Using asterisk 11.16.0 I am unable to retrieve any voicemail with a video
attachment while using any video phone. This does work in my 1.8.23.1
installation. The file is skipped with the ast_streamfile error (and moved to
OLD), and the prompts following that message display the ast_best_codec
Can anyone shed any light on an issue with agent penalties?
I have 2 queues set up with agents working both queues, but where agent
1 should have a penalty for queue 2 and agent 2 should have a penalty
for queue 1. When a call is sent to either queue, it rings agents with
and without penalties
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Steve Dolloff
Sent: Tuesday, April 26, 2005 8:43 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] return a value from dial macro
Does anyone know of a way to pass
Does anyone know of a way to pass a value back to the dial plan after
calling a macro from the dial app in the 1.0 release? I think this
should be pretty simple, but I can't quite figure out how.
The example would work except that the modified value of found is not
usable when Dial ends. I
We use a product from oriontelecom.com.
The interface is rough, but we have not had a single problem since putting this
in.
Stephen Dolloff
DLS Internet Services
847-854-4799 x256
[EMAIL PROTECTED]
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
Does the PAP-NA2 work with the Sipura firmware and tftp provisioning
options?
Stephen
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Eric Merkel
Sent: Wednesday, September 22, 2004 9:07 AM
To: [EMAIL PROTECTED]
Subject:
I have a similar issue with Sipura using compact headers, but not with
regular headers. I am working on reproducing with the latest CVS.
Maybe you are using compact SIP headers on your ATA186?
http://bugs.digium.com/bug_view_page.php?bug_id=0001843
Stephen
-Original Message-
From:
I would like to move voicemail to a dedicated server but I can't figure
out how to make the MWI work if the ATA doesn't register to the
voicemail server. The main reason for this is redundancy. I have two
SIP registrars running and in the case of a failure from the primary,
both the gateways and
Has anyone seen this before? This channel is consistently present on
both of my asterisk servers. Sometimes they disappear for a few seconds
and then come back. It always has the same Call ID.
voip1*CLI sip show channels
Peer User/ANRCall ID Seq (Tx/Rx) Lag Jitter
: RE: [Asterisk-Users] Mystery SIP channels
What address is that? Is it a phone (or address of a PC with a
softphone?)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Steve Dolloff
Sent: Thursday, May 20, 2004 10:41 AM
To: [EMAIL PROTECTED
Get an RMA. I've had a few that did that as well.
Stephen Dolloff
DLS Internet Services
847-854-4799 x256
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Christopher J. Wolff
Sent: Thursday, April 01, 2004 5:50 PM
We have the same complaint here. The caller doesn't hear the receiver
say hello and so no-one knows what's going on.
Stephen
-Original Message-
From: James Sizemore [mailto:[EMAIL PROTECTED]
Sent: Thursday, March 11, 2004 9:38 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users]
] E911 support
Steve Dolloff wrote:
I have the following in my sip.conf entries:
callerid=Anonymous 8885551212
This still passes the number for 911, but flags the call as private.
I
believe this will meet your requirements.
Stephen
OK. I was under the impression that the PSAP got
I have the following in my sip.conf entries:
callerid=Anonymous 8885551212
This still passes the number for 911, but flags the call as private. I
believe this will meet your requirements.
Stephen
-Original Message-
From: Olle E. Johansson [mailto:[EMAIL PROTECTED]
Sent: Thursday,
Can anyone else share their memory use experiences?
I am currently running * with about 100 sip.conf entries and 400
dialplans. The memory usage starts at around 10M and goes up every day.
After 5 days, it is currently at 90M.
Stephen
___
http://www.cisco.com/en/US/products/sw/iosswrel/ps1835/products_programm
ing_reference_guide_book09186a0080080221.html
Stephen Dolloff
DLS Internet Services
847-854-4799 x256
[EMAIL PROTECTED]
-Original Message-
From: Rich Adamson [mailto:[EMAIL PROTECTED]
Sent: Wednesday, February
In terms of your dtmf settings, you need to make sure that the 5300 is
configured with the same dtmf-relay method and codec as Asterisk. I am
also trying to do this using SIP ATAs. It works fine for most calls,
but certain ones do not. I have been working with Cisco on this and it
appears that
Sipura recommended disabling the echo cancellation on the SPA-2000 for
modem pass-through. It does help although still not 100% success rate.
Stephen
-Original Message-
From: Christopher J. Wolff [mailto:[EMAIL PROTECTED]
Sent: Thursday, January 15, 2004 12:14 PM
To: [EMAIL
Some areas in the US also use 10 or 11 digital dialing for all calls,
whether they are local, long, toll or non-toll.
-Original Message-
From: Eric Wieling [mailto:[EMAIL PROTECTED]
Sent: Friday, January 09, 2004 1:53 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] USA dial
I have 2 questions regarding asterisk logs that I really hope someone
can help me with.
Jan 7 09:40:14 WARNING[1009517568]: File chan_iax.c, Line 3537
(registry_rerequest): Received unsolicited registry authenticate request
from '209.242.15.34'
I get this IAX message every minute or so. I have
I really hope that someone can help me with this one.
DTMF tones are not working for certain places that I call, specifically
1-800-882-8880 which is the AA advantage line. It works for almost
everyplace else. If I bypass asterisk, the call works fine.
Network looks like:
SPA-2000 --SIP--
I am having the same problem, but only with one specific user, so I
believe it is network related.
Anyone that can point me in the specific direction of what would cause
this?
-Original Message-
From: WipeOut [mailto:[EMAIL PROTECTED]
Sent: Monday, January 05, 2004 10:22 AM
To:
I simply have 2 asterisk servers and have the clients point to a DNS SVR
record for their proxy. The DNS record lists the primary and secondary
with preference for the primary. This won't stop calls from being
dropped if the primary goes down if you are routing them through the
server, but it
Not all VoIP providers will have Vonage's 911 issues. It's perfectly
possible for a VoIP provider to provide outbound caller information to
the PSAPs if they spend the time and money to do so.
Stephen
Summary: if you're the only caller, calling only to the US, then you
might be crazy to not
would like to do
something similar.
Thanks,
Seth
- Original Message -
From: Steve Dolloff [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, December 16, 2003 11:15 AM
Subject: RE: [Asterisk-Users] Cisco Gateway Integration
I am using it with the AS5350 via SIP
I am using it with the AS5350 via SIP and it works great. I was also
using the ATA186 with SIP but I am switching to the SPA-2000 for a
better feature set.
Stephen
-Original Message-
From: Bruce Hedreen [mailto:[EMAIL PROTECTED]
Sent: Tuesday, December 16, 2003 8:29 AM
To: [EMAIL
Greetings,
I have a couple of questions and figured I would put them all in one
message to not spam the list as much as possible. I have searched
voip-info, google and the list archives for all of these questions. If
I have missed the correct response, please accept my apologies.
I have been
Are there any known issues with call-waiting caller-id for SIP?
Caller-ID on the first call works fine, but when the second call comes
in, I hear the interrupt tone, but the caller-id doesn't display
anything.
I have tried this with the Cisco ATA and the SPA-2000. I have also
tried two
I constantly get the following error messages in
/var/log/asterisk/messages:
Dec 8 10:52:57 WARNING[1009521664]: File chan_iax.c, Line 3324
(iax_ack_registry): Received unsolicited registry ack from '192.168.0.1'
Dec 8 10:52:57 WARNING[1009521664]: File chan_iax.c, Line 4181
(socket_read):
I would be seriously wary of putting a DS3's worth of voice traffic on a
TNT. I don't believe they are rated to handle that much voice. The
APX1000 would be a much better platform, but I don't know if you can
find one used.
Stephen
-Original Message-
From: Ernest W. Lessenger
I am still having these same problems. Anyone with experience with
these apps that could point me in the right direction?
I am having issues with Privacy Manager and Zapateller.
If I set callerid= on a sip user zapateller sends the tones
If I set callerid=Anonymous 8475551212 zapateller
Anyone have any thoughts on this since last week?
I am having issues with Privacy Manager and Zapateller.
If I set callerid= on a sip user zapateller sends the tones
If I set callerid=Anonymous 8475551212 zapateller doesn't send the
tones
If I call from a phone after dialing *67
Speaking of voice prompts, could anyone tell me why the pre-recorded
prompts sometimes sound garbled, but the voicemail messages themselves
sound fine? Is it the format of the prompts?
Stephen
I would like to release prepaid application.
But I have a small problem, we are using their Cisco
I am having issues with Privacy Manager and Zapateller.
If I set callerid= on a sip user zapateller sends the tones
If I set callerid=Anonymous 8475551212 zapateller doesn't send the
tones
If I call from a phone after dialing *67 zapateller doesn't send the
tones
In the last 2 cases, the display
Sent
209.242.15.34:5036voip1 Unregistered 60
Rejected
-Original Message-
From: Philipp von Klitzing [mailto:[EMAIL PROTECTED]
aachen.de]
Sent: Friday, November 14, 2003 4:08 AM
To: Steve Dolloff
Subject: RE: [Asterisk-Users] IAX trunk monitoring
You might want
I have an issue where * tries to route a call over IAX to another server
even if the server is down. I have included the relevant entries from
my iax.conf, extensions.conf, and some debug output. If someone could
tell me what I have configured incorrectly, I would appreciate it.
Thanks,
I have modified the configuration for dynamic host and registered each
server with the other. The iax show users now lists the other iax
device as registered vs unavailable, but I still don't know how to keep
it from calling if the device becomes unavailable.
I changed the extensions file to:
Can someone point me to the echo cancellation settings for a pure sip
setup?
Thanks,
Stephen
Subject: Re: [Asterisk-Users] ReplayTV connecting through Asterisk box
Has anyone had any luck getting a ReplayTV DVR box to connect
through an Asterisk box? Mine seems to dial just fine, but can't
I am seeing strange behavior that I don't understand. Voicemail2 and
voicemailmain2 work fine if I call from a sip phone directly connected
to *, but if I call either of them from an analog line on the other side
of a sip gateway, voicemail seems to ignore digits. If I am recording a
message and
Can someone please explain what I am doing wrong here? I only want the
extensions listed in long-users to be able to access the longdistance
context.
If I do this, I get a congestion tone no matter what I dial. If I add a
[default] context and include = longdistance, then the local callers
can
I am looking for a SIP carrier to handle wholesale residential traffic.
Standard LEC services in the US. Anyone have any suggestions?
Thanks,
Stephen
Stephen Dolloff
DLS Internet Services
847-854-4799 x256
___
Asterisk-Users mailing list
[EMAIL
Here is my Configuration
PSTN - Cisco AS5350 - SIP - ASTERISK - SIP - ATA186
When I call from the pstn to the ATA, the ATA rings but I don't hear
anything on the calling side until the call is picked up.
When I call from the ATA, everything seems to work fine.
When I bypassed ASTERISK,
, this does NOT solve the underlying problem.
On Thu, 2003-10-09 at 15:29, Steve Dolloff wrote:
Here is my Configuration
PSTN - Cisco AS5350 - SIP - ASTERISK - SIP - ATA186
When I call from the pstn to the ATA, the ATA rings but I don't hear
anything on the calling side until the call is picked up
I am looking for examples or instructions on how to route calls to
voicemailmain based on remote-party-id.
I have the following entry in my extensions.conf file:
exten = 200,1,Voicemailmain(${CALLERIDNUM})
I am routing calls to * via SER and sending Remote-Party-ID in the SIP
headers. I am
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