Hi, I 'm using spandsp-0.0.2pre26 , and thereis a bug adding headers: LOCALHEADERINFO and LOCALSTATIONID (I can't see them ). But faxes goes using rxfax and txfax fine. I also have tried development versions, the bug is fixed, but I get bad faxes (I get one page, but my tiff consists of three
I am having a problem with an Asterisk server, in that when it
is receiving a call from another Asterisk server using an IAX2 trunk the phone
rings for 10 ms and then there is a hungup from asterisk and then the phone
rings again before another hangup.
The funny thing is that after I
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Would anyone happen to have a working configuration for the 7971G-GE
(running SIP70.8-0-4SR1S) they would care to share, or allow me to purchase.
Hi Kelvin!
I have Cisco 7970 and firmware SIP70.8-0-2SR1S and I use Another
SEPmac.xml.cnf
In article [EMAIL PROTECTED],
Michiel van Baak [EMAIL PROTECTED] wrote:
On 20:50, Wed 25 Oct 06, Tony Mountifield wrote:
In fact if you do make samples in your asterisk directory, it will
install default configuration files in the right place for you.
do _NOT_ i repeat _NOT_ do this if you
Hi,Reading from www.voip-info.org, i can see that Junghann's chan_capi is now part of bristuff , as of version
0.3.0-pre.What does that really mean ?Shall I understand I can share a Junghanns QuadBRI board between 2 CAPI-enabled software (like a 0.3.0-pre bristuffed Asterisk for instance) ?If
On 2006-10-25 22:33:47 -0700, John Marvin [EMAIL PROTECTED] said:
Martin Joseph wrote:
Transcoding is a bigger hit then mixing as i understand it.
If all the conference members are using ulaw for example, then having
the playback material encoded in ulaw is the big winner. If there are
On Thu, 26 Oct 2006 07:37:40 +0200, Rajkumar S
[EMAIL PROTECTED] wrote:
Hi Lenz,
On 10/25/06, Lenz [EMAIL PROTECTED] wrote:
if you use Local channels for agents (or callback agents), you can
easily
do this in the Dial() command after the Local channel is called.
I am using call back
On 06:54, Thu 26 Oct 06, Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Michiel van Baak [EMAIL PROTECTED] wrote:
On 20:50, Wed 25 Oct 06, Tony Mountifield wrote:
In fact if you do make samples in your asterisk directory, it will
install default configuration files in the right
Hi all,
I have got same problem - bridging between IAX and IAX goes fine without lost
packets. ZAP to IAX - one lag show lost packets. Any ideas and/or solutions?
Best regards,
Dmitry
-Oprindelig meddelelse-
Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Simone
On 09:39, Thu 26 Oct 06, Olivier wrote:
Hi,
Reading from www.voip-info.org, i can see that Junghann's chan_capi is now
part of bristuff http://www.voip-info.org/wiki/view/Bristuff , as of
version 0.3.0-pre.
What does that really mean ?
Shall I understand I can share a Junghanns QuadBRI
I have a TE205P, jumpered for E1, added the missing wct4xxp-line to
/etc/modprobe.d/zaptel, zaptel.conf is just
span=1,1,0,ccs,hdb3,crc4,yellow
span=2,2,0,ccs,hdb3,crc4,yellow
bchan=1-15
dchan=16
bchan=17-31
bchan=32-46
dchan=47
bchan=48-62
Which, according to my reading of the documentation
I'm using 7920 with chan_skinny (from 1.4branch), it working quite well
I reported some chan_skinny bugs in bugtracker and I can confirm, that
are solved very quickly :-)
Is true, that chan_skinny have less features that chan_sccp, but more
important for me is active development/maintenance
Hi Lenz,
On 10/26/06, Lenz [EMAIL PROTECTED] wrote:
[agents]
exten = _2XX,1,Dial(SIP/${EXTEN})
In this dial command you're free to add whatever option you may like,
including the ones to limit call length.
I hope this helps
That did help. Thanks a lot!!
raj
Hi There,
Im not sure about IPF-2600 but on IPF-2200L, it's 12345678 for web access
and 1234 on the phone itself. Give it a try it might be the same for the
your model too.
I just want to know if you are satisfied with the phone or not IPF-2200L is
unsatisfactory in different aspects. First is
Hi,
I have a query regarding pulse dialing at 20 pps.
An Analog Phone is directly connected to the FXS port of Asterisk PBX.
When the analog phone pulse-dials at 20 pps, the pulse digits were not
decoded correctly by Asterisk. For e.g. when the user dials a 2,
Asterisk decodes the pulse digit
[sip_proxy-out]
type=peer
outboundproxy=QUINTUM_IP
You should not be using outboundproxy. Use host=QUINTUM_IP.
OK, I make this, and when I make call, I see in * sip debug mode, that Quintum
send this call to * (SoftPhone-Asterisk-Quintum-Asterisk). So, what I
Hi,
Does anyone know how I could configure the make/break ratio of pulse
dialing in Asterisk ?
regards,
Kwang Mien
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Hi,
i am from Germany, so excuse my School English.
I use Asteriks 1.2.12.1, zaptel 1.2.9 and mISDN Rc25 - since my update
of Asterisk 2 wooks ago, Echos accure in my SIP Calls.
I use SNOM 360, sometimes there is no echo (for example if i call myself
via
Hi there,
I had the same configuration and it nearly took me a week to solve the
problem and atlast I'm not sure if what I've done is the right way.
I need Phone-Ast-Quintum-PSTN, so i defined a trunk in quantum with
proper fxo lines in it then a hop off in the quintum with proper extension
that
Dear Sir/Madam,I have a problem with the OOH323 Channel driver. The problem is that I am currently registered to the a Gatekeeper and there will be other endpoints be registered to the GK also. The Asterisk box is positioned as a PSTN gateway and then the endpoints registered to the GK. The GK
Lets talk about SIP and IAX2
1. The good and bad of both
2. What is the better one and why
3. and any other information that maybe use full
--
Best regards,
Al Bochter
Bochter Services
(Voip PBX) Toll Free: 866-638-1254 EXT: 250
(Voip PBX) Free World DialUp: 780217 EXT: 250
(Voip) Cellular:
A client used to use BT isdn30 and ported the numbers to telewest
several years ago.
Now, the client moved to adept telecom. I *think* adept resells BT
products. We got new numbers from adept (bt?) and the old pbx on the
telewest lines forwards the calls to the new numbers.
On the adept line I got
Hi,
I am using Dial command with L option as follow.
L(15:12)
The function works well in normal call (IP phone - PSTN) and the call
dropped when the time is up. However, it doesn't work in forward
case, (IP phone1 - IP phone 2 (forward to) - PSTN). In the forward
case, there is no
Hi,
The PSTN connection is via a zaptel card, rather than a sip peer.
With thanks,
Tim
On 2006-10-24 06:44:01 -0700, Wildheart
[EMAIL PROTECTED] said:
Hi,
Does anyone know a what to use a different codec for calls which a
re
handset to handset (eg, G711) then when we have
Dan, do you use ooh323 from asterisk 1.2.x or 1.4 branch?
PJ
Dan Austin wrote:
PJ Wrote:
Dan, can you supply your ooh323.conf for me? I would like resolve my
issue with not recognizing dtmf by ooh323 from callmanager
my ooh323 is quite simple, also on callmanager config page for gateway
I need to determine the number of active calls in a group from outside
of Asterisk. Currently I poll the manager API and parse the channel
status list but this is becoming too expensive on CPU.
What are my options? What is considered standard practice ? Update a
DB field? Poll the manager
On Thu, 2006-10-26 at 12:18 +0200, Stefan Agethen wrote:
Hi,
i am from Germany, so excuse my School English.
I use Asteriks 1.2.12.1, zaptel 1.2.9 and mISDN Rc25 - since my update
of Asterisk 2 wooks ago, Echos accure in my SIP Calls.
I use SNOM 360, sometimes there is no echo (for
I have an Asterisk servers (recent SVN version 1.2) and two Sipura ATAs
(one 2000 and one 1001).
I have Three-way Conf Serv and Three-way Call Serv enabled on both ATAs.
When I make a SIP call from phone 1 to phone 2 on my Asterisk box, it
works fine, then when I press the hookflash on phone 1,
On Thu, 2006-10-26 at 06:51 -0400, Al Bochter wrote:
Lets talk about SIP and IAX2
1. The good and bad of both
2. What is the better one and why
3. and any other information that maybe use full
like this?
http://www.voip-info.org/wiki-IAX+versus+SIP
I am having a problem getting the following logic to work,
in a macro.
Basically, if the caller ID matches, set the outbond trunk
to a Zap channel, otherwise use a SIP provider.
exten = s,n,Set(TRUNK=${IF($[${CALLERIDNUM} = 1234567890]?Zap/g1:SIP/LDPROVIDER)})
; use PRI instead of
Hi Alex and dev team,
I've just checked the demo on your site and going to install it without any
time waste. It's absolutely marvelous and there isn’t any free, save as to
open source comparable project on the web. You would have plenty of users
with no doubt in near future.
Regards
---
M.
I know, I know, the wiki link for that one.
But wot I wanted were actual figures related to Asterisk n QoS.
How does Asterisk actualy handle and fare at the following QoS issues :
1) Delay
2) Jitter
3) Packet loss
These and more ideas are welcome.
cheerz
- Ben.
On Thu, Oct 26, 2006 at 06:51:39AM -0400, Al Bochter wrote:
Lets talk about SIP and IAX2
1. The good and bad of both
2. What is the better one and why
3. and any other information that maybe use full
Let's be more specific. For connecting phones? For interconnecting PBXs?
Have you did some
iax using one port, that is good if going through firewalls and is
efektive when trunking multiple calls,
but not using tcp, so it is not so great (tunneling through ssh is not
possible)
iax using own jitterbuffer, that isn't interoperate with generic
jitterbuffer used in SIP
iax-iax
Hi all,
Can tell me somebody what meen : channel.c: Avoided initial deadlock
Our customer makes calls with our softphone (with IAX2).
Sometimes the softphon freezes. The call is ACTIVE but the user cant
hang it up.
At this time in the log file (asterisk/messages) appear the next line:
On Thu, Oct 26, 2006 at 10:48:52AM +0200, Michiel van Baak wrote:
the chan_capi is merged with BRIStuff. This means you no
longer have to download and compile chan_capi manually when
you want to use a CAPI board with bristuffed asterisk.
And while we're at it: How does the bristuff chan_capi
On Thu, Oct 26, 2006 at 12:18:20PM +0200, Stefan Agethen wrote:
Hi,
i am from Germany, so excuse my School English.
I use Asteriks 1.2.12.1, zaptel 1.2.9 and mISDN Rc25 - since my update
of Asterisk 2 wooks ago, Echos accure in my SIP Calls.
I use SNOM 360, sometimes there is no echo
why not using a zap show command and parse the results externally?
l.
On Thu, 26 Oct 2006 13:12:46 +0200, Nick Adams [EMAIL PROTECTED] wrote:
I need to determine the number of active calls in a group from outside
of Asterisk. Currently I poll the manager API and parse the channel
status
Dan, do you use ooh323 from asterisk 1.2.x or 1.4 branch?
PJ
Both(ish) I've run it on 1.2, and have 1.4 in testing now.
The (ish) is my production servers are actually a very old
trunk checkout that is closer to 1.2 than 1.4.
Dan
___
--Bandwidth
zttool is your friend here
red is LOS or no signal coming in
On Oct 26, 2006, at 3:54 AM, Florian Hars wrote:
I have a TE205P, jumpered for E1, added the missing wct4xxp-line
to /etc/modprobe.d/zaptel, zaptel.conf is just
span=1,1,0,ccs,hdb3,crc4,yellow
span=2,2,0,ccs,hdb3,crc4,yellow
Hello Users,
Good Morning,
In Conferemcing How to Disconnect the phone while in between the
Conference .
When I press the ' # ' key for Disconnecting the
Conference..
Below the Following to shows some Warning, ( in Red Color )
from-sip en
*CLI -- Executing
On Thu, 2006-10-26 at 14:23 +0200, Pavel Jezek wrote:
with iax I have still problems with messages like:
[Oct 26 12:58:30] NOTICE[11417]: chan_iax2.c:7075 socket_process: Peer
'wilder' is now TOO LAGGED (2014 ms)!
[Oct 26 12:59:37] NOTICE[11415]: chan_iax2.c:7075 socket_process: Peer
I use SNOM 360, sometimes there is no echo (for example if i call myself
via SIP-Asterisk-SIPProvider-TELEKOM-ISDN)
but if i call other people there occures Echo many times. The Routing is
always the same :
SIP (SNOM) - Asterisk - VoIPProvider - ISDN/POTS
Can i control the
-Original Message-
From: Tony Mountifield [mailto:[EMAIL PROTECTED]
Sent: Wednesday, October 25, 2006 2:51 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: Meetme... No channel type registered for
'zap'
In article
[EMAIL PROTECTED],
Douglas Garstang [EMAIL
Folks,
Anyone know if Asterisk supports IPv6? If not, is support planned?
TIA,
David A. Bandel
--
Focus on the dream, not the competition.
- Nemesis Air Racing Team motto
___
--Bandwidth and Colocation provided by Easynews.com --
We're using it against Telefonica (3 E1 channels) and a Meridian (3 E1
channels) in 2 asterisk boxes interconnected via IAX without problems. I'm
using Asterink 1.2 and Unicall 0.0.3, but it worked well with asterisk 1.0
and Unicall 0.0.2 too. We are handling peaks of 15.000 calls/day with 40
On 26 Oct 2006, at 11:59, Conrad Wood wrote:
A client used to use BT isdn30 and ported the numbers to telewest
several years ago.
Now, the client moved to adept telecom. I *think* adept resells BT
products. We got new numbers from adept (bt?) and the old pbx on the
telewest lines forwards the
I have a couple of useful bits that could be tacked on to this..
1. Telcos required to offer the ability to set the outbound caller id.
2. Telcos required to offer the ability to write to the CNAM database, in
near-real or short time.
3. Telcos required to forward the ANI you provide to the 911
Barry Fawthrop wrote:
Thanks Andrew
I have no plans to VoIP my Faxes to a VoIP provider
I just would like to send them from my desktop (which is windows) to
my PBX (which is AstLinux inside a net 4801)
The PBX connects to PSTN lines via a FXO Gateway (CG-410 in my case)
So really it's
On Thu, 2006-10-26 at 15:40 +0100, Tim Panton wrote:
On 26 Oct 2006, at 11:59, Conrad Wood wrote:
A client used to use BT isdn30 and ported the numbers to telewest
several years ago.
Now, the client moved to adept telecom. I *think* adept resells BT
products. We got new numbers from
with SIP qualify, I can specify, what time in delay I will accept,
with sip and setting qualify=3000 I can circumvent this anoying messages
(bacause delay in reply is about 2000ms, and I accept 3000ms)
with iax, qualify is working different, so setting qualify=3000 will
ping peer every 3s,
Michiel van Baak ([EMAIL PROTECTED]) wrote:
On 09:51, Wed 25 Oct 06, Alex wrote:
Any plans to support multiple virtual pbx-en on one asterisk
instance ?
That's something almost no webbased tool implements. It's
all one asterisk, one pbx while asterisk is very capable
of virtualhosting
Hi Michael, do you have any new information about sqlite and asterisk
in realtime?
what release of asterisk are you using?
On 8/7/06, Michael Iedema [EMAIL PROTECTED] wrote:
Greetings,
I'm trying to replace my extensions.conf with a sqlite database. So
far everything's gone really rocky to be
In article [EMAIL PROTECTED],
Douglas Garstang [EMAIL PROTECTED] wrote:
I'm not having much luck here. I used the default zaptel.conf and zapata.conf
files, and put
a load = chan_zap.so in my modules.conf. On load, asterisk reports:
[chan_zap.so] = (Zapata Telephony w/PRI)
Oct 26
On Thu, 2006-10-26 at 17:43 +0200, Pavel Jezek wrote:
with SIP qualify, I can specify, what time in delay I will accept,
with sip and setting qualify=3000 I can circumvent this anoying messages
(bacause delay in reply is about 2000ms, and I accept 3000ms)
with iax, qualify is working
Can I safely assume that SQLite can be used to code something for
Asterisk Realtime instead of the much used mysql database?
I have read several old posts, but nothing point me to an answer.
maybe ARA--odbc--sqlite or ARA--sqlite?
--
-Original Message-
From: Dave Cotton [mailto:[EMAIL PROTECTED]
Sent: Thursday, October 26, 2006 10:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP v IAX2
On Thu, 2006-10-26 at 17:43 +0200, Pavel Jezek wrote:
with SIP qualify,
Obviously we (as an industry) have to start to take notice of this spoofing.
otherwise big brother will start to legistrate against it. This will
give the CRTC or FCC another excuse to spend a lot of tax payers money on
something which is of marginal value.
My position is that there are only two
I'm seeing an interesting problem in asterisk:asterisk has domain a.com and the sip proxy has domain b.com.The sip proxy is configured as a friend in sip.conf.
If a call comes in to asterisk from the sip proxy, if ${EXTEN} exists in the sippeers table the call goes to the default context else the
As I understand it the main advantege IAX has over SIP is the number of
port it uses and therefore its ability to traverse router/switches and
firewalls
Also the higher number of simulatanious SIP calls travelling through these
devices adds a higher overhead than IAX with it's single port.
On Thu, 2006-10-26 at 13:14 -0400, Henry.L.Coleman wrote:
As I understand it the main advantege IAX has over SIP is the number of
port it uses and therefore its ability to traverse router/switches and
firewalls
Also the higher number of simulatanious SIP calls travelling through these
devices
I am unable to get any softphone to register to my asterisk server
when I am connected via VPN. I have tried Ekiga, LinPhone, and
Twinkle... on multiple machines. It works fine when locally connected
(same subnet). The VPN is not NAT'ing anything... and all other
connections work fine across
On Thursday 26 October 2006 13:14, Henry.L.Coleman wrote:
As I understand it the main advantege IAX has over SIP is the number of
port it uses and therefore its ability to traverse router/switches and
firewalls
Yes.
Also the higher number of simulatanious SIP calls travelling through these
On 14:28, Thu 26 Oct 06, Tzafrir Cohen wrote:
On Thu, Oct 26, 2006 at 10:48:52AM +0200, Michiel van Baak wrote:
the chan_capi is merged with BRIStuff. This means you no
longer have to download and compile chan_capi manually when
you want to use a CAPI board with bristuffed asterisk.
Hey everyone,
I was frustrated with the existing app_cepstral/app_swift TTS modules I've
found on the net, so I hacked up my own. It's been working really well for
me so I thought I'd share.
In developing this, I wanted to avoid:
* the startup delay incurred writing TTS output to a temp file
Does anyone have experience with this product?
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Hi,
i am from Germany, so excuse my School English.
I use Asteriks 1.2.12.1, zaptel 1.2.9 and mISDN Rc25 - since my update
of Asterisk 2 wooks ago, Echos accure in my SIP Calls.
I use SNOM 360, sometimes there is no echo (for example if i call myself
via
I suspect that IAX has less overhead but when we get into voice bandwidth
then the answer gets very complex for any given codec. Andrew mentions SIP
concurrency but I doubt that this buys very much. In reality, in a
single processor world everything gets processes serially.
For *2* IAX would be my
On Thu, Oct 26, 2006 at 01:00:18PM -0400, Henry.L.Coleman wrote:
Obviously we (as an industry) have to start to take notice of this spoofing.
otherwise big brother will start to legistrate against it. This will
give the CRTC or FCC another excuse to spend a lot of tax payers money on
something
This was orignally posted on The Asterisk Blog Forums. See the original post here.Pete101 says: I am having issues with all inbound calls coming
into the system. It is taking like 10 seconds for it to decide where to
route the call. It applies for both PSTN calls and VoiP calls. Does
anyone have
26 okt 2006 kl. 18.57 skrev Douglas Garstang:
-Original Message-
From: Dave Cotton [mailto:[EMAIL PROTECTED]
Sent: Thursday, October 26, 2006 10:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP v IAX2
On Thu, 2006-10-26 at 17:43
I am working around this by having the front desk use the ## transfer.
I am dealing with tech support on the SIP device. (a 24 port SIP to digital
handset converter)
I am not sure which is at fault, asterisk or the SIP device.
--
--
Steven
http://www.glimasoutheast.org
Steven [EMAIL
By the OP reference to wav or gsm, I assume he is talking about the VM
recordings.
Sorry, I do not have the answer though.
--
--
Steven
http://www.glimasoutheast.org
Conrad Wood [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED]
On Wed, 2006-10-25 at 11:24 +0200, Jon Schøpzinsky
On 2006-10-26 09:21:20 -0700, Dave Cotton [EMAIL PROTECTED] said:
On Thu, 2006-10-26 at 17:43 +0200, Pavel Jezek wrote:
with SIP qualify, I can specify, what time in delay I will accept,
with sip and setting qualify=3000 I can circumvent this anoying
messages (bacause delay in reply is about
-Original Message-
From: Olle E Johansson [mailto:[EMAIL PROTECTED]
Sent: Thursday, October 26, 2006 1:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP v IAX2
26 okt 2006 kl. 18.57 skrev Douglas Garstang:
-Original
On 2006-10-26 03:18:20 -0700, Stefan Agethen
[EMAIL PROTECTED] said:
Hi,
i am from Germany, so excuse my School English.
I use Asteriks 1.2.12.1, zaptel 1.2.9 and mISDN Rc25 - since my update
of Asterisk 2 wooks ago, Echos accure in my SIP Calls.
I use SNOM 360, sometimes there is no echo
AFAIK, you will need to do the first. ARA-odbc-sqlite
On 10/26/06, Erick Perez [EMAIL PROTECTED] wrote:
Can I safely assume that SQLite can be used to code something for
Asterisk Realtime instead of the much used mysql database?
I have read several old posts, but nothing point me to an answer.
On Thu, 26 Oct 2006, Michiel van Baak wrote:
On 14:28, Thu 26 Oct 06, Tzafrir Cohen wrote:
On Thu, Oct 26, 2006 at 10:48:52AM +0200, Michiel van Baak wrote:
the chan_capi is merged with BRIStuff. This means you no
longer have to download and compile chan_capi manually when
you want
I just wanted to ask a general question to anyone that
serves as a service provider on the list out there. Are you using OpenSER and
Asterisk for your high availability and redundancy or DUNDI? Anyone have anything
to say as to which would be better for a service provider and why?
On Mon, 23 Oct 2006, Klaus Darilion wrote:
Hi!
This weekend we had a problem with our Asterisk Box which ran flawlessly for
nearly 4 weeks. The Asterisk server sits between the PSTN and a Siemens PBX
and bridges 2 BRI lines. No calls, not incoming, not outgoing. The admin
rebooted the Dell
Hi,
I have to set up an Asterisk with an 4-port BRI card.
Hylafax should send and receive fax.
Will this work reliable?
Any recommandations for an 4-port BRI card?
Other alternatives except analog fax units?
thanks for your help
best regards
Thomas
On Thu, 26 Oct 2006, Thomas Winter wrote:
Hi,
I have to set up an Asterisk with an 4-port BRI card.
Hylafax should send and receive fax.
Will this work reliable?
Any recommandations for an 4-port BRI card?
Other alternatives except analog fax units?
I would recommend the Eicon DIVA
Thomas Winter wrote:
I have to set up an Asterisk with an 4-port BRI card.
Hylafax should send and receive fax.
Will this work reliable?
If the BRI channel driver works correctly, yes, you should be fine to
use iaxmodem.
Lee.
___
--Bandwidth
On 27/10/2006, at 7:22 AM, Thomas Winter wrote:
I have to set up an Asterisk with an 4-port BRI card.
Hylafax should send and receive fax.
Will this work reliable?
I have a Eicon V-4BRI (which is in fact a voice-only board) that does
faxing via HylaFax/IAXmodem and its flawless. However,
On 23:11, Thu 26 Oct 06, Armin Schindler wrote:
snip/snip
chan-capi-cm (chan-capi.org) is a complete rewritten version of chan-capi
with more features and as far as I can tell, much more stable.
You do faxing with chan-capi 0.3.5? But this isn't faxing via CAPI, right?
As far as I know,
Am Thursday 26 October 2006 23:35 schrieben Sie:
On Thu, 26 Oct 2006, Thomas Winter wrote:
I would recommend the Eicon DIVA Server 4BRI cards. They have a
capi interface which is used by chan-capi (chan-capi.org) and
onboards DSPs for the faxing.
You can use this for send and receive faxes
Does SipAddHeader only allow headers to be added to INVITEs, or should it
also allow headers to be added BYEs or SIP responses as well?
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Echo is generated by the analog end to where you place the call, not the IP side of it.
As far as I know the echo cancelation in the Asterisk can only be tweaked in the zapata.conf (since IP calls don't generate it)
I'm afraid there is little you can do to here.Alyed
I've worked with it using Asterisk, and worked really fine
Michael Welter wrote:
Does anyone have experience with this product?
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Moises Silva wrote:
AFAIK, you will need to do the first. ARA-odbc-sqlite
res_sqlite3 in asterisk-addons supports ARA
On 10/26/06, Erick Perez [EMAIL PROTECTED] wrote:
Can I safely assume that SQLite can be used to code something for
Asterisk Realtime instead of the much used mysql database?
Hello All,
To those who have (sorry) deployed Cisco 7960s in a
call center environment, I have a question.
A group of phones (13 of them, all identically configured
except for extensions) is part of a ring group. Phones have 5 lines configured
for the same extension, one line for
@0 PPS may not work
Check the Wiki first to solve the debounce problem, then recompile.
there are also references in the Wiki to make break ratios
Some of us who use old rotary phones have difficulty with 10 pps.
Seems the Zaptel authors didn't completely do their homework with pulse dial
Thanks for your suggestion. I have compiled according to
http://www.voip-info.org/wiki/index.php?page=Asterisk+zaptel+pulse
+dialing
dialing at 10 pps works fine with Asterisk with the newly compiled
wctdm. but when I dial at 20 pps, the pulses cannot be decoded
correctly.
I tried changing the
For anyone interested the problem was we
needed to add a bindaddr= for the IP address of the cluster (virtual IP).
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Klaverstyn, David C
Sent: Thursday, 26 October 2006
4:43 PM
To:
asterisk-users@lists.digium.com
This is
what I would like to do...
exten =7299,1,Voicemail(u8896)
exten
=7299,2,Dial(zap/g1/#641299)
exten =7299,3,Hangup
WhatI would expect to happen
is...
Incoming call is answered by
voicemail...
Voicemail app finishes and the next priority
starts.
This is where the problem lies...
In the VM settings you can set an email to be sent
to you witha copy of the voicemail in wav format or you can have a message
sent to a cell phone or pager telling you that you have a VM. You can also use
FOP (a web GUI that was created in flash) to see who has Vm.
As far as where the call
I am surprised that you are getting echo on SIP calls. You can get echo
in two scenarios on SIP calls.
1. If SIP calls are crossing to PSTN (inbound/outbound). Here you need
to enable echo canceller and AGGRESSIVE if needed in zconfig.h.
2. Second source of echo on SIP calls could be ACOUSTIC.
You will also perceive jitter as echo
If any links are getting busy and routers or switches have to buffer
you will hear what sounds like echo, not to mention if you have a
high packet loss also
Of course jitter would have to be above 100ms or so to be noticeable
as far as acoustic echo,
I saw this problem before... to solve that, I needed to hack asterisk
to remove a header SIP field.
Check your ACT phone log, and you can figure out which filed is that.
Then, comment that filed from your chan_sip.c and recompile asterisk..
and that's it.. it only happens with ACT phones.
I
Not too sure, if this msg did reach the group, so resending.
---BeginMessage---
I know, I know, the wiki link for that one.
But wot I wanted were actual figures related to Asterisk n QoS.
How does Asterisk actualy handle and fare at the following QoS issues :
1) Delay
2) Jitter
3) Packet
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