[asterisk-users] CDR record for call originated from CLI originate
hello List, i am in a situation where i cannot get cdr records for call originated from CLI , i am not able to get when i used application or extension. is there any solution regarding this ,i working since last 3 days onto this. regards Dhaval -- _ -- 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 list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR record for call originated from CLI originate
Hi Dhaval, I 'm in the almost same situation. I've already post a issue with asterisk. https://issues.asterisk.org/view.php?id=17826 Is you only use an originate and not an originate en then redial maybe this link helps you further. https://issues.asterisk.org/view.php?id=17592nbn=16#bugnotes Regards, Arjan Kroon Van: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Namens DHAVAL INDRODIYA Verzonden: 05-10-2010 09:09 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: [asterisk-users] CDR record for call originated from CLI originate hello List, i am in a situation where i cannot get cdr records for call originated from CLI , i am not able to get when i used application or extension. is there any solution regarding this ,i working since last 3 days onto this. regards Dhaval -- _ -- 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 list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISDN - Busy signal on 3rd call
Still I am also facing the call disconnection when there is a third call. I am using Netmod BRI router and the output of the BRI router lines are connected to FXO ports in Asterisk. Where in Asterisk I am facing the call disconnection when there is a third call.. On Tue, Sep 28, 2010 at 4:22 PM, Paulo Santos paulo.r.san...@sapo.ptwrote: Hello, Following my first mail about this issue [1], I think I know now what the problem is. When I have both lines being used and a third call comes in, the person calling doesn't get a busy tone, he gets something like line unavailable. I've been debugging mISDN and I think the reason is because asterisk is sending the release cause as 0. P[ 3] -- channel:0 mode:TE cause:0 ocause:0 rad: cad: The request from the telephone company's switch seems correct, a SETUP message (if 08 is Q.931, 05 is SETUP). 02 ff 03 08 01 04 05 a1 04 03 80 90 a3 18 01 80 6c 0b 01 83 39 31 36 33 39 31 37 34 32 70 03 c1 38 34 I've changed misdn.conf so it sends a release cause as 17 (user busy), but I get the same behaviour - cause:0 ocause:0. Anyone knows how can I force asterisk to send cause 16 or 17 in this situation? Thanks in advance. Best regards, Paulo Santos misdn.conf: http://pastebin.com/FmgECqkU misdn debug: http://pastebin.com/Tg6wPKBD [1] http://www.mail-archive.com/asterisk-users@lists.digium.com/msg244330.html -- _ -- 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 list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thank you with regards, Gopalakrishnan A.N, -- _ -- 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 list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Implementing more than one asterisk instance in the same hardware machine?
Hi All; Did anyone try to implement (installation and configuration and running) for more than one asterisk instance (two or three instances), where each asterisk instance to work on a difference IP than the other where the server already has more than one IP address. We need to implement this situation because in case we need to do testing for any scenario of configuration, then other instances will not be effected. Is it possible without WMWare? Regards Bilal -- _ -- 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 list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Implementing more than one asterisk instance in the same hardware machine?
On Tue, 5 Oct 2010, bilal ghayyad wrote: Hi All; Did anyone try to implement (installation and configuration and running) for more than one asterisk instance (two or three instances), where each asterisk instance to work on a difference IP than the other where the server already has more than one IP address. Yes. We need to implement this situation because in case we need to do testing for any scenario of configuration, then other instances will not be effected. Is it possible without WMWare? Yes. I'm using LXC - which is a softer form of virtualisation - aka Containers. There's not much documentation on it though, but a good Linux sysadmin ought to be able to pick it up though. Gordon -- _ -- 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 list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR record for call originated from CLI originate
Hi Arjan, i am able to solve this problem after adding this patch and adding unanswered=yes onto cdr.conf https://issues.asterisk.org/file_download.php?file_id=24431type=bug regards Dhaval On Tue, Oct 5, 2010 at 1:12 PM, Arjan Kroon | Mobillion arjan.kr...@mobillion.nl wrote: Hi Dhaval, I ‘m in the almost same situation. I’ve already post a issue with asterisk. https://issues.asterisk.org/view.php?id=17826 Is you only use an originate and not an originate en then redial maybe this link helps you further. https://issues.asterisk.org/view.php?id=17592nbn=16#bugnotes Regards, Arjan Kroon *Van:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *Namens *DHAVAL INDRODIYA *Verzonden:* 05-10-2010 09:09 *Aan:* Asterisk Users Mailing List - Non-Commercial Discussion *Onderwerp:* [asterisk-users] CDR record for call originated from CLI originate hello List, i am in a situation where i cannot get cdr records for call originated from CLI , i am not able to get when i used application or extension. is there any solution regarding this ,i working since last 3 days onto this. regards Dhaval -- _ -- 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 list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Implementing more than one asterisk instance in the same hardware machine?
Hi We can run multiple instance of asterisk in same box with different IP and port. U need to install asterisk in different location eg: 1: /home/asterisk1/ 2 , : /home/asterisk2 ,and run both from that path , listen ip and port should be different. command to run: $ /home/asterisk1/usr/sbin/asterisk -g for first asterisk $ /home/asterisk2/usr/sbin/asterisk -g for second asterisk Thanks Nikhil On 10/05/2010 03:42 PM, bilal ghayyad wrote: Hi All; Did anyone try to implement (installation and configuration and running) for more than one asterisk instance (two or three instances), where each asterisk instance to work on a difference IP than the other where the server already has more than one IP address. We need to implement this situation because in case we need to do testing for any scenario of configuration, then other instances will not be effected. Is it possible without WMWare? Regards Bilal -- _ -- 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 list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Implementing more than one asterisk instance in the same hardware machine?
On Tue, 5 Oct 2010, Nikhil wrote: Hi We can run multiple instance of asterisk in same box with different IP and port. U need to install asterisk in different location eg: 1: /home/asterisk1/ 2 , : /home/asterisk2 ,and run both from that path , listen ip and port should be different. command to run: $ /home/asterisk1/usr/sbin/asterisk -g for first asterisk $ /home/asterisk2/usr/sbin/asterisk -g for second asterisk I did that before I moved to LXC, but you can't use the standard port 5060 for all instances, only one - might be OK in testing, but you can't realistically expect punters to change the port their equipment uses... Gordon Thanks Nikhil On 10/05/2010 03:42 PM, bilal ghayyad wrote: Hi All; Did anyone try to implement (installation and configuration and running) for more than one asterisk instance (two or three instances), where each asterisk instance to work on a difference IP than the other where the server already has more than one IP address. We need to implement this situation because in case we need to do testing for any scenario of configuration, then other instances will not be effected. Is it possible without WMWare? Regards Bilal -- _ -- 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 list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Implementing more than one asterisk instance in the same hardware machine?
On 5 Oct 2010, at 15:13, Gordon Henderson wrote: $ /home/asterisk1/usr/sbin/asterisk -g for first asterisk $ /home/asterisk2/usr/sbin/asterisk -g for second asterisk I did that before I moved to LXC, but you can't use the standard port 5060 for all instances, only one - might be OK in testing, but you can't realistically expect punters to change the port their equipment uses... More than one IP on the box. Change the bind address.. Easy, no? Steve -- _ -- 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 list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Registering Multiple Trunks to Service Provider
Any pointers on this one?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ujjval Karihaloo
Sent: Monday, October 04, 2010 12:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Registering Multiple Trunks to Service Provider
We have multiple entries like the one below in our users.conf file... where the
username. Contact and secret changes for different customers and we register on
their behalf to the Service Provider.
For the trunk below: when the call is placed out, Asterisk (1.4.18) sends the
username of abc.com in the MD5 Auth .which obviously does not match the
trunk setup for this Customer with our Service Provider (username below is
3035551122)
I don't see anywhere any config file the username = abc.com where could the
asterisk be picking it up from?
We have more than 10 such entries (all with same host = provider.sip.com value)
and when as INVITE is challenged, the Asterisk does match the correct trunk and
seems to send out correct Auth credentials...but not the one below..
[trunk_1]
;register to SP
allow = ulaw
;context = test
dialformat = ${EXTEN:1}
canreinvite = no
hasexten = no
hasiax = no
hassip = yes
host = provider.sip.com
insecure = very
port = 5060
registeriax = no
registersip = yes
trunkname = test
trunkstyle = customvoip
username = 3035551122
disallow = gsm,g726,alaw
contact = 3035551122
secret = x
--
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Re: [asterisk-users] Implementing more than one asterisk instance in the same hardware machine?
On Tue, 5 Oct 2010, Steve Howes wrote: On 5 Oct 2010, at 15:13, Gordon Henderson wrote: $ /home/asterisk1/usr/sbin/asterisk -g for first asterisk $ /home/asterisk2/usr/sbin/asterisk -g for second asterisk I did that before I moved to LXC, but you can't use the standard port 5060 for all instances, only one - might be OK in testing, but you can't realistically expect punters to change the port their equipment uses... More than one IP on the box. Change the bind address.. Easy, no? Ah, yes. Hm. I obviously wasn't having a good time when I looked at it but I was already running that and LXC at the same time. I'm using LXC in other places too (generic web hosting), so I probably didn't put too much effort into the first way! Gordon -- _ -- 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 list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Implementing more than one asterisk instance in the same hardware machine?
You can use proxmox from proxmox.com. I am using it for the same reason you want to use it. I have been testing it for some time now and it works great. Proxmox is an excellent hypervisor and it is free. Easy to install and simple to setup. Install it drom its ISO. Then you can download a OenVZ CentOS 5.2 instance for it from proxmox website, install it, give it an IP address and you have your server ready. Install on it asterisk as you would on any other system. I have detailed instructions for it on my blog, which I documented when I was setting up asterisk from scratch on a CentOS instance on proxmox. Once you have asterisk all setup, you can simply copy/paste the folder with virtual machine instance using a new name, and you have a second copy of your asterisk setup. Assign it a different IP address. I created 7 copies of my main setup, each with its own IP address. Proxmox also gives you option for hardware level virtulization, called KVM. I haven't tried it. With only OpenVZ you shall not be able to use zaptel/dahdi hardware though, and I don't know if KVM allows for it. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-10-05 10:57 AM, Steve Howes steve-li...@geekinter.net wrote: On 5 Oct 2010, at 15:13, Gordon Henderson wrote: $ /home/asterisk1/usr/sbin/asterisk -g for firs... More than one IP on the box. Change the bind address.. Easy, no? Steve -- _ -- Bandwidth and Colocati... -- _ -- 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 list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP flood attacK
On 03/10/10 21:19, Greg Saunders wrote: Hello all. I was recently the victim of a SIP flood attack. I'm wondering what is the best method to prevent such things in the future. Many thanks Greg do one of the following: - use deny permit lines in sip.conf /or iax.conf to restrict any remote Registrations from known IP address ranges only. Or use iptables rules to do something similar. - use a log scanning tool such as fail2ban or ossec which can react on multiple registration fails and block ip addresses in iptables - enforce strict password policy on all users on the system I think simply relying on alwaysauthreject is very dangerous as it's only a matter of time before the attackers catch on to this and carry on attacking regardless. Sure there's less chance of them getting a correct username/secret combination but in the meantime, the register attempts are practically a DoS attack. Plus that setting further breaks the SIP RFC. I also think that assuming that the attackers will eventually get in one way or another is wise. So put in place appropriate measures to limit the damage they can do (daily spend limits with SIP providers, blocking international and/or premium rate numbers etc...). cheers, Paul. -- _ -- 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 list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] minimum card for dahdi timing source ?
On 02/10/10 17:24, mancyb...@gmail.com wrote: Hi All, for a vicidial server which uses only voip, which is the minimum telephony card which would provide the required clock timing source for conferences to work properly ? Maybe the Digium TDM410PLF card without any daughter card would do the job ? Thank you very much for supporting. Have a nice week-end, Mike The cheapest device I've seen to provide a hardware timing source is the USB voice sync tool from Sangoma: http://www.sangoma.com/products/hardware_products/specialty_tools.html I know of at least one person using this with Vicidial successfully. cheers, Paul. -- _ -- 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 list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7941 localized menus with SIP firmware
http://www.ip-phone-forum.de/showthread.php?t=188877 ~ Andrew lathama Latham lath...@gmail.com * Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software * Learn more about Linux http://en.wikipedia.org/wiki/Linux * Learn more about Tux http://en.wikipedia.org/wiki/Tux On Tue, Jul 3, 2007 at 1:24 PM, Olivier oza-4...@myamail.com wrote: Any reply ? 2007/7/1, Olivier oza-4...@myamail.com: Thanks everybody for your input. Let me summarize localization process : 1. Buring boot, phones download from TFTP server an xml or older .cfg file in which a localization parameter is set. 2. When this parameter is read, phones will then ask CCM or Asterisk or another server to send localized button templates using SIP messages. 3. When SIP messages are received, phones will then localize buttons and menus. 4. Whenever an incoming or outgoing call is processed, phones will no further ask any localization data from CCM or Asterisk or anything as localization was fully done during initialization process. A. Is this correct ? Specifically, shall I understand that if present initialization process is extended to include localization, there will be no need to change Asterisk SIP channel to fully support localized Cisco phones ? B. What are the SIP messages that trigger and reply to localization demands ? Is it possible to tailor dialplan so that these Cisco specific messages are treated without modifying chan-sip ? Cheers ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to test BIG traffic through DAHDI/WANPIPEinterfaces
Hello my friend Ingmar, I would like to know the cable you used? how was the connection? i'm using this one: http://wiki.sangoma.com/Pinouts#A108 Loop Back Is this ok? what should i do my friend, my problems are understand the fisicall connection :( Best Regards!!! 2010/9/24 Ingmar Steen i.st...@teleknowledge.nl Hi DD, We usually use loopback cables and use the open source SIP test tool “SIPp” to initiate SIP calls that are sent from one group of 4 ports to another group of 4 ports. Met vriendelijke groet, Ingmar Steen Teleknowledge *Van:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *Namens *Danny Dias *Verzonden:* vrijdag 24 september 2010 11:05 *Aan:* Asterisk Users Mailing List - Non-Commercial Discussion *Onderwerp:* [asterisk-users] How to test BIG traffic through DAHDI/WANPIPEinterfaces Hello Community, I need to test or simulate many calls through dahdi/wanpipe, i have a Sangoma A108D, and i need to test the stability of the card/drivers/firmwares with a test environment, do you think is possible? What should i do? using some loopback cable maybe? Thanks in advance DD -- _ -- 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 list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] meetme don't play conf-invalid if room does not exist
Has anyone a solution for me - with Meetme(,Ms)asterisk plays conf-invalid if a room not exist - with Meetme(123,Ms) asterisk plays not conf-invalid if the room not exist and asterisk hangup I am happy about any proposal. Thanks Daniel -- _ -- 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 list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to test BIG traffic through DAHDI/WANPIPEinterfaces
On Tue, Oct 5, 2010 at 1:02 PM, Danny Dias ing.diasda...@gmail.com wrote: Hello my friend Ingmar, I would like to know the cable you used? how was the connection? i'm using this one: http://wiki.sangoma.com/Pinouts#A108 Loop Back Is this ok? what should i do my friend, my problems are understand the fisicall connection :( Best Regards!!! 2010/9/24 Ingmar Steen i.st...@teleknowledge.nl Hi DD, We usually use loopback cables and use the open source SIP test tool “SIPp” to initiate SIP calls that are sent from one group of 4 ports to another group of 4 ports. Met vriendelijke groet, Ingmar Steen Teleknowledge *Van:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *Namens *Danny Dias *Verzonden:* vrijdag 24 september 2010 11:05 *Aan:* Asterisk Users Mailing List - Non-Commercial Discussion *Onderwerp:* [asterisk-users] How to test BIG traffic through DAHDI/WANPIPEinterfaces Hello Community, I need to test or simulate many calls through dahdi/wanpipe, i have a Sangoma A108D, and i need to test the stability of the card/drivers/firmwares with a test environment, do you think is possible? What should i do? using some loopback cable maybe? Thanks in advance DD I set up two machines with T1 interfaces, and connected the two with an appropriate t1 cable. One was acting as a network (master), the other as a subscriber (slave) (for timing). wrote two dialplans, one for each machine, that would answer an incoming call on one dahdi line, and call to the next numbered line on the other machine. The other machine was similarly outfit. I'd define the extension for the first line on the t1, and call it with any phone you desire. That call will cascade into 23 separate interlinked calls. If you are clever, the last call in should dial another real phone you have on-hand. You get the picture... right? Phone A dials the exten to call the first exten on the other machine. The dialplan should use the first channel on the t1 to place a call to the first exten on the other machine. On the other machine, the incoming call on channel 1 is answered, and then a dial to the second extension on the first machine, over the 2nd t1 channel. The first machine answers, and uses the 3rd channel to call the other machine and so on till all channels are being used. The last exten answers and dials a phone (dahdi or SIP, no matter) that you pick up. Such a looped call should probably be awful, but it's going thru 23 t1 channels! If you have two t1 interaces in a single card (or two cards), then you do this on one machine. Another approach: set up equal numbers of FZO and FXS lines, and similarly loop s single call thru all the channels.This would require just one machine. Other approaches would involve running multiple threads to call an extension and then hang up and repeating this over and over again on all channels to ascertain the load placed just by call setup and tear-down. This kind of load is different than when all lines are just shoveling data back and forth. Watch your load averages, your %cpu, your swap, etc, as the tests are in full swing. murf -- _ -- 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 list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Steve Murphy ParseTree Corp -- _ -- 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 list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk CDR Radius error
Hello, I'm trying to configure Asterisk with Radius cdr support. Asterisk version 1.6.2.13 Server Radius: Freeradius version 1.X Radius client: radiusclient-ng version 0.5.5 With the Asterisk core debug on 1 when a call terminate, on the console appear this error: Unable to create RADIUS record. CDR not recorded! My cdr.conf is: [radius] usegmtime=yes; log date/time in GMT loguniqueid=yes ; log uniqueid loguserfield=yes ; log user field radiuscfg=/etc/radiusclient-ng/radiusclient.conf When I load the cdr_radius module no error appear. Any suggestion? Regards - Andrea -- _ -- 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 list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] minimum card for dahdi timing source ?
On Tue, 05 Oct 2010 17:30:49 +0100 Paul Hayes p...@provu.co.uk wrote: On 02/10/10 17:24, mancyb...@gmail.com wrote: Hi All, for a vicidial server which uses only voip, which is the minimum telephony card which would provide the required clock timing source for conferences to work properly ? Maybe the Digium TDM410PLF card without any daughter card would do the job ? Thank you very much for supporting. Have a nice week-end, Mike The cheapest device I've seen to provide a hardware timing source is the USB voice sync tool from Sangoma: http://www.sangoma.com/products/hardware_products/specialty_tools.html I know of at least one person using this with Vicidial successfully. cheers, Paul. Hi Paul, very interesting thank you. -- _ -- 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 list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk sharing a line with POTS handsets: how to interoperate cleanly?
I now have an OpenVox A400P and it is working well. Thanks to Ade Vickers for the recommendation, which I second. However, I need to make a slow transition between a conventional multiple-extension setup and a full VoIP network on these premises. So at the moment the Asterisk box shares the PSTN connection with several conventional analogue handsets. The desired result for an incoming call is that the Asterisk server will wait N seconds before answering (which I can arrange easily enough), and if the call has been answered on one of the handsets by that time the Asterisk server should ignore it completely. Otherwise it should start checking CLID, prompting for extensions, and other good stuff, which again I know how to do. What is a good approach to making sure the Asterisk server doesn't pick up a call that has been answered elsewhere? (Ideally in pure dialplan, but a perl AGI would also do.) R -- _ -- 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 list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco SIP 8.5 and 9.0 Issues
Hi list, I was wondering if anyone had any solution to either one of two issues I'm having: I have a cisco 7962G with the latest (from cisco) 8.5(4) SIP Firmware, it works very well for the most part, but after less then a week of heavy usage, eventually the phone gets into a state where it won't accept or let you place any more calls, the screen flashes no free lines available or something along those lines. (power cycle fixes this). So my preferred solution would be to upgrade to the v9.0(3) firmware, but when that's loaded, the phone won't register with Asterisk anymore, does anyone know if I need to adjust my .cnf.xml file, or is it a bug of some sort? Thanks for any input, -- Gerard Saraber Network Admin. Rarcoa, Inc -- _ -- 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 list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting up realtime config.
Hi Mike, Which is the real name for this peer? If you want look the configuration peer on Asterisk console try: CLI sip show peer accountname load To register to this account on Ekiga... accountname is the name of the extensions you have to configure. BR - Andrea -- _ -- 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 list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting up realtime config.
Hi Mike, Which is the real name for this peer? If you want look the configuration peer on Asterisk console try: CLI sip show peer accountname load To register to this account on Ekiga... accountname is the name of the extensions you have to configure. BR - Andrea -- _ -- 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 list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk CDR Radius error
On 5 October 2010 21:16, bakko asannu...@gmail.com wrote: Hello, I'm trying to configure Asterisk with Radius cdr support. Asterisk version 1.6.2.13 Server Radius: Freeradius version 1.X Radius client: radiusclient-ng version 0.5.5 With the Asterisk core debug on 1 when a call terminate, on the console appear this error: Unable to create RADIUS record. CDR not recorded! My cdr.conf is: [radius] usegmtime=yes; log date/time in GMT loguniqueid=yes ; log uniqueid loguserfield=yes ; log user field radiuscfg=/etc/radiusclient-ng/radiusclient.conf When I load the cdr_radius module no error appear. Any suggestion? Regards - Andrea Have you got a dictionary file with the attributes for asterisk? -- _ -- 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 list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk CDR Radius error
Hi, Have you got a dictionary file with the attributes for asterisk? Yes, my radiusclient-ng dictionary include dictionary.digium BR -- _ -- 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 list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco SIP 8.5 and 9.0 Issues
I know this doesn't answer your question directly, but Where are you getting the Sip 9.0 software? It is not available on Cisco's website. I have Sip 8.9 on my phone and I have noticed that after about 45 mins on a call it will hang up and drop the desktop connection that runs through the phone. I am hoping that upgrading to 8.12 will fix the issue, or I just wasted money on a SMARTnet contract. Regards, James I see blindness, not as a disability, but more of an ability. And Sight actually, more of a disability because some people with sight tend to judge others by what they see on the outside, whereas I don't see that. I just see that which is in a person. Patrick Henry Hughes, Louisville Kentucky,2008 On Tue, Oct 5, 2010 at 17:08, Gerard gsara...@rarcoa.com wrote: Hi list, I was wondering if anyone had any solution to either one of two issues I'm having: I have a cisco 7962G with the latest (from cisco) 8.5(4) SIP Firmware, it works very well for the most part, but after less then a week of heavy usage, eventually the phone gets into a state where it won't accept or let you place any more calls, the screen flashes no free lines available or something along those lines. (power cycle fixes this). So my preferred solution would be to upgrade to the v9.0(3) firmware, but when that's loaded, the phone won't register with Asterisk anymore, does anyone know if I need to adjust my .cnf.xml file, or is it a bug of some sort? Thanks for any input, -- Gerard Saraber Network Admin. Rarcoa, Inc -- _ -- 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 list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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 list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Web-meetme
Hi there! I am trying to configure Web-meetme on Asterisk 1.6. I have followed the README and everything looks ok,therefore, when I try to open the webpage appear the folowing messages:DB Error: connect failed Testing with a php script, the message Connected successfully is shown. Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda -- _ -- 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 list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] take input and store in variable
On 10/4/10 12:27 PM, Tom Lohmuller wrote:
I am using a context to change values in a DB. Currently in my context, I
am passing it to
exten = s,1,WaitExten(7) ; 7 seconds to input
exten = s,n,Set(NEW_VAR=${EXTEN}) ;Here is my problem. This is the only
way I know how to 'grab' user input, which was normally from ${EXTEN} but
I realize this won't work for extension 's'..
The short google search I did didn't turn up anything concrete.
try:
exten = s,1,WaitExten(7)
exten = _X!,1,Set(NEW_VAR=${EXTEN})
exten = _X!,n,do other things...
.
.
exten = t,1,Hangup() ;hang up if no input for 7 sec.
--
Edwin Lam edwin@officegeneral.com
Systems Engineer, OfficeWyze, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20
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Re: [asterisk-users] more condition check for gotoif
On 10/3/10 11:20 AM, Daniel Knoll wrote: Hello, is it possible to check more than one condition for GOTOIF in the dialplan? yes. check out asterisk expressions on wiki pages http://www.voip-info.org/wiki/view/Asterisk+Expressions -- Edwin Lam edwin@officegeneral.com Systems Engineer, OfficeWyze, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 -- _ -- 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 list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] take input and store in variable
Carlos Chavez wrote:
On Mon, 2010-10-04 at 14:27 -0500, Tom Lohmuller wrote:
I am using a context to change values in a DB. Currently in my context, I
am passing it to
exten = s,1,WaitExten(7) ; 7 seconds to input
exten = s,n,Set(NEW_VAR=${EXTEN}) ;Here is my problem. This is the only
way I know how to 'grab' user input, which was normally from ${EXTEN} but
I realize this won't work for extension 's'..
The short google search I did didn't turn up anything concrete.
What kind of search did you do to avoid getting the read command?
http://www.voip-info.org/wiki/view/Asterisk+cmd+Read
--
Vardan Harutyunyan,
Senior System Administrator
Enterprise Incubator Foundation
123 Hovsep Emin Street,
Yerevan 0051, Republic of Armenia
Tel: + 374 10 219735
Fax: + 374 10 219777
E-mail: i...@eif.am
www.eif-it.com
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Re: [asterisk-users] Asterisk CDR Radius error
Hi, As per the Asterisk documentation mentioned in the http://svnview.digium.com/svn/asterisk/ ... iew=markup followed the procedure for Call Detail Recording to RADIUS Server. I was getting the following error DEBUG[12542] cdr_radius.c: Unable to create RADIUS record. CDR not recorded! Any one faced this issues with RaidusClinet-ng(0.5.6) with Asterisk 1.6.2.14 I also faced this issue, any reason why cdr_radius.c is unable to create the record itslef.? Thanks, Phaneesh From: bakko asannu...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wed, October 6, 2010 3:20:40 AM Subject: Re: [asterisk-users] Asterisk CDR Radius error Hi, Have you got a dictionary file with the attributes for asterisk? Yes, my radiusclient-ng dictionary include dictionary.digium BR -- _ -- 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 list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
