Re: [asterisk-users] R2 error Seize Timeout

y see or reset the trunk when the issue comes up to see if it matters Good luck > On 08/03/22 11:54, Duncan Turnbull wrote: > > It’s been a r we hike since we used these cards. This example may help > > > > > https://wiki.freepbx.org/plugins/servlet/mobile?contentId=73007

Re: [asterisk-users] R2 error Seize Timeout

It’s been a r we hike since we used these cards. This example may help https://wiki.freepbx.org/plugins/servlet/mobile?contentId=73007457#content/view/73007457 My thinking is it sounds like a timing error. Make sure your provider is the timing source. Once it loses time you will get dropped

Re: [asterisk-users] asterisk and maybe a freepbx question

> On 9/01/2022, at 7:11 PM, John Covici wrote: > > On Sat, 08 Jan 2022 19:17:57 -0500, > Antony Stone wrote: >> >>> On Sunday 09 January 2022 at 00:50:27, John Covici wrote: >>> >>> Hi. I am using asterisk 18.3 and freepbx. >> >> Hm, which version of FreePBX uses Asterisk 18.3? >> >>>

Re: [asterisk-users] problems with natted phones

config > > Marek > > > 2021-09-10 1:19 GMT+02:00, Duncan Turnbull : >> >> >>>> On 10/09/2021, at 4:37 AM, Marek Greško wrote: >>> >>> There are other systems running on the same hardware. It would just >>> leave open

Re: [asterisk-users] problems with natted phones

> On 10/09/2021, at 4:37 AM, Marek Greško wrote: > > There are other systems running on the same hardware. It would just > leave open ports here. > > Do not compare SIP ALG on a closed source device to an opensource > software with active development. I had no such problems in the past >

Re: [asterisk-users] problems with natted phones

} >> >> chain OUTPUT { >>type route hook output priority mangle; policy accept; >>... >> udp dport 5060 ip dscp set 0x04 >>... >> } >> } >> >> table ip6 filter { >> ct helper sip { >>type "sip" protocol udp >

Re: [asterisk-users] problems with natted phones

s > anybody have wide experience with nftables and sip? If you publish your rule set then we could look. Did you write the rules? What have you checked so far? > > Thanks > > Marek > > > 2021-09-07 10:40 GMT+02:00, Antony Stone > : >> On Monday 06 September 2021

Re: [asterisk-users] problems with natted phones

> directions because outgoing rtp stream from asterisk goes to private > address space and incoming stream is blocked. So the outgoing rtp > could not be learnt to send to nat addess. > Maybe a bug but that’s less likely than a config error. Time to debug your nftables. > Marek >

Re: [asterisk-users] problems with natted phones

normal >>>> except >>>> asterisk doesn’t appear to beseeing the rtp packet >>>>> >>>>> Thanks >>>>> >>>>> Marek >>>>> >>>>> >>>>>> >>>>>> Have fun, its all

Re: [asterisk-users] problems with natted phones

> On 6/09/2021, at 7:10 PM, Marek Greško wrote: > > Hello, > > > > 2021-09-06 2:51 GMT+02:00, Duncan Turnbull : >> Hi Marek >> >> I didn't understand your setup originally. >> >> Can you confirm this is correct: >> >&g

Re: [asterisk-users] problems with natted phones

nd remote phones behind some internet > provider. This is the only conversation to look at. > The phone private address is 192.168.100.235. > > Thanks > > Marek > > > 2021-09-05 1:11 GMT+02:00, Duncan Turnbull : > > > > > >> On 5/09/2021, at 10:21 AM, Mar

Re: [asterisk-users] problems with natted phones

> On 5/09/2021, at 10:21 AM, Marek Greško wrote: > > Hello, > > could you please answer my previous question about anonymizing several > parameters? I have the data ready, but will post after answer. I have > no clue whether I could disclose some important data not deleting > them. > >

Re: [asterisk-users] problems with natted phones

t; Hello, > > I agree my knowledge of SIP itself is poor, but I have quite well > general tcp/ip understanding. What sip parameters should be > anonymized? How about tag, branch, call-id, cseq values? > > Thanks > > Marek > > > 2021-09-04 12:36 GMT+02:00, Duncan

Re: [asterisk-users] problems with natted phones

> Thanks > > Marek > > 2021-09-04 0:40 GMT+02:00, Antony Stone > : >> On Saturday 04 September 2021 at 00:34:49, Duncan Turnbull wrote: >> >>>>> On 4/09/2021, at 7:53 AM, Marek Greško wrote: >>>>> >>>>&

Re: [asterisk-users] problems with natted phones

er's router in the previous discussion. And it made a big > improvement in the experience. > > Marek > > 2021-09-03 12:19 GMT+02:00, Duncan Turnbull : >>> On Fri, Sep 3, 2021 at 8:47 PM Marek Greško wrote: >>> >>> Hello, >>> >>> I looked

Re: [asterisk-users] problems with natted phones

another provider which had working SIP > ALG I had no problem even without nat configuration on the asterisk > side. > > The experience is clearly better after disabling SIP ALG, but we still > face problems after asterisk side reboots. > > Could you point me for what should I lo

Re: [asterisk-users] problems with natted phones

Hello, it triggered again. Even disabling RTSp ALG did not help. My fantasy ends here. It agains seems to be reboot triggered on asterisk side. Not every one. But there was surely one before it was last working. Reboot of the router on the phone side fixes the problem. Any other suggestions?

Re: [asterisk-users] HELP! I can't get my Cisco CP-7960G IP hardphone to register on my Asterisk VoIP IP PBX SIP Server with FreePBX GUI

> On 25/12/2020, at 3:08 PM, Turritopsis Dohrnii Teo En Ming > wrote: > > Hi Duncan Turnbull, > > My final issue has been resolved. Very well done Merry Xmas Cheers Duncan > > Please refer to the following post. > > Post: Addendum to Teo En Ming's Guide t

Re: [asterisk-users] HELP! I can't get my Cisco CP-7960G IP hardphone to register on my Asterisk VoIP IP PBX SIP Server with FreePBX GUI

> On 25/12/2020, at 12:40 AM, Turritopsis Dohrnii Teo En Ming > wrote: > > Hi Duncan Turnbull, > > It is a newly created PJSIP extension with default settings. I have never > configured Do Not Disturb settings before. > > Could it be something else? &g

Re: [asterisk-users] HELP! I can't get my Cisco CP-7960G IP hardphone to register on my Asterisk VoIP IP PBX SIP Server with FreePBX GUI

> On 24/12/2020, at 6:39 PM, Turritopsis Dohrnii Teo En Ming > wrote: > > Hi Duncan Turnbull, > > I have finally managed to get my Cisco 7960 IP phone to register on my > Asterisk PBX appliance on Christmas Eve 2020. > > You can read my guide here: >

Re: [asterisk-users] HELP! I can't get my Cisco CP-7960G IP hardphone to register on my Asterisk VoIP IP PBX SIP Server with FreePBX GUI

Xmas Cheers Duncan > On 24/12/2020, at 1:11 PM, Turritopsis Dohrnii Teo En Ming > wrote: > > Thank you for your replies, Duncan Turnbull. > > I am going to run tcpdump on my Asterisk PBX server. > > By the way, I found a Youtube video. > > Youtube video: Cisc

Re: [asterisk-users] HELP! I can't get my Cisco CP-7960G IP hardphone to register on my Asterisk VoIP IP PBX SIP Server with FreePBX GUI

 Sent from my iPad > On 23/12/2020, at 5:33 PM, Turritopsis Dohrnii Teo En Ming > wrote: > > Hi Duncan Turnbull, > > You can watch my Youtube video of my Cisco 7960 IP phone. > > The link is: https://www.youtube.com/watch?v=ip_F08jmmio > > My Youtube video s

Re: [asterisk-users] HELP! I can't get my Cisco CP-7960G IP hardphone to register on my Asterisk VoIP IP PBX SIP Server with FreePBX GUI

Hi there > On 23/12/2020, at 12:45 PM, Turritopsis Dohrnii Teo En Ming > wrote: > > Good morning Duncan Turnbull, > > I have posted my Asterisk PBX server debugging output previously in my > original post. The link is: > > http://lists.digium.com/pipermail/a

Re: [asterisk-users] HELP! I can't get my Cisco CP-7960G IP hardphone to register on my Asterisk VoIP IP PBX SIP Server with FreePBX GUI

but it takes a little bit of time. Cheers Duncan On Tue, Dec 22, 2020 at 10:43 PM Turritopsis Dohrnii Teo En Ming < c...@teo-en-ming.com> wrote: > Good day from Singapore, > > I seem to have found the solution at FreePBX community forums. Please > check out the following

Re: [asterisk-users] HELP! I can't get my Cisco CP-7960G IP hardphone to register on my Asterisk VoIP IP PBX SIP Server with FreePBX GUI

or external Directory location directory_url: "http://10.12.41.1/directory.html; ===== I would then recommend tcpdump to monitor traffic coming from 192.168.1.130 Tcpdump is an important tool to learn to use and you can look at all traffic c

Re: [asterisk-users] Some calls drop after 30 seconds

Hi Carlos On Tue, 8 Sep 2020, 12:36 pm Carlos Chavez, wrote: > Some users have complained that their calls drop after about 30 > seconds. The rtp timeout is usually about 30 seconds. If rtp is only 1 way then the calls will drop after 30 secs. This is usually nat/firewall related so a

[asterisk-users] Asterisk and SIP Proxy on same host = media problem

ealise reSIProcate is old but its relatively straight forward compared to Kamailio (for me), although I eventually plan to figure out Kamailio when I have more time. I don’t know whether I could affect this differently with Kamailio. Thanks very much Cheers Duncan p.s. apologies if

Re: [asterisk-users] Various extensions ring once and go to voicemail

Sent from my iPad > On 15/01/2019, at 10:34 AM, Thomas Peters wrote: > > Duncan: > > You may have it right—I took one phone and set the ring time to 60 seconds. I > now get about 4 rings on that one. > > I wonder how I can change the timing source. In one ver

Re: [asterisk-users] Various extensions ring once and go to voicemail

On Tue, Jan 15, 2019 at 7:42 AM, Thomas Peters wrote: We have an old Asterisk 1.8.7.0 system desperately need to keep alive for another 6 months or so. We had all kinds of hardware problems, so we virtualized it. Thats a while back, I think it tended to use zaptel or dahdi hardware as a

Re: [asterisk-users] SIP connections over OpenVPN connection get one-way voice.

Sent from my iPhone > On 19/04/2017, at 11:43 AM, Ernie Dunbar <maill...@lightspeed.ca> wrote: > >> On 2017-04-18 03:38 PM, Duncan Turnbull wrote: >> -- Original Message -- >> From: "Ernie Dunbar" <maill...@lightspeed.ca> >> To: "

Re: [asterisk-users] SIP connections over OpenVPN connection get one-way voice.

-- Original Message -- From: "Ernie Dunbar" To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" Sent: 19-Apr-17 10:25:59 AM Subject: [asterisk-users] SIP connections over OpenVPN connection get one-way voice. Hi

Re: [asterisk-users] Touch tone stutter

On 23/11/16 13:49, Pete Mundy wrote: One direction that may be worth exploring further is his ATA's config (or perhaps swapping it for a different model). Eg adjusting echo cancellation or line impedance settings. Is the ATA he is using the same as the ATA you use? Failure to correctly

Re: [asterisk-users] Issue command to force SIP client to re-register

Cheers Duncan If not, is there some other standard way to do so – or would I have to post/get to a web GUI of the phone (unique to each model) to force a reset, etc. -Raj- -- _ -- Bandwidth and Colocation Provided by http

Re: [asterisk-users] Configuration management and update deployment - what do you use?

On 19/10/16 05:57, Ludovic Gasc wrote: +10 for Ansible. We use that on our production. Okay, I will investigate Ansible Thanks very much Cheers Duncan -- Ludovic Gasc (GMLudo) http://www.gmludo.eu/ 2016-10-18 14:00 GMT+02:00 marek cervenka <cerva...@gmail.com <mailto:cerva...@gma

[asterisk-users] Configuration management and update deployment - what do you use?

was just curious how people deploy asterisk across multiple platforms and keep them all up to date? What tools are good for this sort of thing? Thanks very much Cheers Duncan -- _ -- Bandwidth and Colocation Provided by http

Re: [asterisk-users] iptables for SIP talk to other port

eth0 -n udp and host 192.168.1.3 should show you packets between your machine and your odd host Cheers Duncan On 17/10/16 11:55, Mike wrote: I'm by no means an iptables guru... Not sure if it's necessary to enable forwarding via: echo "1" > /proc/sys/net/ipv4/ip_forward Also h

Re: [asterisk-users] how to decrypt encrypted SIP user's secret

On 29/06/16 16:37, Nathan Anderson wrote: You must mean that engineer before you used "md5secret" instead of "secret" for each user in sip.conf? If so, why can't you just copy the md5secret line from the old server to the new server for each user? -- Nathan

Re: [asterisk-users] Trouble getting Asterisk Running with FreePBX 11

On 22/04/16 01:52, Daniel Chavez wrote: Hi, I recently had to reinstall Asterisk and FreePBX. asteirsk 11.20 and FreePBX 12. This is running on Centos 6.7 32 bit. When I use amportal start It comes up with the errors below Error in argument 1, char 2: option not found r

Re: [asterisk-users] Best timing source?

On 07/04/16 09:00, Carlos Chavez wrote: On 4/6/16 2:39 PM, Duncan Turnbull wrote: I am starting to think that the problem may be in the server hardware itself. We are using a Dell R220 with 8gb of ram and 2 hard disks in a Raid 1 configuration (Linux Raid). We are using CentOS 7

Re: [asterisk-users] Best timing source?

> On 7/04/2016, at 6:01 AM, Carlos Chavez wrote: > >> On 4/5/16 3:17 PM, Joshua Colp wrote: >> Carlos Chavez wrote: >>> I am currently having a voice quality problem with one of our Asterisk >>> servers. We have checked the network and we have found no problems that

Re: [asterisk-users] Lost outgoing SIP packets

One issue that can catch you is a packet MTU limit in your path to your SIP box lower than your standard MTU. You can check that with ping -s 1500 option Cheers Duncan On 01/04/16 17:12, Pete Mundy wrote: Roel, Just another thought bouncing around... Your ifconfig output was specific

Re: [asterisk-users] Rasterisk freeze on 4G link

st connectivity before with this Also the session is probably timed out rather than gone, in 10-15 mins maybe less it will come back (or does for me) Cheers Duncan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.

Re: [asterisk-users] Rasterisk freeze on 4G link

> On 4/03/2016, at 5:31 AM, Olivier wrote: > > Hello, > > I'm remotely managing an asterisk setup using an OpenVPN client on this > Asterisk box, connecting to an OpenVPN server of mine). > > This box is mainly connected to PSTN. > It is also connected to the Internet,

Re: [asterisk-users] Voice recognition IVR Is it possible?

speech than NZ. I think there are different language options too. Cheers Duncan On 23/02/16 08:56, John Kiniston wrote: I think I saw an old page on the voip-info wiki on how to use CMU Sphinx with asterisk. http://www.voip-info.org/wiki/view/Sphinx IMHO It's not going to be anywhere as good

Re: [asterisk-users] How to deal with error messages passed as Early Media

is not giving you an error code then you have an issue. Cheers Duncan On Wed, Feb 3, 2016 at 8:41 AM, Olivier <oza.4...@gmail.com <mailto:oza.4...@gmail.com>> wrote: Hello, I'm trunking with an ITSP that, when treating an outbound to an unknown destination, either

Re: [asterisk-users] How exactly does asterisk know what IP to send RTP traffic to?

, which is where Asterisk / client will send RTP to respectively . You can look at this using tcpdump. c= is what you are looking for. Some formal examples https://tools.ietf.org/html/rfc4317 Cheers Duncan On 24 Nov 2015, at 10:01, Kevin Long wrote: Hello, I have a somewhat confusing use case

Re: [asterisk-users] Help With Physical Layer

Good luck Cheers Duncan On Wed, May 20, 2015 at 11:34 AM, Tony Kasule timotsm...@gmail.com wrote: Hello users, I have a Digium Te235 and asterisk 13 which have worked well with 1 carrier but we have failed to add a 2nd carrier. The second telco brings their E1 line over finer, terminated

Re: [asterisk-users] outgoing calls not working on sangoma A200

or the channel numbers don't work or the tones aren't working. Possibly your PSTN provider has a problem but you can check this by plugging in an analogue phone to the lines and seeing if you can make a call Cheers Duncan On 20 Jun 2015, at 9:22, kazabe wrote: Hi. I have 3 lines connected

Re: [asterisk-users] Strange and complete failure of Asterisk 1.8

Duncan On 27/05/2015, at 11:55 pm, Stefan Viljoen viljo...@verishare.co.za wrote: Hi all We've had a very strange failure on an Asterisk 1.8 install that has been running for about a year at a customer site. The physical hardware is fine, all other services off the Centos 6.5 server

Re: [asterisk-users] Script to Program Snom Phones

If you use freepbx you can do it with endpoint manager http://schmoozecom.com/endpoint-manager.php It costs I think in the latest freepbx version but there will be earlier versions around It's just generating templates by mac for the tftp server On 10/04/2015, at 4:37 am, Tafadzwa Nyabasa

Re: [asterisk-users] IAX port

/firewall so I would say its a router problem in the Destination NAT process. Cheers Duncan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every

Re: [asterisk-users] Investigating international calls fraud

On 29 Jan 2015, at 11:07, Administrator TOOTAI wrote: Le 28/01/2015 22:03, Steven McCann a écrit : Hello, Hi I'm investigating a situation where there was a hundreds of minutes of calls from an internal SIP extension to an 855 number in Cambodia, resulting in a crazy ($25,000+) bill

Re: [asterisk-users] VPN SIP Phone | PC Traffic

On 9/04/2014, at 10:42 pm, Positively Optimistic positivelyoptimis...@gmail.com wrote: We are using vpn routers to connect home users back to our office network. Basically, shipping a mikrotik router that 'calls home' and establishes a vpn connection for the pc and phone that are

Re: [asterisk-users] Asterisk 1.6

Cheers Duncan On 5/04/2014, at 4:36 am, motty cruz motty.c...@gmail.com wrote: that sounds feasible, Thanks Michelle, On Fri, Apr 4, 2014 at 8:25 AM, Michelle Dupuis mdup...@ocg.ca wrote: If you know your users are all from with your country, or state, or even city, you could restrict

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

on asterisk or your default gateway showing how to get back to the 172. addresses i.e. pointing to the vpn box for 172 addresses? Cheers Duncan On 21 January 2014 05:30, Paul Belanger paul.belan...@polybeacon.com wrote: On Sun, Jan 19, 2014 at 9:51 PM, David Cunningham dcunning...@voisonics.com

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

On 21/01/2014, at 6:40 pm, David Cunningham dcunning...@voisonics.com 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

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

on the phone so the phone will also respond to it. Cheers Duncan On 21/01/2014, at 7:18 pm, David Cunningham dcunning...@voisonics.com wrote: Hi Duncan, Thank you for your reply. Here's the netstat: [root]# netstat -udpln | grep asterisk udp0 0 0.0.0.0:50000.0.0.0

Re: [asterisk-users] Convert Asterisk Appliance (AA50) to Open Asterisk?

Cheers Duncan On 29/12/2013, at 8:37 am, Todd R. tjrl...@live.com wrote: May not be what you are looking for exactly but search Google for Nerd Vittles BeagleBone. I am not suggesting you use that exact solution but, reading the article with give you all sorts if ideas about what you

Re: [asterisk-users] Jetway, Atom, and Digium cards - play well together?

parts. They are not Jetway motherboards but have used the D525 a lot. Just avoid this ethernet controller: Intel 82574L Gigabit controller http://blog.krisk.org/2013/02/packets-of-death.html Cheers Duncan On 5/12/2013, at 5:40 pm, Ira i...@extrasensory.com wrote: Hello Rodrigo, Wednesday

Re: [asterisk-users] Recurring SIP problem with asterisk 11.6 11.7

to follow it up, but usually just set domain names to IPs to avoid it Cheers Duncan On 13/11/2013, at 1:37 pm, Jeremy Kister asterisk...@jeremykister.com wrote: I have regularly (once a week, once per few hundred calls?) been having problems with Asterisk's SIP stack not responding to packets

Re: [asterisk-users] Tired of dropouts and garbled phone calls - where to go next?

On 29/10/2013, at 9:55 am, Mike mike...@microdel.org wrote: On Mon, 28 Oct 2013, Eddie Mikell wrote: All, The users in our organization are well, quite frankly, sick of phone service that is being provided. The choppy phone calls, and drop outs are detrimental to our sales force. I've

Re: [asterisk-users] Installing asterisk and dahdi on ubuntu

On 29/08/2013, at 10:02 PM, Thorsten Göllner t...@ovm-group.com wrote: Permissions: take a look at /etc/udev/rules.d/dahdi.rules. Last line. OWNER and GROUP should be the same as the user running the asterisk process (root or asterisk?). Am 29.08.2013 11:47, schrieb bilal ghayyad:

Re: [asterisk-users] using E1 PRI lines

On 30/07/2013, at 4:22 PM, Akib Sayyed akibsay...@gmail.com wrote: I didnt understand what you were saying.can you please explain I am using digium cards sent from android E1 PSTN line interfaces are either unbalanced 75 ohm( and used to use BNC connectors ) or a 120 ohm balanced

Re: [asterisk-users] RED on DAHDI channel

in and checked dial tone and called the line? Usually its a symptom of something not quite right so its worth paying attention to Cheers Duncan 54 FXOFXSKS (EC: VPMOCT032 - INACTIVE) 55 FXOFXSKS (EC: VPMOCT032 - INACTIVE) RED 56 FXOFXSKS (EC: VPMOCT032

Re: [asterisk-users] Sangoma Wanpipe Driver

else Sangoma support will dial in and help you if you ask them Cheers Duncan On 13/05/2013, at 9:29 PM, Yves A. yves...@gmx.de wrote: Hi, I migrated from asterisk 1.6 to 11.3. The Server has a Sangoma A104 quadPri card installed. OS is a fresh installed Ubuntu 12.04 64bit libpri

Re: [asterisk-users] Logging SIP connection status for review

could easily adapt one to do registrations. I find the sip peers and calls in the last hour quite interesting Cheers Duncan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join

Re: [asterisk-users] asterisk with 1000 extensions

On 7/03/2013, at 9:29 PM, Kamlesh Kumar kamlesh_...@hotmail.com wrote: On Thu, 7 Mar 2013, Bharat Lalcheta wrote: You can use ATA box with pstn phone to reduce cost. Are you wiring a building where multiple-line SIP gateways make sense? How about a description of what you are trying

Re: [asterisk-users] Extension cant pickup calls but can transfer.

On 8/03/2013, at 7:46 AM, Leandro Dardini ldard...@gmail.com wrote: If I was in your shoes, I'll check in the elastix mailing list... Asterisk itself can't be blamed. Leandro I am typing from my mobile phone... Il giorno 07/mar/2013 19:06, Luis H. Forchesatto

Re: [asterisk-users] What would cause a drop between two asterisk systems?

run interoffice from NZ to Australia and many systems in between. No issues at all Cheers Duncan I have seen this assertion from time to time, but never any real details There is a world wide network of users who communicate using IAX, and many with PSTN service from providers using IAX

Re: [asterisk-users] Asterisk calls between 2 private networks

should see where the rtp packets are going to and from when the call comes up and what sip packets are actually saying to each other But renumbering would help especially if you did want a vpn or other networking between the sites Cheers Duncan

Re: [asterisk-users] Need Help

anymore or Centos (ubuntu is my preference) but there are bound to be others who can help solve specific problems for you It won't be particularly hard, but picking your priority issue and focussing on that is a good first step Good luck Cheers Duncan image001.jpg Joe Ruffolo Director

Re: [asterisk-users] Open source asterisk GUI options

On 18/01/2013, at 4:28 PM, Jim Boykin boykin...@gmail.com wrote: Hi, We are looking for the web based console for our asterisk system. We came across AsteriskNow but it's kind of bundle and hence not usable for us. What we need is a separate GUI package which we can add to our existing

Re: [asterisk-users] Recorded reminders

in php Good luck Cheers Duncan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk

Re: [asterisk-users] IAX2 over OpenVPN connection.... working but

On 10/12/2012, at 8:54 AM, Stephen Brown stephen.brow...@gmail.com wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 So a friend of mine and I setup a static key based point to point OpenVPN connection from my box to his for the express intent of carrying IAX traffic encrypted.

Re: [asterisk-users] SIP and RTP on different IP's

://forums.asterisk.org/viewtopic.php?f=1t=84018 Have you tried making the preferred route to these addresses go out eth1, thus getting the required address? Ultimately seems odd the firewall allows access in but not out, guessing you have no control over that? Good luck Cheers Duncan

Re: [asterisk-users] SIP and RTP on different IP's

On 24/11/2012, at 2:19 AM, Tiago Geada tiago.ge...@gmail.com wrote: Hello Folks, I am looking for a way that makes asterisk tell remote SIP party that the IP where they will send RTP is not the same as the one I am comunicating via SIP Can this be done anyhow? I can try and explain:

Re: [asterisk-users] Sending calls from behind NAT

the private IP address (this what I understood if I am right). I tried to use the following in the [general] settings in the sip.conf localnet=192.168.10.2/255.255.255.254 externadd =196.40.164.239 This should be externip not externadd You are still sending them your local address Cheers Duncan

Re: [asterisk-users] high capacity analog - sip gateway

. I prefer to be able to see everything on the CLI Cheers Duncan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http

Re: [asterisk-users] Connecting Skype to Asterisk

On 13/10/2012, at 7:54 AM, Christopher Harrington ch...@acsdi.com wrote: On Fri, Oct 12, 2012 at 1:44 PM, Philip Bennefall phi...@blastbay.com wrote: Hi all, I have an Asterisk PBX under development, that I would like to link to a Skype account if possible. The idea is that people would

Re: [asterisk-users] Failover router recommendation

a call forward on the primary numbers to the secondary if there is a failure Cheers Duncan Niccolò Belli darkba...@linuxsystems.it wrote: Il 09.10.2012 21:24 Mike Diehl ha scritto: I hope no one considers this off topic... I have a phone customer who wants 2 Internet connections so

Re: [asterisk-users] Asterisk Dahdi 1.6.2.23 Iaxmodem

On 2/08/2012, at 6:37 AM, Tim Nelson tnel...@rockbochs.com wrote: - Original Message - Yup, there is your problem. Tell hylafax to extend the amount of time before it times out. We're a bit off topic for the Asterisk list now, but in your Hylafax config.ttyIAX0 config file,

Re: [asterisk-users] Asterisk Dahdi 1.6.2.23 Iaxmodem

Sorry pushed send too fast On 2/08/2012, at 5:59 AM, Eric Wieling ewiel...@nyigc.com wrote: Yup, there is your problem. Tell hylafax to extend the amount of time before it times out. -Original Message- From: asterisk-users-boun...@lists.digium.com

Re: [asterisk-users] So long, and thanks for all the fish!

much Kevin, I have sincerely appreciated your insights and ability to help. I wish you great success in your next role Best wishes Duncan -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan

Re: [asterisk-users] IAX2 Registered OK without IP

On 27/07/2012, at 8:16 AM, Alejandro Imass a...@p2ee.org wrote: On Tue, Jul 24, 2012 at 3:54 PM, Alejandro Imass a...@p2ee.org wrote: On Tue, Jun 12, 2012 at 4:04 PM, Alejandro Imass a...@p2ee.org wrote: we upgraded to 1.8.13.1 and we have much the same problem although after the upgrade

Re: [asterisk-users] callback on busy

On 27/07/2012, at 3:42 AM, Richard Mudgett rmudg...@digium.com wrote: I know the topic comes back like boomerang , but I did not find a nice solution. Does someone has/knows how to achieve call back on busy otherwise called camping? If one is calling the extension and it is busy, then

Re: [asterisk-users] FreePBX: using context other than the default context and the generation for the configuration

= _X.,1,Answer() exten = _X.,n,Verbose(Blocked an 0900 trunk call) exten = _X.,n,Playback(custom/0900-block) exten = _X.,n,Hangup And you can record a message or send it somewhere else or whatever you feel like Cheers Duncan On 12/07/2012, at 6:11 PM, SamyGo wrote: Great tip Duncan

Re: [asterisk-users] chan_sip sending from wrong source address when multiple interfaces are used

We got round it by setting the outbound proxy to the unexpected address and then everything seems happy - we are using Yealink though so it maybe different for other phones You could use IP Tables I guess too Cheers Duncan

Re: [asterisk-users] FreePBX: using context other than the default context and the generation for the configuration

that the route applies to Cheers Duncan On 12/07/2012, at 4:52 PM, SamyGo govoi...@gmail.com wrote: See Route-Permissions module, It lets you restrict certain phones/extensions to follow a dial-plan pattern and dial out to the defined trunk etc meanwhile not breaking any other functionality

Re: [asterisk-users] FreePBX: using context other than the default context and the generation for the configuration

Also you can limit outbound routes to certain extension ranges which can avoid the need for contexts but its up to you Cheers Duncan On 6/07/2012, at 4:20 PM, SamyGo wrote: Hey, If you want to have all the dialplan features for your extensions and still need to implement some outbound

Re: [asterisk-users] Forcing SIP trunk matching order?

Hi James On 29/06/2012, at 6:19 AM, James Lamanna wrote: Hi, I have a bunch of different customers on an Asterisk Box (the PBX). This Asterisk Box is behind another Asterisk box that provides a PSTN connection. Up to this point I've been using IAX between the 2 Asterisk boxes, but I would

Re: [asterisk-users] Asterisk 1.8.13.0 / problem with cdr logging (mysql, odbc)

I think you need the DSN in car_odbr.ini to refer to the one in res_odbc.conf as below On 19/06/2012, at 3:52 AM, Thorsten Göllner wrote: Hi, I am trying now for over 4 hours setting up cdr-logging via odbc into a mysql database. But with no success. Do you have any hint for me? cat

Re: [asterisk-users] CDRs do not record in asteriskcdrdb using Digium repository

Not sure about yum installs but in 1.8 I have had to move to using odbc as the method to populate the mysql database http://www.voip-info.org/wiki/view/Asterisk+cdr+odbc Cheers Duncan On 17/06/2012, at 4:22 AM, Bruce B wrote: Hello, I have done yum install asterisk18 freepbx and it has

Re: [asterisk-users] Setting span orders with Astribank and Sangoma A101

On 12/06/2012, at 12:00 AM, Tzafrir Cohen wrote: On Sun, Jun 10, 2012 at 10:10:29PM +1200, Duncan Turnbull wrote: Hi All Just a quick check on the best way to ensure multiple cards/devices load in the correct order. Asterisk 1.8 with Sangoma A101 had no problems until we introduced

[asterisk-users] Setting span orders with Astribank and Sangoma A101

me set the spans within Dahdi so they always appear on the same number Thanks very much Cheers Duncan-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory

[asterisk-users] Getting unwanted pager email from Asterisk voicemail

the user to compose and send a voicemail while inside [default] 121 = 1234,Duncan testing,dun...@e-simple.co.nz,,attach=yes|saycid=no|envelope=no|delete=no I get the voicemail with attachment Subject [PBX]: New message 1 in mailbox 121 Dear Duncan testing: Just wanted to let you know you were

Re: [asterisk-users] Getting unwanted pager email from Asterisk voicemail

Thanks but my voicemail conf line looks like this 121 = 1234,Duncan testing,dun...@e-simple.co.nz,,attach=yes|saycid=no|envelope=no|delete=no There is no pager email address so I am not sure why its sending a pager email Cheers Duncan On 1/06/2012, at 1:51 AM, cov...@ccs.covici.com wrote

Re: [asterisk-users] Getting unwanted pager email from Asterisk voicemail

I get the voicemail with attachment Subject [PBX]: New message 1 in mailbox 121 Dear Duncan testing: Just wanted to let you know you were just left a 0:08 long message (number 1) in mailbox 121 from 21722545, on Thursday, May 31, 2012 at 08:45:02 PM so you might

Re: [asterisk-users] Fax Server for Asterisk

I had Hylafax sending 1000s of faxes a day twice a week connected to analogue lines using asterisk and iaxmodem for about 4 years. People don't use fax much anymore though No problems whatsoever Cheers Duncan On 31/05/2012, at 6:49 AM, Danny Dias wrote: Just to clarify, i were using fax

Re: [asterisk-users] Fax Server for Asterisk

confirm? any place to check How-To on Hylafax and Iaxmodem? Many thanks!!! Cheers Duncan-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every

Re: [asterisk-users] Asterisk Vs FreeSWITCH for Fax

. Cheers Duncan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing

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