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

advice on google. >>>>> >>>>> Asterisk cli did not show anything interesting. I tried pjsip set >>>>> logger verbose on, but no logs showed anywhere. What am I doing wrong? >>>>> >>>>> 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

Thank you very much for your kind assistance. > > > > > On 2020-12-21 09:58, Duncan Turnbull wrote: > > Hi there > > > > I would normally highlight the part but the email is so long I thought > > I would just note what I can see > > > > It appears

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 I would normally highlight the part but the email is so long I thought I would just note what I can see It appears the Cisco is downloading files. None of the SIP traces show the IP of the phone of the extension Your proxy is at 192.168.1.9 Your phone is at 192.168.1.130 These are

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

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 version (and I can’t recall

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] 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] 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] How exactly does asterisk know what IP to send RTP traffic to?

HI Kevin Is your VPN set as a localnet? The externip only tends to cope with the firewall address. If you put the VPNs in the localnet lists then it won't use NAT to find them. In answer to your question, the SIP session description in the call setup has the IP for media for both parties,

Re: [asterisk-users] Help With Physical Layer

-- Original Message -- From: Tony Kasule timotsm...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: 30/06/2015 8:34:47 p.m. Subject: Re: [asterisk-users] Help With Physical Layer Hello, Anyone to help me with this issue?

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

Hi there This has happened to me before when I changed the tone duration, it was too long and the PSTN receiver no longer understood the tones, but it seems unlikely nothing has changed. The cli or logs should show you whats happening, something will be blocking the call, either the group

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

DNS failure could do this Asterisk used to get stuck in a symmetric DNS request wait state which meant everything ground to a halt as it waited for a reply while DNS timed out. The recommended option was either ip only or a DNS proxy that failed fast this letting asterisk continue Cheers

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

On 10 Feb 2015, at 12:22, Jose Flores Galicia wrote: 2015-02-09 14:36 GMT-06:00 jg webaccounts...@jgoettgens.de: Hi! Sometimes IAX peers are not reachable and with iax2 set debug on I get something like this Tx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: PONG

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

Another option we like, but i depends on your preferences is to run them over openvpn. Works for Mac, Linux and Windows clients. Since all out clients are under our control we use openvpn a lot and yealink and other phones have it built in so they can connect directly once initially setup

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

On 21/01/2014, at 10:24 am, David Cunningham dcunning...@voisonics.com wrote: Hi Paul, The ngrep on the Asterisk server does show it being received. Have you any idea what would prevent it getting from the network stack to Asterisk on that machine? Have you got a static route on

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

Cool That looks like it is arriving at Asterisk - are you sure asterisk is not getting it? If you turn on sip debug in asterisk can you see the SIP packets? It maybe asterisk is ignoring them or replying to them but its going out an interface you hadn’t thought of, I have had that a few times.

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

I think that's a good idea, I turned an AA50 into just a trunk device for a sip box and it worked for a long time The other things that are small and work are the atcom ip Pbx series http://www.atcom.cn/products_ippbx.html They are pretty cheap in NZ but not as low as the beaglebone suggestion

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

We use Atoms with SSDs for customers and they work well We have a some with PCI on the motherboards and haven’t had any issue other than a single issue where a reinstall of the OS cleared up poor ethernet performance. Use almost no power, and with SSDs can almost avoid fans and thus moving

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

Any chance DNS is dying about the same time the problem occurs I get this occasionally every 6-12 months and usually because DNS got messed up and then something didn’t fall back into place when it recovered - networking looks okay on the machine but asterisk is stuck. I have been meaning to

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

Hi Mitch On 28/05/2013, at 5:14 AM, Mitch Claborn mitch...@claborn.net wrote: Asterisk 11.1 We have a situation where one of our incomings POTS lines will not answer. There are 2 lines configured by the Telco as a rollover group (rings the line that is not busy) and they feed into a

Re: [asterisk-users] Sangoma Wanpipe Driver

We have had challenges with the latest kernel versions on Ubuntu and sangoma wanpipe drivers An older kernel - no problem, latest ones, sometime risky. There are release notes on their site stating the supported versions so it might pay to check that But if it compiled ok it might be something

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

On On Wed, 10 Apr 2013, Carlos Alvarez wrote: Is anyone using something to log SIP results (connected/not, latency) that they really like? We do some logging using simple scripts writing the results of sip show peers to a text file if customers report issues, but it would be nice to

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?

On 6/03/2013, at 9:06 AM, John Novack jnov...@stromberg-carlson.org wrote: Carlos Alvarez wrote: On Tue, Mar 5, 2013 at 2:32 PM, Hose hose+aster...@bluemaggottowel.com wrote: We have an asterisk frontend terminating all our SIP phones to, and an asterisk backend with a wildcard PRI

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

On 8/02/2013, at 6:49 AM, Frank fr...@efirehouse.com wrote: I thought about that. I will give it a shot tonight and will post back my results in here. Thanks On 2/7/13 12:39 PM, Eric Wieling wrote: The easiest thing to is renumber one of the networks so they are not using the same

Re: [asterisk-users] Need Help

Hi Joe On 18/01/2013, at 9:05 AM, Joe Ruffolo j...@mrkgroup.com wrote: Hi all! In need of some serious help. We currently run Trixbox and Cent Os on a 2u server for our small business’s phones system. We are using some Polycom Soundpoint IP phones. The whole thing came crashing down over

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

On 13/01/2013, at 10:52 PM, Anselm Martin Hoffmeister ans...@hoffmeister-online.de wrote: Am 13.01.2013 03:17, schrieb Adolphus Enaboifo: Hi List Members , its been about one months since I built my first Asterisk server. What I want to know is: are there ways to make Asterisk take recorded

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

On 23 November 2012 19:39, Duncan Turnbull dun...@e-simple.co.nz wrote: 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

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

On 14/11/2012, at 10:16 AM, bilal ghayyad bilmar...@yahoo.com wrote: Dears; It seems my service provider is requesting a complicated settings to allow me to send from behind NAT. What they said: It shouldn't matter as long as you are handling the NAT correctly your end. We do not

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

On 26/10/2012, at 10:09 AM, jon pounder j...@inline.net wrote: On 10/25/2012 05:01 PM, Steve Totaro wrote: That is just silly. You mean to say that the Adtran and the Adit units are not as reliable as these new devices. No way. I have had channel banks fail yes, and I stick by my

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

On 10/10/2012, at 9:54 AM, cov...@ccs.covici.com wrote: I am sure Mikrotik routers will do this also, although I have not tried it. Mikrotik can do this but it takes some setup. They are very powerful but what you are asking is complex and may require the following - 2 ethernet upstreams or

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!

On 1/08/2012, at 1:59 AM, Kevin P. Fleming kpflem...@digium.com wrote: I've been with Digium for just over seven years, and it's been an incredible experience that I wouldn't have traded for anything. When Mark Spencer invited me to visit Digium (and Huntsville) in early 2005, I could not

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

:) On Thu, Jul 12, 2012 at 10:29 AM, Duncan Turnbull dun...@e-simple.co.nz wrote: You can also specify routes with an callerid qualifier as 09XX/20X This would only have it apply to extensions in the 200-209 range That route can then point to a trunk going nowhere if you want to block them

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

Similar problem On 12/07/2012, at 4:36 PM, Jeff LaCoursiere wrote: On Thu, 2012-07-12 at 15:49 +1200, Alec Davis wrote: I've seen similar. We tried 4 interfaces. On 4 lans, are these considered to be overlapping? 192.168.1.1 192.168.2.1 192.168.3.1 192.168.4.1 Running openvpn on

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

You can also specify routes with an callerid qualifier as 09XX/20X This would only have it apply to extensions in the 200-209 range That route can then point to a trunk going nowhere if you want to block them In freepbx there is a field in outbound route page to select callerid that the

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

The module is custom contexts - its a third party option in the module admin But you can write contexts in the extensions_custom.conf if you want to I wouldn't use freepbx to generate your code - its quite complex code for a roll your own system, but very useful if you learn its gui and options

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

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 an Astribank. root@pabx377:/etc/asterisk# dahdi_hardware -v usb:001/004 xpp_usb+ e4e4:1162 Astribank-modular

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

Hi All I am not sure why but I am getting a pager email as well as a voicemail email when a voicemail is left. I am guessing its a setting somewhere but I can't find it The system is Ubuntu with Asterisk 1.8.12 from source. I am using Freepbx for the configs but freepbx doesn't do much to

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

On 1/06/2012, at 1:24 AM, Danny Nicholas wrote: My guess is that your email provider is forwarding the message since Asterisk should send the same content to both places. Thanks but they are two different messages i.e. one is the standard voicemail one, the other the pager email as below

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

On 30/05/2012, at 10:02 AM, Danny Dias wrote: Hi all, Does Hylafax and IAXmodem works with analog lines? or only with E1? Hylafax can use any fax modems: available E1 or analogue, ISDN as long as it can talk to it to send the commands If you add asterisk and iaxmodem then hylafax can

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

Hi Anita On 4/05/2012, at 12:27 AM, Anita Hall wrote: Hi We are using Spandsp + FreeSWITCH for receiving Fax over T.30 E1/PRI and the results make us sad :( I am presuming you do mean T.30 (standard fax protocol but people don't mention it much) not T.38 as I am not familiar with that

Re: [asterisk-users] Asterisk Capacity

Hi Ashish On 4/05/2012, at 3:41 AM, Ashish Agarwal wrote: Hello, We are currently working on a project where using .call file on asterisk spool, outbound calls will be made from a pri line and a voice clip will be played. We know that pri has a capacity of handling only 30 channels at

Re: [asterisk-users] chan_sip.c:2901 __sip_xmit: sip_xmit of 0x8ef2130 returned -1: Operation not permitted ??

Usually its a firewall issue, or at least it has been for me Its saying it can't form sip packets, and that will be because something isn't letting it, Cheers Duncan On 26/04/2012, at 8:15 PM, Olivier CALVANO wrote: Anyknow know this problems ? I read on the net that it's a possible

Re: [asterisk-users] FXO - GSM Gateway Problem

Hi I have had issues with wiring for incoming calls causing what looks like a hangup when answered but in those cases the call stays up and asterisk thinks its a new call. Have seen it on Avaya too If it is wiring can you test a different incoming line? Cheers duncan On 19/04/2012, at

Re: [asterisk-users] [OT] FreePBX + Trunk over VPN + Local LAN

Either give it a 2nd address on the nic that can access the VPN modem You can have lots of addresses on a nic to access different sinners on the LAN Or just make sure the gateway knows to route the ipvpn traffic via the vpn modem Cheers Duncan On 24/03/2012, at 3:55 PM, Eliezer Croitoru

[asterisk-users] Asterisk won't start - trap invalid opcode

Hi there Happy New Year I have a new install of asterisk 1.8.8.1 on ubuntu server 3.0.0-14-server #23-Ubuntu SMP Mon Nov 21 20:49:05 UTC 2011 x86_64 x86_64 x86_64 GNU/Linux It has a Sangoma A200 card and I thought should be fairly standard but I have a new error when trying to start asterisk

Re: [asterisk-users] Asterisk won't start - trap invalid opcode

On 4/01/2012, at 11:47 PM, A J Stiles wrote: For what it's worth, I once tried installing Asterisk on an old VIA C7 box; and it turns out that this processor, while detecting as an i686, doesn't implement the full i686 instruction set -- and Asterisk is trying to use one of the

Re: [asterisk-users] Asterisk won't start - trap invalid opcode

) On 5/01/2012, at 12:13 AM, Duncan Turnbull wrote: On 4/01/2012, at 11:47 PM, A J Stiles wrote: For what it's worth, I once tried installing Asterisk on an old VIA C7 box; and it turns out that this processor, while detecting as an i686, doesn't implement the full i686 instruction

Re: [asterisk-users] Asterisk won't start - trap invalid opcode

On 5/01/2012, at 8:06 AM, Steve Edwards wrote: On Wed, 4 Jan 2012, A J Stiles wrote: If you stick a /* harmless comment */ in this file and re-save it, this will give the file a new modification time. Then run `make` again. It will recompile just localtime.c (this being the only source

Re: [asterisk-users] Asterisk won't start - trap invalid opcode

On 5/01/2012, at 12:21 PM, James Cloos wrote: DT == Duncan Turnbull dun...@e-simple.co.nz writes: DT I have a new install of asterisk 1.8.8.1 on ubuntu server DT 3.0.0-14-server #23-Ubuntu SMP Mon Nov 21 20:49:05 UTC 2011 x86_64 x86_64 x86_64 GNU/Linux DT The only errors I can see

Re: [asterisk-users] Strange network issue

On 28/07/2011, at 8:41 PM, Paul Hayes p...@provu.co.uk wrote: On 28/07/11 02:58, Mike Diehl wrote: Any ideas? Mike. I'd go on site if possible and see what actually happens at 19:00. Set up a wireshark trace capturing all traffic through their router. -- I am picking a cleaner

Re: [asterisk-users] iptables for Asterisk - Any good guides out there?

Shorewall is a useful way of setting up iptables http://www.shorewall.net/ Cheers Duncan On 15/05/2011, at 1:46 PM, Jeremy Kister wrote: On 5/14/2011 9:45 PM, Jeremy Kister wrote: http://jeremy.kister.net/code/asterisk/iptables.init oops, that's:

Re: [asterisk-users] Cordless VoIP Phones and Access Point hand-off?

Not sure if you are issuing DHCP at the access point or from a central control From a central control should allow seamless roaming within different APs, assuming easy auth to the AP, the only issue you get is when the handset dithers between choosing signals from one or the other, and thats

Re: [asterisk-users] One Way Audio

, length 172 On Wed, Mar 9, 2011 at 7:01 PM, Duncan Turnbull dun...@e-simple.co.nz wrote: Can you do a tcpdump to look at the rtp streams on your box and check they are both generating and aiming at the right places IAX will have no issue with NAT/firewall but SIP / RTP can. tcpdump -nn

[asterisk-users] DTMF not being heard correctly by far end conference system

Hi there I have two different asterisk systems (both 1.4) whose dtmf tones are not being picked up by a particular conference system users are dialling into. I can call myself with the phones and hear the tones, but I am guessing perhaps they are too short or somehow different. I have looked

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