[asterisk-users] asterisk-stat v.2 and mysql 5.1.51

After upgrade to mysql 5.1.51 my asterisk-stat-v2 is not displaying correctly. Does anybody have a similar problem? Is it due to mysql-5.1.51 or the problem is with new glibc-2.11.2 ? -- Joseph -- _ -- Bandwidth and Colocatio

Re: [asterisk-users] Sending calls to a particular T1 port.

Don't forget to setup a d-chan for that group since it's not part of the 1st one. > -Original Message- > From: maill...@lightspeed.ca > Sent: Fri, 12 Nov 2010 12:44:39 -0800 > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] Sending calls to a particular T1 port. > > O

[asterisk-users] Asterisk and Tandberg Gatekeeper

Has anyone had any luck getting Asterisk 1.6.2.13 to register to a Tandberg Gatekeeper? The logs on the Asterisk end seem to show that the registration request is sent, and the Tandberg Gatekeeper responds. However, the response doesn't seem to be what Asterisk was expecting. Here is my ooh323.c

[asterisk-users] Scheduled maintenance for various Asterisk community services

Between 10:00AM and 1:00PM CST on Saturday, November 13, the services below will experience extended outages as the servers that host them are upgraded and reconfigured: downloads.digium.com downloads.asterisk.org bamboo.asterisk.org packages.asterisk.org svn.digium.com svn.asterisk.org issues.a

Re: [asterisk-users] "scratchy" sound on TE410P

On 11/12/10 10:44 AM, Jeff LaCoursiere wrote: > > > On Thu, 11 Nov 2010, Russ Meyerriecks wrote: > >>> >>> Yes, this is a snapshot after about 24 hours since I cleared the counters. >>> I see what you mean - how can I have 76 seconds of errors but no bumped >>> error counters. I ran again just now

Re: [asterisk-users] Call failed becaus of SIP tanslate

Ok, it seems like I don't have g729 codec intsalled, can I install this codec within asterisk 1.2 or just 1.4 and 1.6 are supported? On Fri, Nov 12, 2010 at 2:56 PM, khalid touati wrote: > Hi Guys, > I have a the following issue when I ma trying to place a call through my > voip provider, I am u

Re: [asterisk-users] Sending calls to a particular T1 port.

Oh, this is most excellent. Although it means that my google-fu has failed me. ;) > On Fri, Nov 12, 2010 at 1:42 PM, Jonathan Thurman > wrote: > I didn't read the whole thing, but it looks pretty OK at a glance. > > http://www.cadvision.com/blanchas/Asterisk/DahdiT1trunk.html > > I hope that helps

Re: [asterisk-users] Official Documentation for Asterisk 1.6 Realtime ODBC Tables

On Fri, Nov 12, 2010 at 1:46 PM, Brett Woollum wrote: > Yeah a production system that crashes is not fun..  I hear you there. > > Maybe the solution will be to design some sort of method for each asterisk > server to auto prune and load as necessary. The first issue that is coming to > mind is t

[asterisk-users] Call failed becaus of SIP tanslate

Hi Guys, I have a the following issue when I ma trying to place a call through my voip provider, I am using an asterisk 1.2.21.1, do you have an idea what could fix this issue (as you can see when the other party answered, the call get dropped because of probably sip incompatibility) Nov 12 14:31:

Re: [asterisk-users] Official Documentation for Asterisk 1.6 Realtime ODBC Tables

Yeah a production system that crashes is not fun.. I hear you there. Maybe the solution will be to design some sort of method for each asterisk server to auto prune and load as necessary. The first issue that is coming to mind is that I'm doing configuration and db changes/updates on a differe

Re: [asterisk-users] Official Documentation for Asterisk 1.6 Realtime ODBC Tables

Inline response :D On Fri, Nov 12, 2010 at 12:54 PM, Brett Woollum wrote: > Hi Sherwood, > > Thanks for the reply. Most definitely mate, since I've used realtime so much, I enjoy digging in there. However, I use the MySQL realtime architecture, so forgive me if we find there's differences betwee

Re: [asterisk-users] Official Documentation for Asterisk 1.6 Realtime ODBC Tables

Hi Sherwood, Thanks for the reply. That's interesting to me. What is the point of rtcachefriends = no if it causes weird things like this to happen? As mentioned, I'd like to stay real-time and fully database driven for everything. Not only does it make life easier in terms of changing settin

Re: [asterisk-users] Sending calls to a particular T1 port.

On Fri, Nov 12, 2010 at 1:42 PM, Jonathan Thurman wrote: > On Fri, Nov 12, 2010 at 10:17 AM, Ernie Dunbar > wrote: > > that goes from port 4 on the live server to port 1 on the backup server. > > > In /etc/asterisk/chan_dahdi.conf: > > > > group=4 > > context=local > > switchtype = national > > s

Re: [asterisk-users] Sending calls to a particular T1 port.

On Fri, Nov 12, 2010 at 1:17 PM, Ernie Dunbar wrote: > We have two Asterisk servers. One is a live server supporting our > customers, and the other is a backup server that's being upgraded and > pressed into service. Both servers have a Digium TE405P T1 card in them, > and in order to test the T1

Re: [asterisk-users] Sending calls to a particular T1 port.

On Fri, Nov 12, 2010 at 10:17 AM, Ernie Dunbar wrote: > that goes from port 4 on the live server to port 1 on the backup server. > In /etc/asterisk/chan_dahdi.conf: > > group=4 > context=local > switchtype = national > signalling = pri_cpe > channel => 73-95 > context = default > group = 63 What

Re: [asterisk-users] Context issue

El 12/11/10 12:13, Adrian Marsh escribió: How odd... If I specify the host=dynamic then it goes to the wrong context. If I specify the host=192.168.50.132, then it goes to the correct context. If I don't specify the host at all, then it also goes to the correct context... (but then of cours

[asterisk-users] Sending calls to a particular T1 port.

We have two Asterisk servers. One is a live server supporting our customers, and the other is a backup server that's being upgraded and pressed into service. Both servers have a Digium TE405P T1 card in them, and in order to test the T1 service on the backup server, I've created a T1 crossover cabl

Re: [asterisk-users] Context issue

How odd... If I specify the host=dynamic then it goes to the wrong context. If I specify the host=192.168.50.132, then it goes to the correct context. If I don't specify the host at all, then it also goes to the correct context... (but then of course I can't use that account for outbound cal

[asterisk-users] Context issue

Hi, Running 1.4.15. I've a SIP user as below. My default context in sip.conf is [incomming_pstn] I'm having trouble with inbound calls going to the wrong context. [test-ubi] username=test-ubi type=friend secret=XXX host=dynamic canreinvite=no context=testinbound nat=yes a

[asterisk-users] Asterisk Sip trunking routing problem

  Hi   This problem is driving me crazy. I have two severs which are trunked by the sip   Asterisk box A is natted thus in my sip.conf file I use the following externip=DDNS address   Asterisk box B is NOT natted and has a static IP. Asterisk box A and B are both registered with each other.   B ho

Re: [asterisk-users] "scratchy" sound on TE410P

On Thu, 11 Nov 2010, Russ Meyerriecks wrote: >> >> Yes, this is a snapshot after about 24 hours since I cleared the counters. >> I see what you mean - how can I have 76 seconds of errors but no bumped >> error counters. I ran again just now: >> >> r...@vigw3:/etc/asterisk# dahdi_maint -s 1 >> S

Re: [asterisk-users] "scratchy" sound on TE410P

On 11/11/10 11:06 PM, Carlos Chavez wrote: > On Thu, 11 Nov 2010 20:08:09 -0600, Russ Meyerriecks wrote >> On 11/11/10 7:23 PM, Carlos Chavez wrote: >> >>> I seem to be having the same problem with a new server. I am using a >>> TE220 with a VPM450 module. Using Dahdi 2.4.0 and Asterisk 1.6.2

Re: [asterisk-users] Asterisk 1.6.20 / CDR issue with app-dial / bug or feature?

Sorry for top-post, but takes longer to un-work when both ends are outlook. Did sox on goodbye and hello to get lengths. sox goodbye.gsm goodbye.wav stat Samples read: 7360 Length (seconds): 0.92 Scaled by: 2147483647.0 Maximum amplitude: 0.332275 Minimum am

Re: [asterisk-users] Asterisk 1.6.20 / CDR issue with app-dial / bug or feature?

Exactly! The call duration is not correct in this case. That is "my" problem. Am 12.11.2010 15:23, schrieb Danny Nicholas: From: as

Re: [asterisk-users] Asterisk 1.6.20 / CDR issue with app-dial / bug or feature?

1.6.1.20 :-) Am 12.11.2010 15:12, schrieb Olivier: 2010/11/12 Thorsten Göllner Hi, it's me again with a cdr-issue. I have the following example extensions.conf: # dial

Re: [asterisk-users] Official Documentation for Asterisk 1.6 Realtime ODBC Tables

On Fri, Nov 12, 2010 at 9:36 AM, Sherwood McGowan wrote: > On Fri, Nov 12, 2010 at 7:52 AM, Brett Woollum wrote: >> More information:  When I have "rtcachefriends = yes" in sip.conf, >> everything seems fine. With "rtcachefriends = no" I see this behavior. >> >> I'd rather not cache. I'm aiming f

Re: [asterisk-users] Official Documentation for Asterisk 1.6 Realtime ODBC Tables

On Fri, Nov 12, 2010 at 7:52 AM, Brett Woollum wrote: > More information:  When I have "rtcachefriends = yes" in sip.conf, > everything seems fine. With "rtcachefriends = no" I see this behavior. > > I'd rather not cache. I'm aiming for as near real-time as possible. > > Any thoughts? > > Brett Wo

Re: [asterisk-users] Asterisk 1.6.20 / CDR issue with app-dial / bug or feature?

_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: Friday, November 12, 2010 8:12 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 1.6.20 / CDR issue with app-dial

Re: [asterisk-users] Asterisk 1.6.20 / CDR issue with app-dial / bug or feature?

2010/11/12 Thorsten Göllner > Hi, > > it's me again with a cdr-issue. I have the following example > extensions.conf: > > # dial in > exten => 100,1,Playback(hello) > exten => 100,n,Dial(local/200,20,rtg) > exten => 100,n,Playback(pleasewait) > exten => 100,n,wait(10) > exten => 100,n,Playback(go

Re: [asterisk-users] Official Documentation for Asterisk 1.6 Realtime ODBC Tables

More information: When I have "rtcachefriends = yes" in sip.conf, everything seems fine. With "rtcachefriends = no" I see this behavior. I'd rather not cache. I'm aiming for as near real-time as possible. Any thoughts? Brett Woollum br...@woollum.com - Original Message - From:

Re: [asterisk-users] Official Documentation for Asterisk 1.6 Realtime ODBC Tables

Hi Brad, I did notice that bug in the bug tracker. That's different from the behavior I am seeing. I don't get multiple values in the "Mailbox". I just upgraded to 1.6.2.14 and it's still there. By the way, the quantity of SIP NOTIFY's generated is significant. It appears to be way more that

Re: [asterisk-users] Official Documentation for Asterisk 1.6 Realtime ODBC Tables

Hi Paul, 1.6.2.13. I'll go ahead and update to 1.6.2.14 and see how that works. I did see a couple bugs in the bug tracker for this, but they were resolved a while ago (I want to say 1.6.1 timeframe...). There was also a post on this list about the problem arising from LDAP integration, but I

Re: [asterisk-users] Official Documentation for Asterisk 1.6 Realtime ODBC Tables

>-Original Message- >From: asterisk-users-boun...@lists.digium.com >[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of >Paul Belanger >Sent: Friday, November 12, 2010 7:58 AM >To: Asterisk Users Mailing List - Non-Commercial Discussion >Subject: Re: [asterisk-users] Official

Re: [asterisk-users] Official Documentation for Asterisk 1.6 Realtime ODBC Tables

On Fri, Nov 12, 2010 at 6:07 AM, Brett Woollum wrote: > I'm having an issue where Asterisk continuously sends out a GAZILLION "SIP > NOTIFY" messages when a user has a voice message in their INBOX. This issue > is only present when my SIP users and peers are configured from my ODBC > backend (MySQ

[asterisk-users] Official Documentation for Asterisk 1.6 Realtime ODBC Tables

Hi All, I'm having an issue where Asterisk continuously sends out a GAZILLION "SIP NOTIFY" messages when a user has a voice message in their INBOX. This issue is only present when my SIP users and peers are configured from my ODBC backend (MySQL). A static configuration of users in sip.conf re

Re: [asterisk-users] Asterisk Playback sound dropping on linphone

Hi On 11/11/2010 03:35 PM, Matteo Fortini wrote: > Hi, > I dial on A* from a linphonec to a Playback() extension, then suddenly > the sound stops after a while, without any notice. > I enabled debug both in linphone and A*, and the RTP packets are sent > from A* and received from linphone. It does

[asterisk-users] Asterisk 1.6.20 / CDR issue with app-dial / bug or feature?

Hi, it's me again with a cdr-issue. I have the following example extensions.conf: # dial in exten => 100,1,Playback(hello) exten => 100,n,Dial(local/200,20,rtg) exten => 100,n,Playback(pleasewait) exten => 100,n,wait(10) exten => 100,n,Playback(goodbye) exten => 100,n,Hangup # for local dial ex

Re: [asterisk-users] Random call drops on IAX2

anybody? On 11/10/2010 06:51 PM, Sebastian wrote: > Hello list, > > I have an Asterisk setup with the following details: > > 1. 3 x internal extensions / sip hardphones - Grandstream 2000 > 2. 2 x internal extensions / dahdi cordless phone > 3. 1 x 2 FSX ports OpenVOX pci card > 4. 1 x internal si