGUPCAUSE})
exten =
8 jan 2008 kl. 21.10 skrev Lars Bensmann:
On Fri, Jan 04, 2008 at 09:57:02PM +0100, Lars Bensmann wrote:
I can see if I can install a vanilla 1.4 off-hours and just test the
SIP-phones. Although I don't know when I will be able to do so.
OK. I tested this today it it behaved exactly like
On Tue, 8 Jan 2008, Steve Edwards wrote:
or
AGI(routing.pl,--callerid=${CALLERID(all)},--exten=${EXTEN})
Oops -- assuming you use getopt_long() (or it's Perl equivalent).
Thanks in advance,
Steve Edwards
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Tim Panton wrote:
On 8 Jan 2008, at 08:17, Armin Schindler wrote:
On Tue, 8 Jan 2008, CSB wrote:
We are experiencing slightly distorted audio with playing of
recordings on
our Asterisk server when the call comes in over our Eicon Diva
You could hack it up by dropping them both into the same conference.
You'd have to tweak the messages and other conference settings, but it would
certainly work. Not as efficient as bridging though.
Tim.
- Original Message -
From: Douglas Garstang [EMAIL PROTECTED]
To:
I am having an issue with the CallerID Number not being passed to my
phone in the SIP packet. The CallerID Name is passed just fine and
displayed on the phone with no issue. I have done a NoOp() in my
extension.conf and successfully seen both the CallerID name and number
correctly. So that leads
in sip.conf under the definition for the sip user add
callerid=whatever
- Original Message -
From: Lutgring, Sam
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Tuesday, January 08, 2008 4:37 PM
Subject: [asterisk-users] CallerID Number incorrect in SIP
I am calling get_data from an agi script using Asterisk::AGI like so:
$AGI-get_data('enter-conf-pin-number');
..and I am expecting to hear the file play back when I call. I do not.
My log entry looks like this:
-- Launched AGI Script /var/lib/asterisk/agi-bin/pbx_dev.agi
pbx_dev.agi:
Anyone else have problems with phones like SPA-922, SPA-921, etc?
Inbound audio is perfect but the remote end reports audio quality
issues on the audio the handset is sending out. It's not the
network I've tried asterisk 1.2, 1.4. I've used ulaw, G726, G793
G729. Ulaw seems to be the least
I have a server behind a firewall. It is publicly addressed. Should
NOT be trying to NAT (how would I know).
The connection is a SIP trunk to Broadvoice. I am calling the
Broadvoice # from my cell and the call is being routed to my server.
With one firewall the INVITE contains information
On Tue, 2008-01-08 at 17:26 -0500, Andrew Joakimsen wrote:
Anyone else have problems with phones like SPA-922, SPA-921, etc?
If I remember correctly, the SPA-9XX phones default to sending packets
every 30ms intead of every 20ms. Log in as Admin, click on the Advanced
link, and go to the SIP
We also use the Linksys SPA IP phones for our clients. We always change this
setting to 0.020, which vastly improves audio performance.
What are peoples thoughts on changing it to something lower, e.g. 0.010?
Thanks,
Daniel Cole
-Original Message-
From: [EMAIL PROTECTED]
Yep it was set to 0.030.. but the odd thing is the issue is random and
also whenever I call my mobile phones to test it seems to work fine on
the old setting.
On Jan 8, 2008 5:48 PM, Jared Smith [EMAIL PROTECTED] wrote:
On Tue, 2008-01-08 at 17:26 -0500, Andrew Joakimsen wrote:
Anyone else
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CSB wrote:
Sounds very similar to an issue I was having.
Are you using mISDN?
No. Incidentally, what's the benefit of using mISDN?
Just that its in tree and what Digium recommends for the b410p.
I'm still not 100% about it as there seems to be
Hi,
I'm trying to setup a mobile (ericsson W300i) and I'm having some difficulties
(to pass DTMF through the mobile and to get sound). I'd like too know how
could debug what are the common way to debug get information.
remote mobile = mobile on asterisk (by bluetooth) = asterisk
I'd like to
Can you describe the issue more please? Can the remote person not hear you at
all? Or is there distorted/broken voice?
Cheers,
Daniel Cole
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Joakimsen
Sent: Wednesday, 9 January 2008 9:26 AM
To:
Hi,
I just installed Antonio Gallo's agx-ast-addons package
in order to use app_txfax with asterisk-1.4.
Compiling according to docs went well.
However, I'm getting an error after the first page
of fax:
/usr/src/agx-ast-addons/app_txfax.c:438 txfax_exec:
Transmission loop error
The (very
On Tue, Jan 08, 2008 at 09:47:40PM +0100, Johansson Olle E wrote:
There is a setting for enforcing the call limit for both inbound and
outbound on a peer only.
Thanks for pointing me in the right direction. The limitonpeers=yes was
already set as I read in the documentation. But I set it in
Hello,
What is the maximum WAV specs that can be used with asterisk
recordings for the Background() application?
Also, is there a place where someone can provide a custome dialplan
autoattendent for free?
--
-Shane
Blog: http://blind-geek.com/blog/
CoOwner: http://sjtechzone.com
AIM: inhaddict
This issue of phone vendors not supporting OPTIONS according to RFC 3261
often comes up on this list. Like Kevin Fleming said, an OPTIONS request is
supposed to be responded in the same way as an INVITE. Almost all SIP phone
vendors have construed OPTIONS as some kind of a keep-alive request,
Replying to myself. :)
I just noticed the deadlock message still displayed on the console at the end
of a normal call, so the the deadlock message is not related to the early CANCEL
- Original Message
From: Douglas Garstang [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent:
Hope someone can help.
I have a situation where asterisk is sending a SIP CANCEL message before the
Dial() timeout has hit. It doesn't always do it.
Normally, we send an INVITE to the ITSP. They respond with a 100 Trying, then a
180 Ringing, or 183 Session Progress. It seems to be at this
The issues i have been having are probably similar to the original
message, I use the Linksys 9XX Series phones and we used to always
receive complaints from the person we were calling that they could
hardly hear us.
I fixed this by:
Going into the Phone section of the config and setting the
I have found with a number of clients to who we have installed the LinkSys
phones, that when you get the input gains to 6, that the phones have a tendency
to pick up too much background noise. Have you experienced this at all?
Cheers,
Daniel Cole
-Original Message-
From: [EMAIL
robert,
with limited info below, are you port forwarding on the router with the
public IP, ports 10,000-20,000, 5004, along with 5060? and the other
router (internal, I assume)???
how do you have two firewalls configured with one * box?
do you have captures on both sides of the internal (I
No, I haven't experienced this.
I think were lucky because most voip phones are in there own offices, I
will check with our sales manager this afternoon who sits in the call
center and see what the background noise is like on her phone.
I guess i'm just lucky that its a quiet environment, But
steve,
thanks for pointing that out, I forgot the exact reason.
as for the hearing/audio problem... if all else works the conferencing
should also... I haven't used freepbx, do they handle the port
filtering?
# tcpdump -i eth0 udp
should show if the packets are getting in/out...
I have
On Tuesday 08 January 2008 19:46:50 Shane D wrote:
What is the maximum WAV specs that can be used with asterisk
recordings for the Background() application?
All recordings must currently be in single channel, 8kHz format. The
maximum length of an uncompressed wav file is approximately 38
hours
Hi Steve,
I see. I have this now,
*CLI zap show channels
Chan Extension Context Language MusicOnHold
pseudodefault en
*CLI load chan_zap.so
Unable to load module chan_zap.so -- on the log file it says, it as
already loaded that's why it's unable to
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Hash: SHA1
Nhadie wrote:
Hi Steve,
I see. I have this now,
*CLI zap show channels
Chan Extension Context Language MusicOnHold
pseudodefault en
That means the zap channel should be ok.
One thing you could do is go
I will be out of the office on Wednesday, January 9, 2008. If this is an
emergency, please call Customer Service at (877) 791-7700. Thank you.
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To
Ok, no worries :)
Most of our clients have a relatively open common work area, where the phones
are located. I would be interested to know what your sales manager has
experienced.
Cheers,
Daniel Cole
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of
Hello all,
is there any way to tell asterisk what port to use for source of any
registration request?
for example the simple register command,
register = user:[EMAIL PROTECTED]:port
will send the register packet from asterisk_IP:5060 to proxy:port .
Is there anyway to have asterisk to use
As using OPTIONS requests main benefit is to non-phone specific, what shall
we do when most vendors do not comply with RFC ?
2008/1/9, Raj Jain [EMAIL PROTECTED]:
This issue of phone vendors not supporting OPTIONS according to RFC 3261
often comes up on this list. Like Kevin Fleming said, an
2008/1/8, Jean-Louis curty [EMAIL PROTECTED]:
Hi,
I succesfully install spandsp chan_misdn and digium card. the rxfax works
fine and I get the fax result by email.
I would like to do the same using a Patton gw + zaptel but I can't receive
fax anymore,
which patton product do you use ?
how
9 jan 2008 kl. 02.48 skrev Raj Jain:
This issue of phone vendors not supporting OPTIONS according to RFC
3261
often comes up on this list. Like Kevin Fleming said, an OPTIONS
request is
supposed to be responded in the same way as an INVITE. Almost all
SIP phone
vendors have
9 jan 2008 kl. 06.55 skrev Al lists:
Hello all,
is there any way to tell asterisk what port to use for source of any
registration request?
for example the simple register command,
register = user:[EMAIL PROTECTED]:port
will send the register packet from asterisk_IP:5060 to proxy:port .
Hello,
I'm using Asterisk with Realtime extensions and ODBC CDR. And I'm have
some trouble with the CDR userfield that is not changed when using the
SET command in the realtime dialplan.
In my dialplan (extensions.conf, the file) I'm setting the userfield
like this :
exten =
Hi Matt,
I tried
/usr/local/src/zaptel-1.2.22.1# ./zttest -v
and it just freezes at this.
Opened pseudo zap interface, measuring accuracy...
no more outputs, when i cancelled this is what i got.
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