Re: [asterisk-users] Solution to connect an audio system to MeetMe

Hello, Le 16/01/2014 14:20, Darryl Moore a écrit : Yup. That's what i do. The CLI version of linphone set to autoanswer, with the audio jacks tied to our exernal sound system. Works well. The echo cancellation in linphone helps a lot for speakerphones. Indeed, works well :-) Thanks. On J

Re: [asterisk-users] Meetme Show Activity in Minus

Solved On Wed, Jan 22, 2014 at 12:44 PM, Chandrakant Solanki < solanki.chandrak...@gmail.com> wrote: > Hello All, > > Asterisk: 1.8.13.0 > Dahdi : 2.6.2 > Kernel : 2.6.32-431.3.1.el6.i686 #1 SMP Fri Jan 3 18:53:30 UTC 2014 i686 > i686 i386 GNU/Linux > OS : CentOS 6.4 > > When I show meetme roo

[asterisk-users] Meetme Show Activity in Minus

Hello All, Asterisk: 1.8.13.0 Dahdi : 2.6.2 Kernel : 2.6.32-431.3.1.el6.i686 #1 SMP Fri Jan 3 18:53:30 UTC 2014 i686 i686 i386 GNU/Linux OS : CentOS 6.4 When I show meetme room details using "meetme list" command it shows Minus in activity column. Any Idea. >meetme list Conf Num Parties

[asterisk-users] Register => plain text password

Hello, Is there anyway to encrypt or scramble a bit the secret used to register with a provider? Im talking about the register => fromuser@fromdomain:secret@host directive in sip.conf This clever dude modified the code back in 1.4:

Re: [asterisk-users] qualify=yes & outboundproxy

Are you absolutely sure you need to use the outboundproxy setting rather than using a peer/friend? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Lemberger Sent: Tuesday, January 21, 2014 7:53 PM To: ast

[asterisk-users] qualify=yes & outboundproxy

I'm having some trouble turning with trunk monitoring while using an outbound proxy. While all other sip messaging (e.g. calls) respects the outboundproxy setting, Options packets from setting qualify=yes do not. Asterisk tried to send the Options message directly to the "host=" IP, instead of th

Re: [asterisk-users] Asterisk not receiving call from VPN address

On 1/21/14, 4:38 PM, David Cunningham wrote: Hi Andres, Thanks for the idea. We did send bindaddr to the VPN address and restarted Asterisk, but unfortunately that didn't solve the issue. Asterisk didn't complain, but still the "sip set debug on" didn't show the packets. Ok, that is progres

Re: [asterisk-users] Asterisk Fax detection *11.7

Sorry, I missed the line showing the call had been answered. On 22/01/2014 8:11 AM, Larry Moore wrote: Hello, Perhaps you need to have directmedia=no set for the channel, the call doesn't appear to have been answered hence asterisk won't be able to hear any tones to determine for itself if the

Re: [asterisk-users] Asterisk Fax detection *11.7

Hello, Perhaps you need to have directmedia=no set for the channel, the call doesn't appear to have been answered hence asterisk won't be able to hear any tones to determine for itself if the call is an incoming fax. Larry. On 21/01/2014 6:51 PM, Jakob-Matthias Böttger wrote: Hello everybod

Re: [asterisk-users] Asterisk not receiving call from VPN address

> > > At this point in time, you'll need to show us a .pcap on the Asterisk > box, when you make a call to it via Kamailio. > > -- > Paul Belanger | PolyBeacon, Inc. > Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) > Github: https://github.com/pabelanger | Twitter: > https://twit

Re: [asterisk-users] Asterisk Fax detection *11.7

On Tue, Jan 21, 2014 at 5:51 AM, Jakob-Matthias Böttger wrote: > Hello everybody > > I'm trying to enable the Digium res_fax app at my *11.7 Server. > > a fax show stats comes up with > FAX Statistics: > --- > > Current Sessions : 0 > Reserved Sessions: 0 > Transmit Attempts

Re: [asterisk-users] Asterisk not receiving call from VPN address

On Tue, Jan 21, 2014 at 5:18 PM, David Cunningham wrote: > On 22 January 2014 09:11, Steve Edwards wrote: >> >> (Please don't top-post.) >> >> >> On Wed, 22 Jan 2014, David Cunningham wrote: >> >>> We did send bindaddr to the VPN address and restarted Asterisk, but >>> unfortunately that didn't s

Re: [asterisk-users] core show channels truncates channel names?

On Tue, Jan 21, 2014 at 3:39 PM, Michelle Dupuis wrote: > When I issue a 'core show channels' command I notice that long usernames > (and channel number) are truncated. For example, if the username is > FONEMITEL1234567890 for a trunk, then it will show > > SIP > Privilege: Command > Channel

[asterisk-users] core show channels truncates channel names?

When I issue a 'core show channels' command I notice that long usernames (and channel number) are truncated. For example, if the username is FONEMITEL1234567890 for a trunk, then it will show SIP Privilege: Command Channel Location State Application(Data)" IAX2/FONEMI

[asterisk-users] AMI version to Asterisk version mapping

Is there a mapping of AMI versions to Asterisk versions? eg: AMI 1.0 = Ast 1.4 AMI 1.1 = Ast 1.6 etc... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory

Re: [asterisk-users] Asterisk not receiving call from VPN address

On 22 January 2014 09:11, Steve Edwards wrote: > (Please don't top-post.) > > > On Wed, 22 Jan 2014, David Cunningham wrote: > > We did send bindaddr to the VPN address and restarted Asterisk, but >> unfortunately that didn't solve the issue. Asterisk didn't complain, but >> still the "sip set d

Re: [asterisk-users] Asterisk not receiving call from VPN address

(Please don't top-post.) On Wed, 22 Jan 2014, David Cunningham wrote: We did send bindaddr to the VPN address and restarted Asterisk, but unfortunately that didn't solve the issue. Asterisk didn't complain, but still the "sip set debug on" didn't show the packets. Have you confirmed via 'net

Re: [asterisk-users] Asterisk not receiving call from VPN address

Hi Paul, Thanks, we did try restarting Asterisk after the VPN was up but that didn't solve the issue either. On 22 January 2014 02:55, Paul Belanger wrote: > On Tue, Jan 21, 2014 at 10:47 AM, Paul Belanger > wrote: > > On Tue, Jan 21, 2014 at 12:40 AM, David Cunningham > > wrote: > >> Hi Pau

Re: [asterisk-users] Asterisk not receiving call from VPN address

Hi Andres, Thanks for the idea. We did send bindaddr to the VPN address and restarted Asterisk, but unfortunately that didn't solve the issue. Asterisk didn't complain, but still the "sip set debug on" didn't show the packets. On 22 January 2014 01:40, Andres wrote: > David, > > It seems to m

Re: [asterisk-users] how to provision asterisk's phonebook to Polycom vVX310's

On Tuesday, January 21, 2014 12:32 PM, Adrian Serafini wrote: > On 01/21/2014 01:55 PM, Stanley van Dijk wrote: >> Hi, >> Am running a freepbx install and created trunks, extensions and groups. >> Now I'd like to hand out the Asterisk phonebook to the phones (all VVX >> 310's). Is there an easy wa

Re: [asterisk-users] how to provision asterisk's phonebook to Polycom vVX310's

On 01/21/2014 01:55 PM, Stanley van Dijk wrote: Hi, Am running a freepbx install and created trunks, extensions and groups. Now I'd like to hand out the Asterisk phonebook to the phones (all VVX 310's). Is there an easy way to do this? Best, Stanley Even the old ones could view a webpage. Ha

[asterisk-users] how to provision asterisk's phonebook to Polycom vVX310's

Hi, Am running a freepbx install and created trunks, extensions and groups. Now I'd like to hand out the Asterisk phonebook to the phones (all VVX 310's). Is there an easy way to do this? Best, Stanley -- _ -- Bandwidth and C

[asterisk-users] Unknown problem sending outbound fax

All; I'm having a problem sending an outbound fax using Asterisk-1.8.15-cert3 and the spandsp fax module using a SIP trunk. I'm seeing hundreds of these: ERROR[14423]: udptl.c:294 encode_open_type: UDPTL (SIP/runcentral_outbound-0074): Buffer overflow detected (59 + 134 > 175) Has

[asterisk-users] What is dahdi.auto_assigned_spans and why should you care? (II)

Hi again, Some other things missing from the previous message: setting E1/T1/[J1?] and upgrading. E1/T1 - an E1/T1/J1 should be configured to be either E1 or T1.There are a number of ways to set this. Specifically: quite a few different ways. With the new DAHDI initialization scheme we also

Re: [asterisk-users] DUNDI or ENUM or ?

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Le 20/01/2014 12:03, Jean-Denis Girard a écrit : > Hi list, > > I'm looking for the best / recommended solution for automatic discovery > of phone numbers for a multiple Asterisk system. This would be for an > administration, with many branches (~30),

Re: [asterisk-users] Asterisk Fax detection *11.7

Please paste the actual code. First has to be the Wait and then any other thing. Leandro 2014/1/21 Jakob-Matthias Böttger > i already added a Progess() and Wait(5) and it still does not detect > faxes. > > > Am 21.01.2014 16:53, schrieb Leandro Dardini: > > I am not sure, but try to add a wai

Re: [asterisk-users] Asterisk Fax detection *11.7

i already added a Progess() and Wait(5) and it still does not detect faxes. Am 21.01.2014 16:53, schrieb Leandro Dardini: I am not sure, but try to add a wait(2) as first command. When I want fax detection, I insert always a small delay for letting the fax detection routine to detect it. Lea

Re: [asterisk-users] Asterisk Fax detection *11.7

I am not sure, but try to add a wait(2) as first command. When I want fax detection, I insert always a small delay for letting the fax detection routine to detect it. Leandro 2014/1/21 Jakob-Matthias Böttger > Hi > > The log i've posted > > > == Using SIP VIDEO CoS mark 6 > == Using SIP RTP

Re: [asterisk-users] Asterisk not receiving call from VPN address

On Tue, Jan 21, 2014 at 10:47 AM, Paul Belanger wrote: > On Tue, Jan 21, 2014 at 12:40 AM, David Cunningham > wrote: >> Hi Paul, >> >> Using ngrep/tcpdump shows the packet clearly going from the Kamailio server >> and arriving at the Asterisk server. This is why it's a mystery that >> Asterisk do

Re: [asterisk-users] Asterisk not receiving call from VPN address

On Tue, Jan 21, 2014 at 12:40 AM, David Cunningham wrote: > Hi Paul, > > Using ngrep/tcpdump shows the packet clearly going from the Kamailio server > and arriving at the Asterisk server. This is why it's a mystery that > Asterisk doesn't see the call coming in. We tried removing the firewall (so

Re: [asterisk-users] Asterisk not receiving call from VPN address

David, It seems to me that Asterisk is not seeing/binding to your VPN interface. You need to debug that first. I would set en explicit bind statement in sip.conf to the VPN interface address and nothing else. Then start your asterisk and watch the log messages. It should confirm that it c

Re: [asterisk-users] Asterisk Fax detection *11.7

Hi The log i've posted == Using SIP VIDEO CoS mark 6 == Using SIP RTP CoS mark 5 -- Executing [12345678912 @from-sip:1] Answer("SIP/abcde-0016", "") in new stack > 0x7fd11404cd00 -- Probation passed - setting RTP source address to 123.456.789.123:17108 -- Executing [123456

Re: [asterisk-users] Dialing a SIP URI with an ";ext=" parameter

I am going to try a Lync server/asterisk integration, so I really appreciate! Leandro 2014/1/21 Lincoln King-Cliby > Ok, so now I just feel kind of stupid. After I got home I decided to play > with this a little more. > > > > After far too long I realized that part of the issue was Asterisk pa

Re: [asterisk-users] Asterisk not receiving call from VPN address

Hi Larry, No, they are on separate machines. On 21 January 2014 17:54, Larry Moore wrote: > Is Kamalio running on the same system as Asterisk? > > > On 21/01/2014 2:41 PM, David Cunningham wrote: > >> Hi Larry, >> >> Thanks for the reply. We have all of those settings left out of our >> sip.c

Re: [asterisk-users] Read factory0x7f32f4005940 was pretty quick last time, waiting for them

Hi, My problem has been solved.In my case because of CPU High load . On Mon, Jan 20, 2014 at 3:34 PM, mahdieh saeed wrote: > Hi every body > > our Calls are begging dropped for no reason and it starts with the sound > quality dropping and then the caller unable to hear our call center agents. >

Re: [asterisk-users] Asterisk not receiving call from VPN address

Hi Duncan, We have "sip set debug on" and nothing is shown, even though tcpdump/ngrep on the same server does. It's very strange. The output of "ip address list" is: [root]# ip address list 1: lo: mtu 16436 qdisc noqueue state UNKNOWN link/loopback 00:00:00:00:00:00 brd 00:00:00:00:00:00

Re: [asterisk-users] Asterisk Fax detection *11.7

It is really more interesting the receiving part. Can you paste here? Leandro 2014/1/21 Jakob-Matthias Böttger > Hello everybody > > I'm trying to enable the Digium res_fax app at my *11.7 Server. > > a fax show stats comes up with > FAX Statistics: > --- > > Current Sessions :

[asterisk-users] Asterisk Fax detection *11.7

Hello everybody I'm trying to enable the Digium res_fax app at my *11.7 Server. a fax show stats comes up with FAX Statistics: --- Current Sessions : 0 Reserved Sessions: 0 Transmit Attempts: 0 Receive Attempts : 1 Completed FAXes : 1 Failed FAXes : 1 D

[asterisk-users] Best strategy to find and solve voice quality problems

Hi, in my company we use an asterisk installation with around 50 soft- and hardphones of all kind. From time 2 time the users (almost only Softphone users) report some voice qualities... mostly echoes. These problems do not occur on all PCs at the same time and since setup of our PBX almost an