Re: [asterisk-users] Can't make call with TDM410P
On Sat, Jun 23, 2012 at 10:32 AM, neo haux neo.h...@gmx.com wrote: Actually I can start and receive SIP calls (PC client, iphone client) but I have an issue with calling external number throught PSTN (certified-asterisk-1.8.11-cert2). I'm having this error when making a call: *CLI == Using SIP RTP CoS mark 5 -- Executing [9000@local:1] Dial(SIP/3000-0006, DAHDI/1/4384019357,10) in new stack [Jun 23 16:18:09] WARNING[28781]: app_dial.c:2218 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) == Everyone is busy/congested at this time (1:0/0/1) -- Executing [9000@local:2] Hangup(SIP/3000-0006, ) in new stack == Spawn extension (local, 9000, 2) exited non-zero on 'SIP/3000-0006' My configs : *CLI dahdi show channels Chan Extension Context Language MOH Interpret Blocked State pseudo default default In Service Where are your channels? That's why you are receiving the error Unable to create channel of type 'DAHDI'. Define your channel groups in /etc/asterisk/chan_dahdi.conf Then is should look like this: # asterisk -rx dahdi show channels Chan Extension Context Language MOH Interpret BlockedState pseudodefaultdefault In Service 1from-external en default In Service 2from-external en default In Service 3from-external en default In Service 4from-external en default In Service 5from-external en default In Service 6from-external en default In Service ... - Julian -- _ -- 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 client floods port 5060 and gets blocked
On Thu, Oct 28, 2010 at 1:42 AM, Jonas Kellens jonas.kell...@telenet.bewrote: On 10/28/2010 12:52 PM, Gordon Henderson wrote: On Thu, 28 Oct 2010, Jonas Kellens wrote On 10/28/2010 10:44 AM, Kevin Keane wrote: I assume that you checked and the remote IP is a legitimate IP phone? If not, it could be an attempt to break into your system. If it is a legitimate IP phone, make sure that the SIP configuration is correct -- if the SIP authentication fails, you can see this happening. 1. This is a legitimate phone, yes. 2. Registration goes as follow : REGISTER SIP/2.0 401 Unauthorized Re-Register with Digest 200 OK Is it s Snom phone? I've seen Snoms do this... Gordon I have this with Snom 320, Snom 370, Grandstream GXW4008 and YeaLink T28... Yes, I have seen this with Snom 370s... It's maddening. I'm going to start testing out the version 8.x firmware. - Julian -- _ -- 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] Simple dialplan issue
I have an issue where a particular dialplan works but another doesn't. I'm
not sure why. To me they look identical and it has me stumped.
This works:
[to-test]
exten = _X., 1, SetCallerPres(allowed)
exten = _X., 2, Monitor(wav,/tmp/test-${UNIQUEID},mb)
exten = _X., 3, Ringing
exten = _X., 4, Dial(SIP/9...@a-test,20,ro)
exten = _X., 5, GotoIf($[${DIALSTATUS} = ANSWER]?9)
exten = _X., 6, GotoIf($[${DIALSTATUS} = NOANSWER]?7)
exten = _X., 7, Dial(SIP/9...@a-test2,20,ro)
exten = _X., 8, GotoIf($[${DIALSTATUS} = ANSWER]?9)
exten = _X., 9, Hangup
This does NOT work:
[to-test]
exten = _X., 1, SetCallerPres(allowed)
exten = _X., 20, Monitor(wav,/tmp/test-${UNIQUEID},mb)
exten = _X., 30, Ringing
exten = _X., 40, Dial(SIP/9...@a-test,20,ro)
exten = _X., 50, GotoIf($[${DIALSTATUS} = ANSWER]?90)
exten = _X., 60, GotoIf($[${DIALSTATUS} = NOANSWER]?70)
exten = _X., 70, Dial(SIP/9...@a-test2,20,ro)
exten = _X., 80, GotoIf($[${DIALSTATUS} = ANSWER]?90)
exten = _X., 90, Hangup
This does NOT work either:
[to-test]
exten = _X., 1, SetCallerPres(allowed)
exten = _X., 2, Monitor(wav,/tmp/test-${UNIQUEID},mb)
exten = _X., 3, Ringing
exten = _X., 4, Dial(SIP/9...@a-test,20,ro)
exten = _X., 5, GotoIf($[${DIALSTATUS} = ANSWER]?200)
exten = _X., 6, GotoIf($[${DIALSTATUS} = NOANSWER]?7)
exten = _X., 7, Dial(SIP/9...@a-test2,20,ro)
exten = _X., 8, GotoIf($[${DIALSTATUS} = ANSWER]?200)
exten = _X., 200, Hangup
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Re: [asterisk-users] Simple dialplan issue
On Tue, Sep 22, 2009 at 6:34 PM, Matt Riddell li...@venturevoip.com wrote:
On 23/09/09 3:32 PM, Julian Yap wrote:
I have an issue where a particular dialplan works but another doesn't.
I'm not sure why. To me they look identical and it has me stumped.
This works:
[to-test]
exten = _X., 1, SetCallerPres(allowed)
exten = _X., 2, Monitor(wav,/tmp/test-${UNIQUEID},mb)
1, 2, yep.
[to-test]
exten = _X., 1, SetCallerPres(allowed)
exten = _X., 20, Monitor(wav,/tmp/test-${UNIQUEID},mb)
Normally 2 comes after 1 rather than 20 - looks like you're missing 2
through 19 here :)
Hmm, I guess I was under the understanding that it would work like that.
Why wouldn't this work?:
[to-test]
exten = _X., 1, SetCallerPres(allowed)
exten = _X., 2, Monitor(wav,/tmp/test-${UNIQUEID},mb)
exten = _X., 3, Ringing
exten = _X., 4, Dial(SIP/9...@a-test,20,ro)
exten = _X., 5, GotoIf($[${DIALSTATUS} = ANSWER]?200)
exten = _X., 6, GotoIf($[${DIALSTATUS} = NOANSWER]?7)
exten = _X., 7, Dial(SIP/9...@a-test2,20,ro)
exten = _X., 8, GotoIf($[${DIALSTATUS} = ANSWER]?200)
exten = _X., 200, Hangup
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Re: [asterisk-users] Simple dialplan issue
Thanks all, I worked this out with your help. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IP door opening devices
C F, Does the 2nd port of the ATA with 2 FXS ports just work like a 'pass-through' that is connected to the DTMF Relay? Or am I totally off track? Any ATA's with 2 FXS ports that you can recommend? Thanks, Julian On Thu, Jul 24, 2008 at 3:27 AM, C F [EMAIL PROTECTED] wrote: leave the existing keypad there. as for integrating it with asterisk. use an ata with 2 FXS ports. one FXS port connect to a viking door box http://www.vikingelectronics.com/ and set the ATA to do hotline on it. that door box is a regular analog phone in the shape of a door box that when call is pressed it goes offhook hence the requirement of hotline mode. it also has auto answer that when you call the box it goes off hook automaticaly. then use a relay from http://www.mikesandman.com/ that gets activated on ring connect that to the second FXS and that will unlock the door. On 7/24/08, Chris Bagnall [EMAIL PROTECTED] wrote: Greetings list, We have a client with an analogue door intercom/opening unit which we're attempting to replace with an IP variant. The existing unit has the following functionality: 1) Intercom - visitor hits call, talks to operator 2) Door opening - operator can open the door by dialling a 4-digit PIN followed by * (the door unit interprets the DTMF tones) 3) Door opening - the door unit has a numeric keypad to enable approved persons to enter by entering the 4-digit PIN on the keypad We've tried getting the existing unit working with an ATA, but it's only about 50% reliable (hangup not always detected, DTMF not always detected, etc.), so it's probably time to look at fully IP alternatives. Any suggestions gratefully appreciated. Regards, Chris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New Polycom SpectraLink 8002 Wifi SIP Handset
Michael, Nice review. Can this phone be provisioned without using TFTP? For example, over the internet? Are there other provisioning methods? Does it always require user input to start up? It looks like you need to input the SIP login and password details every time? That sounds very cumbersome from an end-user perspective. For example, if you manually assign an IP address and TFTP server, then can you power it on and start using it without further input? - Julian On Sat, Jul 5, 2008 at 10:39 AM, Michael Graves [EMAIL PROTECTED] wrote: If anyone is interested I just completed a review of this handset. I had it installed and in use the past couple of months. My review is here: http://www.smallnetbuilder.com/content/view/30498/80/ Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4.20-rc1 Now Available
The subject should read Asterisk 1.4.20-rc2 Now Available On Wed, May 7, 2008 at 11:24 AM, The Asterisk Development Team [EMAIL PROTECTED] wrote: The Asterisk development team has released Asterisk version 1.4.20-rc2. This release is a release candidate for the upcoming official release of 1.4.20. It includes a fix for a SIP channel driver regression introduced in 1.4.20-rc1, among a number of other changes. For a full list of changes since the last release candidate, view the contents of the ChangeLog that is distributed with the release. The release candidate is available on the download site. http://downloads.digium.com/pub/telephony/asterisk Please provide release candidate testing feedback to the asterisk-dev mailing list, or the issue tracker, http://bugs.digium.com/. Thank you for your continued support of Asterisk! ___ -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Big difference in CPU utilization with MeetMe
There is a bug in 1.4.19.1 with IAX. That's your issue. On Wed, May 7, 2008 at 12:38 PM, Kevin Ragsdale [EMAIL PROTECTED] wrote: Hello everyone, We are building a new * server based on a Supermicro motherboard with a 2.8 Xeon processor and a TE220B card. We're using the PBX In a Flash distribution. What we've found is that with a 4 user MeetMe conference, the CPU usage is consistently around 16%. This in comparison to our existing PSTN gateway * box running 1.09 (it hosts our conferences and terminates our T1s). With 23 users and processing all PSTN phone calls, CPU usage averaged from 3-8%. This is an older Supermicro, with a 2.4 Xeon processor. In both cases, the connections are via IAX trunks from our main PBX here, and in two remote locations. We use g711 u-law only - no other codecs are used. If we connect the same number of users through a PRI connection directly to the new server, the CPU is 1% or less, so obviously we've pooched something. We saw this same behavior when we split off the users to a 1.4x based PBX, and we thought it was the server hardware in the new machine, which was an older Dell 2650. But now we're not so sure. I know this is kind of vague, but can anyone suggest what might be happening? New Server CentOS 5, Kernel version 2.6.18-53.1.14.el5 Asterisk 1.4.19.1, and the SVN Zaptel drivers for the TE220B problems posted recently 2.8 Xeon, Hyperthreading disabled, 4GB RAM, 3Ware 9550SX RAID Old Server Fedora, Kernel version 2.4.22-1.2199.nptl Asterisk 1.0.9 2.4 Xeon, Hyperthreading off, 1GB RAM Thanks for the help, Kevin ___ -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk in Production ?
On Tue, May 6, 2008 at 1:38 AM, Benoit Plessis [EMAIL PROTECTED] wrote: We are actually running an AsteriskNow appliance with asterisk 1.4.18.1 and it's quite unstable. We have ~30 IAX2 SoftPhones and encounter some Avoiding IAX destroy deadlock and now that we have added a Queue, it's worse than ever. The queue goes stuck quite often (agent are stuck in 'In use' state and if they logoff they can't log-in till an asterisk restart). There's an IAX issue with the security patch for 1.4.18.1... and 1.4.19.1. There's another thread on this. - Julian ___ -- 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
Re: [asterisk-users] IAX issues with 1.4.19.1
That was a bug in the release. From the 1.4.20-rc1 Changelog: 2008-04-30 16:30 + [r114891] Russell Bryant [EMAIL PROTECTED] * include/asterisk/dlinkedlists.h (added), channels/chan_iax2.c: Merge changes from team/russell/iax2_find_callno and iax2_find_callno_1.4 These changes address a critical performance issue introduced in the latest release. The fix for the latest security issue included a change that made Asterisk randomly choose call numbers to make them more difficult to guess by attackers. However, due to some inefficient (this is by far, an understatement) code, when Asterisk chose high call numbers, chan_iax2 became unusable after just a small number of calls. On a small embedded platform, it would not be able to handle a single call. On my Intel Core 2 Duo @ 2.33 GHz, I couldn't run more than about 16 IAX2 channels. Ouch. These changes address some performance issues of the find_callno() function that have bothered me for a very long time. On every incoming media frame, it iterated through every possible call number trying to find a matching active call. This involved a mutex lock and unlock for each call number checked. So, if the random call number chosen was 2, then every media frame would cause 2 locks and unlocks. Previously, this problem was not as obvious since Asterisk always chose the lowest call number it could. A second container for IAX2 pvt structs has been added. It is an astobj2 hash table. When we know the remote side's call number, the pvt goes into the hash table with a hash value of the remote side's call number. Then, lookups for incoming media frames are a very fast hash lookup instead of an absolutely insane array traversal. In a quick test, I was able to get more than 3600% more IAX2 channels on my machine with these changes. On Thu, Apr 24, 2008 at 6:51 AM, Mike Clark [EMAIL PROTECTED] wrote: I upgraded one of our servers to 1.4.19.1 last evening, but ended up having to drop back because of IAX calls failing at a near 50 % rate. Here is the message that we would receive on the console (multiple times), and then it would hangup the call. Avoiding IAX destroy deadlock Anyone else having similar problems? Thanks, Mike Clark ___ -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Aastra 57i / 57i CT phones fail to re-register
Hi, Has anyone had experience with the Aastra 57i / 57i CT phones? The problem I'm having is that the phone isn't re-registering itself with the Asterisk server. As a result, the phone loses the registration with the server and is no longer able to receive incoming calls. Outgoing calls work fine. Right now, I have the following sip registration settings: sip registration period: 180 sip session timer: 30 .. But I've tried other settings but it doesn't work for me. I have Polycom phones as well and they work perfectly. Perhaps it's something I'm missing in the configuration? Thanks, Julian Running firmware: Firmware Version2.0.2.1040 Firmware Release Code SIP Boot Version2.0.1.1055 Date/Time May 4 2007 19:58:37 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
