Re: [asterisk-users] *****SPAM***** Re: IAX port 4569

my port is 4569 is in Stealth mode (so it is closed) :-/ > > > Thelma > On 06/05/2017 02:19 PM, Victor Villarreal wrote: > > I think you need to increase verbose output and search in > > /var/log/asterisk/full for any error message related to IAX2 registration > >

Re: [asterisk-users] IAX port 4569

I think you need to increase verbose output and search in /var/log/asterisk/full for any error message related to IAX2 registration or simil. 2017-06-05 17:12 GMT-03:00 : > No, I don't think it is IP table issue, I've not upgraded dd-wrt for a > while and it was zoiper

Re: [asterisk-users] IAX port 4569

No. The 0.0.0.0 listen address is fine. El 5 jun. 2017 10:06, escribió: > I'm getting: > netstat -a |grep 4569 > udp0 0 0.0.0.0:45690.0.0.0:* > > Should I be getting localhost IP? > > Thelma > > On 06/05/2017 06:48 AM, the...@sys-concept.com

Re: [asterisk-users] IAX port 4569

Another idea: * Run netstat -tulpn command on Linux box AND look if there are an Asterisk process listening on 4569 UDP port on 0.0.0.0 El 5 jun. 2017 10:00, "Victor Villarreal" <mefhigos...@gmail.com> escribió: > Dear Thelma, > > Yes. Asterisk listen on port 4

Re: [asterisk-users] IAX port 4569

Dear Thelma, Yes. Asterisk listen on port 4569 UDP on default config. Please, look at the Asterisk logfile, for clues about your issue. Or enable IAX2 debug vía Asterisk CLI. Other ideas: * Check that your server firewall permit UDP port 4569 incoming traffic. * Run tcpdump over the network

Re: [asterisk-users] Automatically dial a number, then an extension

Hi John, I think we need to known how you play the audio to the customers, before we can help you. Are you using AMI? Or AGI maybe? Or Call files? What Asterisk version do you have? El 15 may. 2017 12:35, "Tech Support" escribió: > All; > > I have an application

Re: [asterisk-users] Hack attempt sequential config file read looking for valid files.

Hi David, Tim, Try to use Bail2Ban at last resort. Fail2Ban is a ractive approach, that permit the traffinc AND ONLY BLOCK them after certain level triggered. Use iptables to block the unused services faced to public networks like Internet. And configure these services properly, so they listen

Re: [asterisk-users] Hack attempt sequential config file read looking for valid files.

Hi, Jerry, I don't know what S.O. you have in the Server, but you can check the man page (https://linux.die.net/man/8/in.tftpd) for tftpd and use the options --address, so you can tell tftp from what interface/port this service listen request. >From the IP in your logs (69.64.57.18) the request

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

Hi Ernie, When one-way audio appear (no matters if there is a VPN or NAT server on the diagram) I simply : * Enable SIP debug on Asterisk server. Excecute 'sip set debug ip x.x.x.x' on Astrisk CLI, where x.x.x.x is the IP of the phone or SIP peer you want to debug. * Make a test call and

Re: [asterisk-users] Voicemail asking for login

Hi Darcy, What Pete think is correct. Maybe excecuting the following command at Asterisk console, will help you: asterisk> voicemail show users And you will get a list of all mailbox configured in your system. Search for the user with problems. Finally, in the Asterisk wiki you can find more

Re: [asterisk-users] PBX selection

Hi Speed Boy. I agree with Emiliano Vazquez too. Additionally, you and your team must think others points before choose Asterisk: * Asterisk is build to work on Linux. So your team needs some skills like setting up a basic Linux server (Debian, Centos, etc), donwload software from Internet,

Re: [asterisk-users] Commit dialplan & other config. in memory to disk?

Hi Nathan, Personally, I create a git repo on /etc/asterisk/ folder. With this approach, you not only can backup current dilplan on another location (another private server, or private repo on Bitbucket account). You can follow all the change history you made. Simply install git, then go to

Re: [asterisk-users] Manager events showing in CLI

Ok, Please, check your manager.conf and logger.conf for any clue about debugging options, into the Asterisk configuration directory. El 26 mar. 2017 14:52, "Telium Technical Support" escribió: > I tried that but it had no effect. Still see things like: > > > > [2017-03-26

Re: [asterisk-users] Manager events showing in CLI

Hi Ron, I don't remember right now, but you can try this command: cli> manager set debug off Cheers El 26 mar. 2017 3:58, "Telium Technical Support" escribió: I somehow cause AMI events to appear as output in the CLI, and I can’t figure out how to turn them off. Can

Re: [asterisk-users] Which tool to automatically restart Asterisk ?

Hi, Oliver. Maybe something like this (add this script to your crontab): 8<-- #!/bin/bash # # File: asterisk-watchdog.sh # Date: 2015.05.26 # Build:v1.0 # Brief:Secuencia para monitorizar procesos. # # ${PATH}:

Re: [asterisk-users] Turn on SIP debugging from DialPlan

Hi Derek, SIP debug can be enabled via Asterisk CLI (console) with the command: asterisk> sip set debug on If you know via what trunk your call goes, you can use the following command instead: asterisk> sip set debug ip xxx.xxx.xxx.xxx Where the xxx is the IP of your trunk (voip to pstn

Re: [asterisk-users] Disallow CALLS without registry

Hi Antony, Sory but I don't understand why your Asterisk accept anon calls with the conf you provide us. Maybe a full excerpt of an incoming call will help. Last, there exist dialplan like GROUP and GROUP_COUNT that permits you count the number of calls in a custom group fashion. El 10/2/2017

Re: [asterisk-users] Using g729 now that patents have expired

Hi Steve, I understand your question and your point, but I use the g729 codec from the link that Carlos share, for almost 6 years from Asterisk 1.4 to v13 without a single problem. So, sory but I don't share your phrase "from a lesser know web site". About your question, I did not known that

Re: [asterisk-users] Asterisk - Vtiger integration

Hi Alejandro, The documentation about your question is here: https://wiki.vtiger.com/vtiger6/index.php/PBX_Manager After a few seconds of read, I think that VTigerAsteriskConnector can run on a separate server than Asterisk PBX. VTigerAsteriskConnector connects to Asterisk via Asterisk Manager

Re: [asterisk-users] Polycom SoundStation IP 6000 does not register

Hi Yves, Maybe your switch put your Polycom inside a Voice VLAN, based on the MAC of the phone. Maybe with the snom this not happen because your switch don't see the MAC of the Snom as a "supperted IP Phone". 2016-12-21 13:59 GMT-03:00 Yves : > sorry... typo > the

Re: [asterisk-users] Cisco IP 8841 asterisk integration

With all the money you plan to invest in firmware, licenses, etc., you have bought a Grandstream IP phone or Yealink... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk

Re: [asterisk-users] iowait issues on CentOS 7

Hi Luca, IO delay maybe come from Hard Disk lattency. You can exec an "lsof " command to view what file asterisk proccess hold down when load spike. If there are some call recording, you can configure Asterisk to make it in a temp location, a RAM Disk in Linux. If you make hard usage of the

Re: [asterisk-users] iaxmodem errors.

Hi John! I'm not sure why are you using iaxmodem... I use it a few years ago with Asterisk 1.4 In Asterisk v11 fax is managed using res_fax. Please see https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_ReceiveFAX_res_fax You only need download, compile and install the spandsp

Re: [asterisk-users] Force hangup not working on stuck channel

Hi Carlos, Did you try with the following CLI command: CLI> channel request hangup CHANNEL_NAME ??? El nov. 3, 2016 1:16 PM, "Carlos Chavez" escribió: > I am unable to force a hangup on a channel that has been stuck for over > two days: > > IAX2/from-CD-11006

Re: [asterisk-users] Asterisk 13.11.2 unable to register on Centos 7 64bit

Ok. Please, note that 192.168.1.37 (I suspect) is the internal LAN address Of the Polycom hardphone. If this is true, then you have NAT issues. The REGISTER message are received by your PBX, but when respond, Asterisk send the next SIP message to the IP informed by the phone, that is the

Re: [asterisk-users] Asterisk 13.11.2 unable to register on Centos 7 64bit

Hi Motty, Please, set Verbose to 3 and Debug to 3 At Asterisk CLI. Then "sip set debug on". Now try to register again. At last, " sip de debug off". Examine tour console or full log file to find some clue ir send me back some trace. Cheers. El oct. 13, 2016 1:45 PM, "Motty Cruz"

Re: [asterisk-users] Asterisk 13.11.2, 13.11.1, 13.10.0 and certified-13.8-cert3 : freeze on 'sip reload'

Hi Jonas! Do you currently use any TLS technology in your Asterisk? Like SIP-TLS o pjSIP-TLS support ? If don't, please go to modules.conf and start disabling some modules that you don't use. For example, I can see some other modules related to calendars. If you don't use this, please disable

Re: [asterisk-users] send a call to moh until user is available

Hi Tux John, The behavior you need is cover in Asterisk within a Queue. 1. Create a new queue in queues.conf and assign as static member, the 4450 extension. 2. In your dialplan, you need to route the incomming calls to the new queue and pass as argument the timeout in seconds you want when

Re: [asterisk-users] asterisk-users Digest, Vol 147, Issue 5

Hi all ! Thanks for your feedback and sory for the delay. Respond: > Date: Mon, 3 Oct 2016 21:05:55 -0300 > From: Marcelo Terres > > I think that you need the dev files too. In Debian 8, the package is > libmysqlclient-dev. > > But Debian 8 uses libmysqlclient-18. Where did

[asterisk-users] Asterisk 11.23 with libmysqlclient20 on Debian 8

Hi List! I'm facing a problem while compiling Asterisk-11 on a Debian 8 server. The mysql-server version installed is 5.7 and come from the official mySQL community repo for Debian. After compile, install and execute Asterisk, the comman "lsof -p `pidof asterisk` | grep mysql" don't produce any

Re: [asterisk-users] how to read sip debug

Hi Thufir, The analysis of a SIP Debug depends on what the problem to be solved. If you experience problems with inbound calls from a SIP trunk or provider, you can type in Asterisk cli 'core set debug 3' and then 'sip set debug ip xxx.xxx.xxx.xxx' where xxx is the IP of your SIP provider or

[asterisk-users] Compiler errors when 'make asterisk' for D100 transcoding board

Hi List, I solve this issue and I want share it with this community. The sng-tc-linux-1.3.8 package don't compile across Certified Asterisk. Only normal Asterisk like 11.22.0 version. We have this version in production with the D100 board. Working. Cheers -- GnuPG Key ID: 0x39BCA9D8

[asterisk-users] Identify more demanding routine inside Asterisk

Hi List ! I'm facing a problem with the CPU consumption in Asterisk 11.22.0. I could decrease a lot of load, migrating both the astdb.sqlite3 and call recordings (with Monitor app) to a tmpfs mount in RAM (with noatime and nodiratime flags), manually spread each of the hardware interrupts

Re: [asterisk-users] nagios asterisk check SIP

On Fri, Jun 17, 2016 at 11:22:48AM +0200, Thomas wrote: > Iam loocking for an programm to check the SIP port of an Asterisk asterisk. > > Ome time ago I have used > #/usr/bin/sipsak > but it seemed that it is not working anymore? Hi Thomas, Maybe this links help you:

Re: [asterisk-users] queue_log - odbc vs AMI

Hi Marek, Here, we have an Asterisk v11-cert11 and found that there is NOT equal the CDR via AMI and CDR in Database. Please, check my gist: https://gist.github.com/MefhigosetH/89462e599a996dedf048f8d2b4e94d47 We have in use some custom dialplan variables in CDR (ie.: groupcount and rptqos),

Re: [asterisk-users] SPA112 flapping

Hi Mike, I would try the following: * If you can login through HTTP, check the uptime of the Cisco device. Make sure the device is not rebooting. * If you can, make a 'ping' from the PBX to the device and annotate milli-seconds of response. Then compare then to the default 'qualify' sip setting

[asterisk-users] Compiler errors when 'make asterisk' for D100 transcoding board

Hi there ! Someone in this wonderful list tried to install Sangoma transcoding board D100 on Asterisk v11 ? I followed each of the steps in the wiki [1], but when running 'make asterisk' receipt compilation errors about the absence of some header files [2]. I exchanged some mail with the

Re: [asterisk-users] Maximum number of PRI calls on 1 asterisk box (no HW echo)

Hi James, we did sucessfully run two TE410P with 8xE1 in a decent Intel P4-3200 machine with quite heavy line usage. No codec conversion course. I don't believe that there is a hard limit of E1s coded into Asterisk. But the maximum lines you can squeeze out of your specific hardware depends on

Re: [asterisk-users] Maximum number of PRI calls on 1 asterisk box (no HW echo)

2010/3/25 Steve Edwards asterisk@sedwards.com: On Thu, 25 Mar 2010, Tzafrir Cohen wrote: [snipping a lot of interesting technical and historical details] As you can see, there's actually a limit at the DAHDI level. DAHDI_MAX_SPANS, which is 128. Likewise there's DAHDI_MAX_CHANS which is

Re: [asterisk-users] Maximum number of PRI calls on 1 asterisk box (no HW echo)

2010/3/25 Zeeshan Zakaria zisha...@gmail.com: Tzafrir, so you have actually worked with more than 192 concurrent zap channels, which means more than 8 spans, on a single server, and can verify that it actually works without freezing asterisk. As I have written before - I did use 8 E1 in one

Re: [asterisk-users] G.711a or G.711u ???

2010/3/23 Alejandro Cabrera Obed aco1...@gmail.com: Dear all, I have an Asterisk SIP server in a LAN environment and I want your opinion in order to decide the use of an audio codec: What audio codec is better, G.711a or G.711u ??? Which suites to my LAN voip calls ??? That depends most on

Re: [asterisk-users] register = 2345:passw...@sip_proxy/1234

2010/3/19 tjoen tj...@dds.nl: register = tjoen:mypas...@sip_proxy/1234 [sip_proxy] type=peer host=ekiga.net I guess you need to register to the actual hostname, not the peers name. register = tjoen:mypas...@ekiga.net/1234 Chris --

Re: [asterisk-users] Digium TE4xx T1 Bonding

2010/3/11 Eric Wheeler aster...@ew.ewheeler.org: 4. Does anyone have a couple TE2xx or TE4xx cards that can test such a configuration? I would like to research their capability before purchasing a couple $1200 cards. Hi Eric, I have four spare TE411P but never used bonded T1 or T1 for data

Re: [asterisk-users] Hardware requirements question.

Yes, this machine will be enough for that task. Performance wise. The other good thing is that it is not very likely that someone will steal your PBX. As far as I remember it is a 7 rack unit box which weights approx. one metric ton. ;-) But remember - if anything dies in the box and you have to

Re: [asterisk-users] Playback in h extension

2010/3/5 Danny Nicholas da...@debsinc.com: Not possible.  H exten is called by a hangup. Well - sometimes not both parties hang up at the same time. ;-) If you want to play something to the originating party after die Dial()ed party hangs up use the option g in the Dial command to get more

Re: [asterisk-users] Do i need install Dahdi or libpri ?

2010/2/25 Zhang Shukun bit...@gmail.com: next ,i want to dial from asterisk to PSTN now. i have see the sample in the extensions.conf relevent to PSTN as follow: ; If you are freely delivering calls to the PSTN, list them here ; ;exten = _1256428,1,Dial(DAHDI/G2/${EXTEN:7}) ; Expose all

Re: [asterisk-users] 4 PCIe cards in one asterisk server

Not wit four - but two of them in a single core 3GHz machine worked flawlessly doing only switching and IVR without codec conversion. Many will suggest that you split your lines on two machines to to prevent a total loss when a machine fails. This will add some work on setup but maybe save you

Re: [asterisk-users] Question regarding digital card TE412p

Hi! Having two TE410P with heavy load in a Pentium4 3,2GHz system running Asterisk 1.2 was no problem. It did only IVR and bridging with no transcoding though. Chris 2009/12/14 das sandesh sandesh...@gmail.com: Hi, I was able to implement T122p one port PRI and was able to call out, but I am

Re: [asterisk-users] G729 Pass through

Hi! Are you sure you are getting Astrisk out of the media path? I guess reinvite must be allowed. Then it should work without transcoding licenses. Maybe you should take a look at the SIP DEBUG info to see what codec Asterisk is trying to negotiate with the trunk. You could disallow alaw and

Re: [asterisk-users] Sangoma A101DE with Dell PE 2850

2009/12/8 Ricardo Melendez rmelen...@utep.com.mx: First I see at sangoma page that A101DE is PCI-Express  (I think  x1 for the size of the connector) Yes, it is PCIe x1. There is an A101D wich is PCI(-X). for PCI Express one x4 lane width one x8 lane width I can connect the card to any

Re: [asterisk-users] automon = *1 one touch recording

2009/12/8 Joseph syscon...@gmail.com: After pressing *1 console is not showing anything indicating that the call is being recorded: -- Executing [...@office-closed:1] Playback(SIP/479-1270-680060b0, transfer) in new stack     -- SIP/479-1270-680060b0 Playing 'transfer' (language 'en')    

Re: [asterisk-users] Pbx-cards

mattias schrieb: But are not pbx card and modem the same? There are single FXO cards (to connect to a analogue line) that are basically PCI modem with a special driver. But the chances that your modem is compatible to this one specific type is very little. Chris

Re: [asterisk-users] Asterisk 1.4 and Fax

2009/11/2 Doug Lytle supp...@drdos.info Dan Journo wrote: I need to get it up and running before we can put in the order to transfer the fixed line number over to SIP. Faxing over SIP is never a good idea. And why would that be? I think that faxing over SIP using T.38 is a fantastic

Re: [asterisk-users] Asterisk 1.4 and Fax

2009/11/2 Doug Lytle supp...@drdos.info Christian Victor wrote: 2009/11/2 Doug Lytle supp...@drdos.info mailto:supp...@drdos.info Faxing over SIP is never a good idea. And why would that be? I think that faxing over SIP using T.38 is a fantastic idea. As far as I know, T.38

Re: [asterisk-users] GSM cellphone as cheap gateway?

Olivier schrieb: 2009/9/21 Vijay Gandhi vi...@gandhiinfotech.com There are FTC’s available, What is it (a FTC) ? a cable ? Any pointer to that (Google is helpless)? ? My guess would be fixed to cell or FX to cell adapter. Chris ___ --

Re: [asterisk-users] All the four lights blinking

2009/9/11 ABBAS SHAKEEL shakeel.abbas@gmail.com Thanks you very much Kevin.I will try it by connecting one end of Ethernet cable to one slot and other to second slot . Configuring one as pri_net and the other as pri_cpe. I will provide you feed on monday either i succed or not

Re: [asterisk-users] Best ISDN BRI solutions?

2009/8/6 Alex Balashov abalas...@evaristesys.com Sure it is. Just get a media gateway that does T.38 - and does it relatively well. Wich the Pattons do quite well afaik. Chris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com

Re: [asterisk-users] [asterisk]q: asterisk 1.6.1 install

tom schrieb: hi just donwloaded the 1.6.1 branch and made configure install. so far so good. after staerting asterisk with: asterisk -cr Could not load features.conf == Registered application 'ParkedCall' == Registered application 'Park' == Manager registered action

Re: [asterisk-users] Different codecs for reading and writing

Philipp Kempgen schrieb: Elliot Murdock schrieb: I am wondering how the Asterisk community has been working on solutions to deal with the asymmetric quality of ADSL. Voip is becoming popular and a bottleneck does exists on the ADSL upload side. One participant's upload is the

Re: [asterisk-users] Announcement: Howler-optimised G.729A Solution for Asterisk

Jeff LaCoursiere schrieb: I have a question in to them about how that floating licensing works, though. Does that mean that with every call a license check must be made? I don't see how it would work otherwise, and that means my whole business - every call - is dependant on their license

Re: [asterisk-users] Question about core CDR system for multilpe servers

Danny Nicholas schrieb: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Miguel Molina Sent: Thursday, June 04, 2009 11:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

Re: [asterisk-users] IP phone recommendation

Right! Whatever somebody likes more! I just say that the Snoms look better at the side of my Mac. Wich is of course by far the superiour system. ;-) Chris John Novack schrieb: Hasn't this religious argument/discussion gone on long enough?? zoach...@securax.org wrote: I personally find

Re: [asterisk-users] FritzBox 7270

Manoj Panicker - FOES schrieb: However I can always call any one pre-configured PSTN number using the call forwarding feature, however I should be able to use my sogtphone and dial a PSTN number using the integration which is not happening today. As far as I know the FritzBox only supports

Re: [asterisk-users] FritzBox 7270

2009/5/24 Manoj Panicker - FOES manoj.panic...@emirates.com Kare, Thanks much appreciated. It connected as soon as I created a SIP account. However I must try and figure out as how to get this box use IAX2. Are you sure the FritzBox actually supports IAX2? As far as I remember it does

Re: [asterisk-users] FOR IMMEDIATE RELEASE: NEW CHANNEL DRIVERFORASTERISK RELEASED TODAY

Duuh guys - it's so easy. Ever thought of simply compressing the compressed data AGAIN??? Do that the necessary amount of times and - tadaa - it's done. Chris 2009/4/1 Brent Davidson br...@texascountrytitle.com Cary Fitch wrote: It uses proprietary EDC. (Extreme Data Compression) The 140

Re: [asterisk-users] Need to find small footprint asterisk platform

2009/3/30 Peer Oliver Schmidt po...@theinternet.de The Horst-Box Professional has a lot of problems in the ADSL area (like stopping transfers after a dozen or so megabytes for example), and I have had lots of needs to hard-reboot the box, after enabling VoIP functionality. Well - I never

Re: [asterisk-users] Need to find small footprint asterisk platform

Here in germany D-Link sells a device called the Horst-Box Professional wich is a ADSL modem/router with WiFi and an integrated embedded asterisk platform with 1xBRI in, 1xBRI out and 3xFXS if my mind serves me right. Size is about 180x250x50mm. Its been around for some years so maybe it is

Re: [asterisk-users] Help: RED alarm on Wildcard TE122 card

Andreas-Johann Ulvestad schrieb: When inserting the cable going into TE122 into an ISDN phone, the phone works perfectly. That should not happen with an E1 line as your phone normally has a BRI (S0) connector with only two b-channels. Seems that your line is configured ar BRI and not PRI.

[asterisk-users] SIP trunk with 250 lines

Hi! A customer of mine wants to connect an asterisk system with 240 to 480 lines to a PSTN switch. To save the costs for E1 cards and the corresponding E1 mainlines he wants to connect the system to the switch by a SIP trunk. Phones will be connected to the server through the same SIP trunk as

Re: [asterisk-users] SIP trunk with 250 lines

2009/3/24 Cary Fitch ca...@usawide.net First Issue to be addressed is how many simultaneous calls and bandwidth availability. Number of “lines” (numbers) is not a limitation in it self unless they are all in use. Sorry for being a bit unclear in this point. What I meant was 240 to 480

Re: [asterisk-users] SIP trunk with 250 lines

2009/3/24 Danny Nicholas da...@debsinc.com Here are a few “look outs”; Using conference rooms will increase your bandwidth requirements. On board Network controllers will affect performance in this “high-use” scenario. 250 simultaneous calls will use about 7.5Mb of bandwidth depending on

Re: [asterisk-users] SIP trunk with 250 lines

2009/3/24 Grygoriy Dobrovolskyy megaho...@gmail.com If the switch is fine why not ? But i wander why kind if switch is that 240-480 fxo ? ;) Sounds like a big overkill. And i dont see a problem with asterisk, if not too much transcoding involved and with the right hardware. It's an ISDN

Re: [asterisk-users] SIP trunk with 250 lines

2009/3/24 Danny Nicholas da...@debsinc.com I use a Dell with the 1Gb Ethernet on board, but had to clock it down to 100 Mhz because * has an issue with Dell on board Ethernet. Ah - good to know. I think we will use SUN machines. But I'll keep that in mind. Chris

Re: [asterisk-users] conference and wifi phones

2009/3/24 Steve Gladden aster...@michiganbroadband.com I REALLY like the Snom M3 DECT SIP base. Yeah - it's such a pitty that you always have to buy it bundled with one of these crappy handsets. Or is there a way to get only the base that I don't know? Chris

Re: [asterisk-users] Magic SIP Phone

Maybe the Siemens DE380 IP R could help you. It's a brand new IP phone with an integrated router. Chris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

Re: [asterisk-users] Magic SIP Phone

grandstream gxp-2000 works fine for that. depending on firmware rev its two ports are either a switch or router. Grandstream removed this functionality in recent softwware upgrades - I guess they needed the code space for other things. Why would you want a router in the phone and not let

Re: [asterisk-users] Portech MV3770 Caller-ID

2009/3/11 Håkan Källberg h...@simulina.se Hello! Does anyone of you have Caller Presentation working in the other direction?? My mv370 is working well, execpt the Caller ID on outgoing GSM calls. This works with the SIM card/Provider I am using if I put the SIM card in a telephone, but not

Re: [asterisk-users] Portech MV3770 Caller-ID

2009/3/10 Sasa s...@shoponweb.it Hi, I have modified in Mobile/Setting the parameter SIP From from tel/user to tel/tel and now I view the correct incoming number. Thanks. Glad I could help. It took me nearly a month to figure that out. ;-) Chris

Re: [asterisk-users] Portech MV3770 Caller-ID

2009/3/9 Sasa s...@shoponweb.it Hi, I have a problem with Asterisk-1.4.22 (with TB 2.6.2) Portech MV-370, my problem is that when arrived an external call I don't view (on my internal phone) the phone number but I have the number extension that is ... ..now what parameter can I modify

Re: [asterisk-users] Bounty- CDR Bug Fix

2009/3/4 Atis Lezdins a...@iq-labs.net Bottle of Riga Black Balsam (45%), just have to figure out a way to send it :) Balsam??? By mail? Doesn't that count as liquid explosive? ;-) Chris ___ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] call file concurrency

2009/2/27 Bill Michaelson b...@cosi.com Is there a convenient way to limit the number of call files (outgoing directory) that are processed concurrently? Afaik only by limiting the number of call files in the directory. ___ -- Bandwidth and

Re: [asterisk-users] GSM codec is a good choice ???

2009/2/25 Alejandro Cabrera Obed aco1...@gmail.com But in my case, I don't need trascoding because every chanel is in GSM and voicemail has gsm sound files. And for the moment, my Asterisk is not connected to the PSTN, so there is no trascoding gsm-to-PCM or to analog. So I think gsm is a

Re: [asterisk-users] Quiet 24 port POE gig switch

2009/2/2 Singer XJ Wang w...@pythian.com [snipped] You can do that by using fans other than the tiny, whiney, 40mm fans that vibrate at 6000 to 18,000 Hz. A couple of 80 or 120 mm muffin fans at the back or front, pushing air in (hence the deep dimensions), but the top and bottom would need

Re: [asterisk-users] Attacking DECT

2009/1/1 Olivier oza-4...@myamail.com To attack DECT equipments, a ComOnAir module was used. This module is a PCMCIA addon which provides DECT connectivity. I don't think this module is available or manufactured anymore. So it seems difficult for anyone to reproduce this DECT attack. Or

Re: [asterisk-users] whisper time remaining

on how you bridge the channels. It is clear is that, from a programmer point of view, I would have preferred to call it it an Asterisk Manager command :-) Kind Regards, Victor Alexander Lopez wrote: If you know the channel that you need to ‘whisper to’, You could always create a call via

[asterisk-users] whisper time remaining

via Asterisk Manager. Does anyone know how to implement this? Thanks a lot. Regards, Victor ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http

Re: [asterisk-users] whisper time remaining

the call. Victor Alvarez wrote: Hello everyone, I'm trying to find out a way to whisper the time remaining for a prepaid application on a established channel. Unfortunately I think there is a lack of PlayBack/Background commands which can be applied on a working channel as well

Re: [asterisk-users] Call files

Steven Howes schrieb: Have created a system that involves using call files in the outgoing spool folder. On some occasions it retries which is fine is there any way to view calls waiting retries from the CLI? Using 1.4 btw. Have googled to no avail (although it is near the end of the

Re: [asterisk-users] PoE switch recommendations?

Hi Ken, we are quite satisfied with Linksys SRW248G4P. 48 port PoE, 4 GB uplinks and 2 GBIC slots. VLAN, QoS and all the like is on board. Around US$600 I guess. Only drawback in my opinion is that they are loud like a starting airplane. You definately don't want them next to your desk. ;-)

[asterisk-users] Pressing 0 to get an external line

Hi Asterisk users! I have a little problem with an Asterisk 1.4.22 installation for a customer. The PBX is connected to an E1 line and we have a few snom 300 attached to it. The goal is to emulate traditional german PBX behaviour wich is the play a stuttered internal dialtone after pickup and

Re: [asterisk-users] Pri to sip interfaces

Tom Moore schrieb: What are your suggestions to people who have pbx systems that interface with the world over pri and want to convert them to sip interfaces so that they can use sip trunking? I'd go for a Patton SmartNode. See www.patton.com - they have SIP gateways up to 4 T1/E1. Christian

Re: [asterisk-users] PRI Splitter

2008/8/31 Olivier [EMAIL PROTECTED] What happens if the PC supporting this card is powered off ? It is powered over USB from the main (internal USB) and backup (external USB) server. If one of the power fails it will switch to the other server. If both servers power fail you have a problem

Re: [asterisk-users] PRI Splitter

2008/9/1 Karl Fife [EMAIL PROTECTED] It is powered over USB from the main (internal USB) and backup (external USB) server. If one of the power fails it will switch to the other server. If both servers power fail you have a problem anyway. ;-) This is incorrect. According to Jim Rhodes

Re: [asterisk-users] IVR question

I'm setting up my IVR system, how can I register in a mysql database the IVR menus accessed by the clients ? Just use the MYSQL-Functions in the dialplan to write the menues name (and datetime maybe) in a table. To access MYSQL from the dialplan you need to have the asterisk-addons.

Re: [asterisk-users] BRI AND DATA connection

Anton schrieb: Does anyone tried BRI with asterisk for DATA transfer? My customer wants BRI connection, but he wants it for the data, and I have to bring connection to his office, so I see the connection as follows: E1-(CORE_ASTERISK)-(IAX2)-(EDGE_ASTERISK)-BRI - so Why would

Re: [asterisk-users] Website callback

I don't know if there is something like that prebuilt. But is seems to be quite easy. Push the call events in the database, let a cron run ever minute and create a .call file for evry call thet is due. The alternative is to not use a database and create a .call file with a future date/time. Afaik

[asterisk-users] Asterisk 1.4.20.1 problems

Hi! I just upgraded my Asterisk server from 1.4.6 to 1.4.21 and now I experience some strange behaviour. 1) The Asterisk CLI (asterisk -r) stops responding after some minutes. I cant CTRL-C or exit the CLI anymore and no activity is shown. Just like if the connection is interrupted. 2) When I

[asterisk-users] CLIR missing in MySQL CDR records

Hello! I just encountered a strange thing in my mysql cdr records. From a certain date on Asterisk (1.4.6) stopped to populate the CLIR and SCR flieds in the cdr table. As far as I know no changes happened to the system on that date and until then CLIR are recorded properly. The CLIR is still

[asterisk-users] How to Configure 1.4.17 to Store CDR's in PostgreSQL

I'm having a heck of a time saving my CDR's into a PostgreSQL database. I've installed PostgreSQL on a remote server and it is successfully storing voicemail messages but I cannot get the 1.4.17 system to store CDR records there. Has anyone successfully configured a 1.4 system to store CDR's in a

Re: [asterisk-users] R2-Unicall Asterisk as CPE and as CO

.. in the first place.. that pbx is not mine, I didnt configured it and I cant even touch it, Im just putting asterisk in between right now. Im gonna try that.. Thanks for the help! -- Regards.. Victor Toofic ___ -- Bandwidth and Colocation Provided

  1   2   3   >