Re: [asterisk-users] Asterisk 1.8.3 BLF stopped working
On 12/02/11 04:02, Bryant Zimmerman wrote: I am running 1.8.3 and my BLF lights have stopped working. The hints appear to be intact when I use core show hints. But none of the phones are getting the BLF updates. This has happend in the past and I have had to restart my server. What could be causing this to occur. It did not do this with the 1.6.x builds. Is there a way to reload the hints or force a refresh without re-starting Does a restart actually fix the problem? If not, compare the hint context in core show hints and the Subscr.Cont. line in sip show peer xxx, where xxx is one of the extensions attempting to subscribe to hints. Make sure the two match. I've had this problem before, and that was the cause. -- _ -- 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] No SIP requests coming in with Allow Anonymous SIP set to OFF - If set to ON it all SIP DEBUG show the requests just fine - Where is the problem?
On 09/12/10 07:06, Zeeshan Zakaria wrote: I think this may be because ... So you think, don't know. Maybe you knew if you knew the FreePBX code, or bothered to look into it. For God's sake, stick a sock in it. Others are attempting to help. You are not. -- _ -- 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] IPSec on asterisk
I don't know exactly what help you expect to receive in this forum. Asterisk itself has nothing to do with VPNs of any kind, and you should take your questions regarding the setup and configuration of them to the appropriate place. On 09/09/10 18:26, Deepika Nijhawan wrote: I am not getting anything in debug because call is not reaching us from other end, it is inbound connection over ipsec. Thanks. *From:* Deepika Nijhawan [mailto:deepika.nijha...@oxygen8.com] *Sent:* 08 September 2010 17:10 *To:* 'asterisk-users@lists.digium.com' *Subject:* IPSec on asterisk Hi, I am trying to configure ipsec on asterisk. Have configured /etc/racoon/racoon.conf and /etc/raccoon/psk.txt. Also have policy file in same folder. Have run racoon. Still I can’t receive calls. Can anyone please tell if any extra step is needed. Thanks -- _ -- 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 on Vmware
On 08/11/10 18:46, Tino wrote: Thanks Gareth for your quick reply. It is the lateset version and i think i need access to Dahdi interface. Is there any disadvantages other than this. If you need access to cards installed in the machine, you can forget running Asterisk under VMware. VMware does not allow direct access to the underlying hardware on the machine. -- _ -- 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] Can sip clients connect with each other directly (RTP session) ?
On 06/19/10 15:19, Kamonwat Sookkara wrote: Dear Asterisk friends, Please help me to clarify my doubt. After monitor SIP and RTP traffic with Wireshark. I found that both SIP and RTP traffic between 2 sip clients must be passed through Asterisk. Is it possible that 2 sip clients connect with each other directly for RTP session after sip session completed ? By default it is yes, however within a LAN environment you can usually allow clients to re-invite directly between themselves. Check the canreinvite option out.// -- _ -- 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] Which issue is keeping you from updrading to1.6.2 ?
On 05/21/10 09:07, Leif Madsen wrote: Danny Nicholas wrote: If I'm going to bother with 1.6.2, I'll wait a few months for 1.8. But in the spirit of your question: (1) dialplan conversion (2) loss of functions like Gosub Can you be more specific about what 1) and 2) mean? Number 1 is the same thing that people have been griping about for a very *very* long time with Asterisk. When you upgrade from one version to another, commands change, get deprecated and removed without suitable replacements, meaning that you have to go through your dialplan with a fine toothcomb finding every little change. Number 2 was referred to in number 1. For whatever reason, Digium choose to keep removing useful commands and features *without* supplying an equivalent alternative. When asked, the answer is you can do it in dialplan. What its not addressed is that the dialplan required to replace the functionality of one command can run into tens of lines of code. -- _ -- 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] PRI lines do not have CallerID activated yet it is
Exactly what is the problem you've having with CallerID? Are you not receiving it, or are you not able to send it? Which carrier are you using and what make and model card is the line connected to? For incoming calls on ISDN-10/20/30 lines, no special configuration is required to receive caller ID, however your carrier must be presenting it to you. Likewise no special configuration (beyond dialplan functionality to set the number to be sent) is required to set caller ID on outgoing calls, again provided your carrier will allow this. The format of the number to be set varies a great deal, but as a rule of thumb, you set the caller ID in the same format you receive it in. Any outgoing caller ID that does not fall within the allocated number range for the service will just about always be ignored, resulting in the default caller ID being presented to the called party. On 03/22/10 22:25, Nathanial Allan wrote: Hi, I am having some trouble setting up Caller id on my asterisk system, I need to know if there is anything special that needs to be done for an australian connection specifically as I have tried what most web sites on google reccomend but without success. I have not had much experience with asterisk as I have inherited this system from the previous sysadmin who has not documented anything so I am unsure what data you guys need from me, please advise what is needed and I will get it to you asap. Thanks in advance -- _ -- 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] adding agent with 2 phones to a queue - SOLVED
Glad to see I was able to point you in the right direction. On 03/14/10 23:56, Magnus Benngård wrote: queue add member Local/1...@agents to 0317998989 penalty 1 as Magnus Benngard state_interface hint:1...@agents - did the trick :) On Sun, 14 Mar 2010 11:38:13 +0100, Magnus Benngård magnu...@inputinterior.se wrote: I tried, [agents] exten = 1,hint,SIP/0317998975SIP/0317998985 exten = 1,1,Dial(SIP/0317998975SIP/0317998985) and queue add member Local/1...@agents to 0317998989 sip*CLI queue show 0317998989 0317998989 has 0 calls (max unlimited) in 'rrmemory' strategy (0s holdtime, 109s talktime), W:0, C:1, A:0, SL:0.0% within 0s Members: Local/1...@agents (dynamic) (Not in use) has taken no calls yet No Callers did call out from 0317998975 sip*CLI core show hints 0317998...@inputinterior.se: SIP/0317998975State:InUse Watchers 0 0317998...@inputinterior.se: SIP/0317998985State:IdleWatchers 0 1...@agents : SIP/0317998975SIP/0 State:InUse Watchers 0 Looks correct to me... but: sip*CLI queue show 0317998989 0317998989 has 0 calls (max unlimited) in 'rrmemory' strategy (0s holdtime, 109s talktime), W:0, C:1, A:0, SL:0.0% within 0s Members: Local/1...@agents (dynamic) (Not in use) has taken no calls yet No Callers Do understand that I have missed something here, shouldn't it be InUse?, Calling the queue (both phone are ringing) and answer gives: sip*CLI queue show 0317998989 0317998989 has 0 calls (max unlimited) in 'rrmemory' strategy (1s holdtime, 109s talktime), W:0, C:1, A:0, SL:0.0% within 0s Members: Local/1...@agents (dynamic) (Not in use) has taken no calls yet No Callers I am completly lost. :( On Sun, 14 Mar 2010 01:08:53 +1100, Rob Hillis wrote: Your best option is likely to be to create a separate context that calls both numbers, like so... [agents] exten = 1,Dial(SIP/0317998975SIP/0317998985) ...then add Local/1...@agents to the queue. On 03/14/10 00:03, Magnus Benngård wrote: Hi! We have alot of users who are having 2 phones, 1 fixed and 1 DECT. I am looking for a way to log them into a queue and let both phone rings. Let me try to explain: 0317998975 is a fixed phone, 0317998985 is a DECT. 0317998989 is a queue. queue add member SIP/0317998975 to 0317998989 works ofc. sip*CLI queue show 0317998989 0317998989 has 0 calls (max unlimited) in 'rrmemory' strategy (0s holdtime, 0s talktime), W:0, C:0, A:1, SL:0.0% within 0s Members: SIP/0317998975 (dynamic) (Not in use) has taken no calls yet But what i would like to do is something like: queue add member SIP/0317998975SIP/0317998985 to 0317998989 But that doesnt work. :( sip*CLI queue show 0317998989 0317998989 has 0 calls (max unlimited) in 'rrmemory' strategy (0s holdtime, 0s talktime), W:0, C:0, A:1, SL:0.0% within 0s Members: SIP/0317998975SIP/0317998985 (dynamic) (Unknown) has taken no calls yet Did try to add a hint: exten = kalle,hint,SIP/0317998975SIP/0317998985 just for testing purposes, that did work: ka...@inputinterior.se : SIP/0317998975SIP/0 State:Idle Watchers 0 But I cant figure out howto connect the queue with kalle, or maybe it is not possible? /Magnus -- _ -- 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] adding agent with 2 phones to a queue
Your best option is likely to be to create a separate context that calls both numbers, like so... [agents] exten = 1,Dial(SIP/0317998975SIP/0317998985) ...then add Local/1...@agents to the queue. On 03/14/10 00:03, Magnus Benngård wrote: Hi! We have alot of users who are having 2 phones, 1 fixed and 1 DECT. I am looking for a way to log them into a queue and let both phone rings. Let me try to explain: 0317998975 is a fixed phone, 0317998985 is a DECT. 0317998989 is a queue. queue add member SIP/0317998975 to 0317998989 works ofc. sip*CLI queue show 0317998989 0317998989 has 0 calls (max unlimited) in 'rrmemory' strategy (0s holdtime, 0s talktime), W:0, C:0, A:1, SL:0.0% within 0s Members: SIP/0317998975 (dynamic) (Not in use) has taken no calls yet But what i would like to do is something like: queue add member SIP/0317998975SIP/0317998985 to 0317998989 But that doesnt work. :( sip*CLI queue show 0317998989 0317998989 has 0 calls (max unlimited) in 'rrmemory' strategy (0s holdtime, 0s talktime), W:0, C:0, A:1, SL:0.0% within 0s Members: SIP/0317998975SIP/0317998985 (dynamic) (Unknown) has taken no calls yet Did try to add a hint: exten = kalle,hint,SIP/0317998975SIP/0317998985 just for testing purposes, that did work: ka...@inputinterior.se: SIP/0317998975SIP/0 State:IdleWatchers 0 But I cant figure out howto connect the queue with kalle, or maybe it is not possible? /Magnus -- _ -- 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 does holdtime get calculated for queues
On 02/17/10 03:39, Warren Selby wrote: I had a customer ask me this question today, and I was surprised to say I didn't have an exact answer for them. They have a relatively small support queue for their business (three agents, and rarely more than one person in line at any given time in the queue, if all agents are on a call), Their support calls can range anywhere from 30 seconds to 30 minutes. Occasionally, they'll get four or five calls in a short time span, and the first three that are answered may end up being rather long. The people in line are listening to hold music and hearing the periodic hold announcements and estimated wait times (you are currently caller number 2 in line, your estimated wait time is ). The complaint they're receiving is that the estimated hold time tends to jump around a bit - the example I was given was that a caller was told when they entered the queue, they were currently caller number 2 and the estimated hold time was less than 2 minutes. The caller waited upwards of 10 minutes, and then hung up when the estimated wait time announcement told them the wait time was 8 more minutes. I have a call centre that is very similar to yours - two or three operators, normally minimal hold times with occasional busy times (although in this case, it's when there's a special event on that the volume of calls pick up rather than the length of each call) and found that Asterisk's estimated hold time simply wasn't accurate enough. In my situation, I simply disabled the hold time and the complaints about inaccurate wait times were replaced by lots of praise for the mere fact that they were told your call is number 3 in line. The problem seems to be when Asterisk is used to long periods of next to no waiting time - something you can't really fault Asterisk for since it's not psychic. I found that after 10 or 15 minutes and a half dozen calls, the wait time became pretty accurate and stayed that way until the queue emptied again. It may be for the type of queues that we're talking about here that trying to calculate hold times simply isn't feasible. -- _ -- 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] rawplayer in asterisk 1.0.0
On 02/17/10 05:01, Steve Howes wrote: On 16 Feb 2010, at 17:36, Arjan Kroon | Mobillion wrote: We are using asterisk version 1.0.0. Wow. Yeah, that about sums it up. A little googling reveals that Asterisk 1.0 was announced on January 14th, 2005 - over five years ago. I would have thought that even if upgrading to 1.2 or 1.4 wasn't an option that upgrading to a bugfixed release of 1.0 would have gone a long way towards resolving this kind of problem. After all, Asterisk was a much less mature product at that stage - and common wisdom has /always/ been to avoid the first two or three releases of a new version. (though with Asterisk 1.0, there wasn't another stable version to run - everything up to that point was developmental) -- _ -- 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] Important security alert: update your dialplans now!
On 02/15/10 20:00, Randy R wrote: Olle, this may be a stupid question, but shouldn't a native santitize function be urgently added to the code base in all versions or change the dialplan compîler to ignore dangerous characters? Whilst I agree with this, the unfortunate attitude we seem to get from Digium on most of these issues is you can already do this in dialplan, therefore we don't need to invest any effort in it. The fact that a workaround may be quite difficult to implement properly doesn't come in to it. The most obvious example of this one is the deprecation and removal of chan_agent without any sort of replacement being introduced because it's already possible to do in the dialplan. -- _ -- 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] Dial script
I think he's referring to the fact that you seem to be looking to put together the telephone equivalent of a spam service. I'd be advising rm -rf / as well. On 02/06/10 16:19, Thomas Perron wrote: karl, does it make you feel good ? wow. pathetic. On Fri, Feb 5, 2010 at 11:00 PM, Karl Fife karlf...@gmail.com wrote: Try this: #rm -rf / - Original Message - From: Thomas Perron thomas.per...@gmail.com To: asterisk-users@lists.digium.com Sent: Friday, February 05, 2010 8:54 PM Subject: [asterisk-users] Dial script Does anyone have a Dial script or a hint on how I can dial 1 numbers in sequence? When the calls are answered, I play a .gsm or .wav. Then, if user presses a defined digit, the call gets bridged to me. -- _ -- 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 -- _ -- 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 1.4 or 1.6 automated install script for CentOS 5.3 or 5.4
On 01/16/10 04:27, Bruce Nik wrote: Hi Guys, Other than than yum repository (which fails when installing freepbx with it) are there any automated install scripts out there that would install Asterisk 1.6 or 1.4 onto a CentOS LAMP system? If the script install FreePBX that would be a BONUS. Try PBX-in-a-Flash. Undoubtedly it won't do everything you want out of the box, but I suspect it will do /most/ of what you want out of the box. -- _ -- 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 1.4 or 1.6 automated install script for CentOS 5.3 or 5.4
On 01/17/10 01:15, Tzafrir Cohen wrote: Try PBX-in-a-Flash. Undoubtedly it won't do everything you want out of the box, but I suspect it will do /most/ of what you want out of the box. But will not let you debug that install script. I tend to distrust running such a hidden script What hidden script are you referring to? -- _ -- 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] 10/100 voip phones and gigabit connection
On 01/15/10 17:54, randall wrote: hi all, i noticed that a lot of VOIP phones have a double network interface allowing you to use only 1 LAN cable for both the phone and your desktop, a really nice feature that can save a lot of cable, but most are 10/100 connections while i have a gigabit network. Off course there are phones available with a Gigabit connection but these are at least 3 to 4 times as expensive. another option would be to have both desktop and voip phone each a dedicated line ( basically having 2 seperate networks ), already have these in place from the old/current situation but i was hoping to clear some cables. does anybody know of another solution to this or is my conclusion above simply all the choice there is? You've hit the nail on the head. A VoIP phone with two network ports is probably best thought of as a two port switch. Like any switch, if you connect a gigabit NIC to a 10/100 switch, you'll end up with a 100 megabit connection. The only way to get a gigabit connection to your PC is via a phone that has gigabit ports, or have a separate cable back to the switch. Best practice is usually to segregate phone and PC networks anyway - it helps avoid degradation of VoIP quality when the LAN becomes heavily loaded. -- _ -- 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] UK dialing tone
On 01/10/10 05:03, --[ UxBoD ]-- wrote: Hi, I use VoIPTalk as my provider and unsure of a minor issue. When people call me they get a US ring tone instead of UK. Is this a Asterisk configuration issue or one for VoIPTalk ? I am running 1.6.2.0 and IAX2 trunks. This is almost always this is VoIP provider problem. Most of the time, your Asterisk server sends back a 180 RINGING message to your VoIP provider and it's up to either them or the telco they use to terminate calls to send the ringback tone remotely. -- _ -- 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] Showing name of extension when calling
On 12/23/09 12:23, Russell Bryant wrote: Wasn't this scheduled for 1.6.2? I don't believe so, but I could be mistaken. It's certainly not in 1.6.2 as best I can tell from looking over the source code :-) Nope, it's not in 1.6.2. It went into trunk after the 1.6.2 feature freeze. Wouldn't that imply that it will be in 1.6.3? ___ -- 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] Patch for app_dial.c: exit when just one ext is busy.
Leif Neland wrote: I made a patch to app_dial.c to make Dial(ext1ext2ext3,tumeout,B) return busy when just one extension is busy. Forgive me for the question, but /why/ do you want this behaviour? Isn't the whole point of dialling multiple extensions so that a call has a greater chance of being answered? ___ -- 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] softphone/debug panel with BLF
Indeed it does. You add contacts and set the softphone number to extension@server Leif Neland wrote: Philipp Kempgen skrev: Leif Neland schrieb: Mostly to debug/test BLF, is there a softphone or another app. which can subscribe to hints on Asterisk? X-Lite? http://www.counterpath.com/x-lite.html Philipp Kempgen It does not subscribe to hints on Asterisk. Leif ___ -- 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] music on hold
Peter Evans wrote: On Tue, Nov 11, 2008 at 02:35:19PM +0800, 邱磊 wrote: hii guys: i get the message from the asterisk: Started music on hold, class 'default', on Local/s...@skype-web-callback-dial-263to263-1775,1 [2008-11-11 14:32:41] WARNING[1781]: format_wav.c:156 check_header: Unexpected freqency 11025 [2008-11-11 14:32:41] WARNING[1781]: file.c:322 fn_wrapper: Unable to open format wav [2008-11-11 14:32:41] WARNING[1781]: res_musiconhold.c:259 ast_moh_files_next: Unable to open file '/data/TOMSKYPEIVR/var/lib/asterisk/moh/callback1': No such file or directory '/data/TOMSKYPEIVR/var/lib/asterisk/moh/callback1': No such file or directory This could be a tough one. If you can't solve this without help, should you really be playing with ancient scrolls of wisdom in the first place? Actually Peter, this isn't the real giveaway as to what the problem is. That would be the line that says [2008-11-11 14:32:41] WARNING[1781]: format_wav.c:156 check_header: Unexpected freqency 11025 The wave files aren't properly encoded for 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
Re: [asterisk-users] Time of Day Routing
Tilghman Lesher wrote: Regardless of how you think it should work, the poster above described precisely the way it works. If your end boundary is 12:00, it will evaluate as true all the way up until 12:01:59. If you don't want that, another poster has suggested using 11:59, which will work fine. Given backwards compatibility concerns, this is unlikely to change. Forgive me for saying so, but since when has Digium concerned itself with backwards compatibility in the past? I'm sure I don't need to tell you how much stuff gets broken between major releases. ___ -- 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] SNOM Phones Displays NR Frequently
Chris Bagnall wrote: First things first. You are running /very/ old versions of firmware - particularly on the 300 and 320. Upgrade them. I've been running 7.3.14 for some time without a problem, though it appears that 7.3.23 is now out. I concur about upgrading the software, but I'd stick with 7.3.14 for now - at least until there's more feedback on stability of .23 and .24. On the OP's original question, I have noticed Snoms (of any version) aren't very good at handling DNS failure - they tend to cache DNS lookup failure almost indefinitely. If DNS lookups to your registrar occasionally fail, you might want to specify registrar via IP rather than by name. I've never run in to that particular problem myself. I /do/ have a Zultys phone here that will try only once to resolve DNS - and wait forever if it doesn't get a reply. The Snom never has the same problem. (the usual cause for DNS resolution failure is that I've rebooted the one DNS server on my home network for one reason or another) I've been running 7.2.23 on my Snom since I sent this original message and it seems to be fine. No crashing, no problems with sound quality. ___ -- 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] SNOM Phones Displays NR Frequently
oi geli wrote: Hi, I am using SNOM phones with Asterisks for few years. They used go occasionaly NR, and I would reregister them from the phone web interface. But it started doing frequently for last few months. Here are SNOM Phone and the firmware version; snom190-SIP - Version-Code: snom190-SIP 3.56m snom320-SIP - snom320 jffs2 v3.36 snom300-SIP - snom300-SIP 6.5.2 Asterisk version - Asterisk CVS-01/29/04-16:41:27 I would appreciate any help to fix the NR problem. First things first. You are running /very/ old versions of firmware - particularly on the 300 and 320. Upgrade them. I've been running 7.3.14 for some time without a problem, though it appears that 7.3.23 is now out. Secondly, your version of Asterisk is not a stable release and it is ancient. Upgrade it to a stable release - and it would probably be worth the effort to migrate to at least 1.2, if not 1.4. ___ -- 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] Lagged Extension
800ms is horrendous lag for a VoIP connection. If I were you, I'd be investing some time in finding out why the lag is so great. Even if I do a ping to a UK address, I'm getting pings of no more than 300ms from Australia. Unless you've got multiple satellite connections in the path (in which case VoIP is pretty close to a lost cause anyway) I'd be looking at these high pings as a symptom of a network problem somewhere else. Ishfaq Malik wrote: I have an extension which is in a different country and is constantly lagged (about 800ms). When anyone tries to call this extension we get a No route to destination message. ___ -- 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] Lagged Extension
Low bandwidth is another possibility, but I'd have though that any connection slow enough to generate that much latency wouldn't be usable for VoIP in the first place. Ishfaq Malik wrote: Cheers Rob, I was thinking it was due to a low bandwidth connection at the other end but from what you're saying it sounds like this is not the case. ___ -- 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] Sangoma A200
Alex Samad wrote: Voltage isn't the issue - the difference is in the impedance. Australia I get this in my dmesg when I load up the rdm410 modules [1083334.103487] Freed a Wildcard [1083336.171371] ALAW override parameter detected. Device will be operating in ALAW [1083338.040522] Boosting ringer on slot 1 (89V peak) [1083338.040542] Port 1: Installed -- AUTO FXS/DPO [1083340.340472] Boosting ringer on slot 2 (89V peak) [1083340.340492] Port 2: Installed -- AUTO FXS/DPO ALAW is good - that's the default in use for Australia. FXS ports aren't relevant since they're for extensions, not PSTN lines. uses complex impedance (220+820Ohm resistors with a 120nF capacitor) whereas the US uses a straight resistor. Did yo buy from the us or local ? Local I believe, though I can't tell you through whom. (I didn't handle the purchase of the card - merely the installation) ___ -- 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] Sangoma A200
Yes, although not for connecting to the PSTN - I've used one for connecting to a legacy NEC PABX. Voltage isn't the issue - the difference is in the impedance. Australia uses complex impedance (220+820Ohm resistors with a 120nF capacitor) whereas the US uses a straight resistor. Alex Samad wrote: Hi I was wondering if any one has used these cards, I am looking at this as a replacement for the tdm410, I have some issues with installing the tdm410 in a small case because of the power plug being at the end of the board. I am in australia seems like we have a different setup for out fxs voltage, any one in oz using this card ? Thanks Alex ___ -- 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] Outgoing CallerID for KPN in Belgium
Bart Coninckx wrote:
Hi,
I'm using a ISDN-30 E1 line from KPN Belgium.
The challenge is to get a correct CallerID on outgoing lines.
When I put this in my dialplan:
exten = _0.,1,Set(TEMPVAR=${CALLERID(num):1})
exten = _0.,2,Set(CALLERID(num)=144622${TEMPVAR})
exten = _0.,3,NoOp(${CALLERID(num)})
exten = _0.,4,Dial(Zap/g1/${EXTEN:1},,)
The resulting CallerID is accepted by the telco, but on phones it shows for
instance as:
+14462241, whereas it should be +3214462241. So it seems the telco adds a
+. I've tried to then use:
exten = _0.,2,Set(CALLERID(num)=32144622${TEMPVAR})
but the telco seems not to accept this since it sends the general CallerID
out.
Any clues on what I need to change to get this working? Is it something in
zapata.conf? Is it related to nationalprefix and internationalprefix?
Your best bet is to ask your telco what caller ID format you should be
presenting. Asterisk is obviously working as expected, but for whatever
reason the telco is either not accepting the right format, or not
processing the presented caller ID correctly.
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Re: [asterisk-users] SIP 482 Loop detected
jonas kellens wrote: Do you understand what is happening ? I don't understand what this sentence means : SIP/3starsnet-08d70ea8 is making progress passing it to SIP/twinkle-08de0490 Pretty simple really. Your SIP trunk 3starsnet is making progress with the call and Asterisk is passing that message on to SIP/twinkle. Entirely normal. ___ -- 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] No exten available after pass between servers
Dan Pilcheck wrote: The call will go over the server fine, but when the Call Center server answer, the CLI returns: NOTICE[4296]: chan_iax2.c:7398 socket_read: Rejected connect attempt from 10.0.10.20, request '2...@2xxx' does not exist What context are the phones in the extension range 2XXX in? I don't know what Vicidial's default context for extensions is, but I'd be surprised if it's 2XXX. Can you show us the results of sip show users after you remove the secret from the output? ___ -- 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] IP phone recommendation
Cary Fitch wrote: We have a bunch of SNOM 360’s we are not using. I agree they are not intuitive to the user. They work ok in general. I would part with 15 or so at an attractive price, one or more, I like the Grandstream 2000 series. Easy to use, easy to set up, good web page. Priced nice on the wholesale market. Just don't /ever/ upgrade the firmware if you get one that is semi functional. My experience with the first five GXP2000s I bought means that I will not ever consider Grandstream products again. Maybe I got a bad batch, but they've been such a nightmare, I'll be happy to never see the things again. ___ -- 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] IP phone recommendation
Christian Stredicke wrote: Check out the snom 300 or the snom 820... Good lord... talk about two extremes... :) The Snom 300 is pretty good, but the 320 is much better and costs around a *third* of what the Snom 820 does. Stick with the older model snoms. So far I've seen nothing about the 820 to justify the significant extra expense. ___ -- 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] IP phone recommendation
Alex Samad wrote: I have been looking at a snom 300, which seems okay. the display goes a bit haywire occasionally - not sure why yet. Are the 320 worth the extra money ? IMO yes, though it really depends on what you want from the phone. The Snom 320s handle transfers considerably better than the 300s and all round are a little easier to work with and use. The 12 programmable keys that can be used for lines, BLF indication, speed dials as well as other things are a big plus. They are easy to configure - and unlike many other makes of phone do /not/ reboot on the drop of a hat. The only real problem I have with the Snom 320 is the display - the display is a bit too small and /isn't/ backlit. considerably better than the 300s. Having said that, the display on the 320s are still much better than those of the 300s. ___ -- 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] IP phone recommendation
Jeff LaCoursiere wrote: We are still talking about a $175 phone. How about the Polycom IP 320? $85 at 888voipstore. Can't go wrong with Polycom for voice quality. True, Polycom's are brilliant for voice quality, but unlike the Snom, a Polycom /will/ reboot on the drop of a hat /and/ take a damned long time to do it (~45-60 seconds) In addition, the web interface should be taken away and shot - the only real way to configure them is through (T)FTP. They are however, extraordinarily configurable through the XML config and they are very stable. Once they're configured they work very nicely. The lack of a decent number of BLF keys (even with a very expensive sidecar you only get two more keys than a standalone Snom320) puts me off a little. However, for a conference phone, the Polycom's can't be easily beaten. Their handsfree call quality is in a league of it's own. ___ -- 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] IAX2 trunking with Older Asterisk, version ?
The clue in the log is no authority found. Something in the
configuration at the other end doesn't match the configuration at this
end - almost certainly the username and password.
Why are you including the IP address when dialling the trunk? If your
peers are set up with IP addresses (which they are) it should not be
necessary.
By the way, it's a *very* bad idea to post passwords in a public forum.
Tharanga wrote:
my sip phone registered on 1.6, when i dial 4567 from 1.6 version, it wont go
to 1.6 voice mail. it says
== Using SIP RTP CoS mark 5
-- Executing [4...@sip:1] Dial(SIP/312-09f9a720,
IAX2/trun...@147.120.203.98/4567,10,t) in new stack
-- Called trun...@147.120.203.98/4567
[Jun 1 11:01:18] WARNING[8178]: chan_iax2.c:8991 socket_process: Call
rejected by 147.120.203.98: No authority found
-- Hungup 'IAX2/trunk14-9738'
== Everyone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel 'SIP/312-09f9a720' status is 'CHANUNAVAIL'
[trunk14]
type=friend
host=147.120.203.98
auth=plaintext
secret=XX
context=sip,sip2,sip3
;keyrotate=off
permit=0.0.0.0/0.0.0.0
1.6 EXTENSIONS.CONF
[globals]
TRUNKIAX14=IAX2/trun...@147.120.203.98
[sip]
;exten = 4567,1,Dial(${TRUNKIAX14}/${EXTEN}|10|t)
exten = 4567,1,Voicemail(${EXTEN},u)
~
1.2 EXTENSIONS.CONF
[Jun 1 05:20:31] NOTICE[9536]: chan_iax2.c:8782 socket_process: Rejected
connect attempt from 147.120.203.71, who was trying to reach '4567@
[trunk14]
type=friend
host=147.120.203.71
auth=plaintext
secret=Mah
context=sip,sip2,sip3
;keyrotate=off
permit=0.0.0.0/0.0.0.0
[globals]
TRUNKIAX14=IAX2/trun...@147.120.203.71
[sip]
exten = s,1,wait(1) ; Answer the line
exten = s,n,BackGround(demo-congrats)
exten = s,n,ResponseTimeout,5
exten = s,n,Dial(SIP/${EXTEN},20,t)
;exten = s,n,BackGround(goodbye)
exten = s,n,Hangup
exten = 4567,1,Dial(${TRUNKIAX14}/${EXTEN},10,t)
Asterisk versions may differ. I do IAX trunk successfully even
between Asterisk 1.0.2 and 1.4.xx
please post your Dial command.
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Re: [asterisk-users] An outside Caller ID not shown,
Sounds like you're looking at the wrong variable. You should be looking at CALLERID(num). peace keeper wrote: Hi there, I am using the Asterisk as the PBX, and need to know the caller ID for the incoming call, but when I show the caller Id, it gives the Zaptel channel that recieves the inbound calls in the asterisk, Am I missing some configuration ! what should I do to be able to exteract the real callerId (that from the outside) from that channel, Thanks in advance, ___ -- 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] advice on OrderlyStats (or other cc software)
Louis-David Mitterrand wrote: Hi, Is anyone here using OrderlyStats with asterisk in a call center setting? If so what what is your experience with it? Is that software really free for asterisk users? Or is there a better option out there? The short answer is OrderlyStats isn't really free for Asterisk. The long answer is that OrderlyStats is free for Asterisk systems with two or less agents. That's really only applicable for the tiniest of call centres. I haven't used OrderlyStats, so I can't speak for the relative merits of it. However, I have used QueueMetrics (which incidentally is /also/ free for call centres of two or less simultaneous agents) and am fairly happy with it. It's not spectacularly pretty - only the latest version has begun to introduce graphs and charts, but it's functional. The price is similar to that of OrderlyStats and the licence you purchase for both of them is time limited - 4 years in the case of QueueMetrics, 5 for OrderlyStats. QueueMetrics will offer a 50% discount for non-profit organisations - I don't know whether OrderlyStats offers the same thing or not. My suggestion is that you get trial licences for both and give them a go. My suggestion is that you rev ___ -- 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] Parked calls for multiple customers
carl Lougher wrote: Ok cheers. Any idea when 1.6 goes stable for prod? Theoretically it already has, however as was the case with 1.4, I suggest you tread very carefully when it comes to migrating to 1.6. ___ -- 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] Outgoing Queues
Sebastian wrote: Anyone thought about something like outgoing queues? Many people have. I know QueueMetrics has methods for this kind of thing, and I'm fairly sure that Vicidial does as well. ___ -- 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 routine maintenance activities
Darrick Hartman wrote: Rob Hillis wrote: Daily? No. However, after implementing a weekly restart of Asterisk, I've found the instance of lockups and CPU utilisation spikes have decreased significantly. Unless you're using some unstable modules, there really should be no need to restart Asterisk. Is there a certain activity that is causing these lockups? I have low power systems which haven't had Asterisk restarted in months many times. Granted, these are mostly low call volume systems, but unless there is a memory leak, you should not needed to restart the Asterisk process. (my guess is one of the modules you are using has some sort of problem). This particular system isn't low power - it's a full blown server. Since I don't work at this place, I don't know what people are doing at the time the system freezes up. It's been some time since I updated Asterisk at this site, so they're probably running version 1.4.17 - 1.4.20 there. (it's a voluntary organisation where I've since become sick of (a) the politics and (b) their expectation that I drop what I'm doing to help them, regardless of whether I'm at work or not) If I were to do things again, I'd be running Astlinux on a net 5501 with an integrated hard drive (for voicemail/IVR and so on) Only time I've ever had to reboot my Astlinux box at home (on an ALIX-3) is when it's time to upgrade Astlinux. ___ -- 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 routine maintenance activities
Darrick Hartman wrote: If I were to do things again, I'd be running Astlinux on a net 5501 with an integrated hard drive (for voicemail/IVR and so on) Only time I've ever had to reboot my Astlinux box at home (on an ALIX-3) is when it's time to upgrade Astlinux. That's what we like to hear! Did you update to the latest version (0.6.5)? Is the Pope Catholic? ___ -- 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 routine maintenance activities
Kurian Thayil wrote: On Wed, 2009-04-22 at 15:24 +1000, Lee, John (Sydney) wrote: Daily Asterisk restart Do you think its mandatory in production env? Daily? No. However, after implementing a weekly restart of Asterisk, I've found the instance of lockups and CPU utilisation spikes have decreased significantly. ___ -- 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 1.4 to 1.6 extensions.conf
Benny Amorsen wrote: Michael mich...@networkstuff.co.nz writes: pbx.c:3143 pbx_extension_helper: No such label 'outgoing|PHONE NUMBER|1' in extension 'PHONE NUMBER' in context 'phones' [Apr 20 15:43:15] WARNING[11793]: pbx.c:8650 pbx_parseable_goto: Priority 'outgoing|PHONE NUMBER' must be a number 0, or valid label You want , instead of | Asterisk always shows parameters separated by a pipe symbol in the log. He may very well have parameters separated by a comma in the dialplan. ___ -- 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 1.4 to 1.6 extensions.conf
Michael wrote: pbx.c:3143 pbx_extension_helper: No such label 'outgoing|PHONE NUMBER|1' in extension 'PHONE NUMBER' in context 'phones' [Apr 20 15:43:15] WARNING[11793]: pbx.c:8650 pbx_parseable_goto: Priority 'outgoing|PHONE NUMBER' must be a number 0, or valid label PHONE NUMBER = the number I called. This dialplan worked fine in version 1.4. It's normal for dialplans to have to be rewritten slightly between major releases of Asterisk. It would help if you could post the parts of the dialplan that are giving you trouble. ___ -- 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] Alcatel OmniPCX Enterprise + Asterisk with E1
Sebastian Milioto wrote: 2. What E1 card should I buy for Asterisk? Is the physical interface (conectors) E1 identical as T1? The connectors are identical, however the protocol isn't. However, just about all the T1 cards I'm aware of support E1 as well - usually selected by a jumper on the card. 3. If cost wasn't a problem, do you suggest another interconection way technically better? May be replacing Asterisk with another device with an in-box E1? Theoretically the Alcatel OmniPCX could be trunked to Asterisk via SIP. Sometime very soon, I'll be trying to integrate ~40 OmniPCX systems via SIP, but haven't begun work on it yet. ___ -- 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] Ring group howto
Michael wrote:
On Fri, 03 Apr 2009 12:32:03 you wrote:
Like:
exten =
5226001454,1,Dial(SIP/3615221401SIP/3615221402SIP/3615221407SIP/52260014
0 5,20)
That is what I am currently doing - though is there a cleaner way?
The only cleaner way is to define the group in [globals] as follows:-
[globals]
group1 = SIP/3615221401SIP/3615221402SIP/3615221407SIP/52260014
...and then refer to this variable in the dial statement...
exten = 5226001454,1,Dial(${group1},20)
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Re: [asterisk-users] Xorcom and Doorbell
Loic Didelot wrote: Hi, I am trying to connect a doorbell to a Xorcom device. And the setup is quite simple. But when I push the doorbell all I see on the asterisk cli is: -- Starting simple switch on 'Zap/11-1' [Apr 2 13:00:40] DEBUG[8771]: chan_dahdi.c:6180 ss_thread: not enough digits (and no ambiguous match)... -- Hungup 'Zap/11-1' What number do you have your doorbell configured to dial when the button is pushed? Can you post the context that the doorphone's channel is configured to use? I defined the extension s,h,i,t,T etc... in my context. Any idea what I might do wrong? Your doorbell won't be dialling any of those extensions, of that you can be sure. ___ -- 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] What is the one thing that polycom can do...
Paul Hales wrote: I would love to see the agent login/logout stuff working - but that's just me. I'd like to see the damn web interface become usable. ___ -- 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] hum noise
Rilawich Ango wrote: My configuration is simple as below. SIP phone - asterisk - CISCO - T1 Do you mean the hum noise is created by electric-magnetic field? Asterisk can do nothing to eliminate it? That would be my bet. No, Asterisk can't do anything to remove EM noise. That's up to you since EM noise can just about always be filtered out. The most likely source is from telephone cables being run too close to power cables, fluorescent lights and so on. Improper shielding of components inside your PC could also be your problem. If your connection to the outside world is T1, it's unlikely to be an external problem. (digital signals aren't affected in this manner by EM interference) It's possible the SIP phone itself is faulty - other than that, make sure your Asterisk box and Cisco gear aren't located right next to heavy duty power cables or transformers. ___ -- 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] Need to find small footprint asterisk platform
Anthony Plack wrote: Hey all, I have a potential project which calls for a very small form-factor computer like this: http://www.marvell.com/products/embedded_processors/developer/kirkwood/sheevaplug.jsp However, I am needing an FXS port integrated into a small footprint computer. Nothing larger than a WiFi router or gateway device, but the smaller the better, and able to run Asterisk with at least a spare USB port and preferably WiFi on the system (but no necessary). One of the more common embedded platforms for Asterisk is the Soekris net5501 (or 4501 if you don't need as much processing power) The case would be the size of a larger router and has capacity for a single full height PCI device - such as a TDM400. Astlinux has an image for the 5501, so you can get away without a hard drive if you want to. (though Astlinux will happily use one as it's UnionFS partion) This will also give you a USB port. ___ -- 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] Recommended USB Headsets ?
About two and a half years ago, I upgraded a small call centre from corded handsets to X-Lite with Plantronics CS60 USB headsets. X-Lite lasted about two or three months before we ditched it in favour of Eyebeam. X-Lite disables too many features to be useful. With the Plantronics headset, X-lite and Eyebeam are about the only ones that support answering and terminating calls from the headset properly, so you're pretty well locked in there. (there are others that support the Plantronics headset, but every one I found was commercial and usually vendor specific) The Plantronics headsets are still going. Every few months, I get a complaint about calls dropping out, and every time it's been because the staff member in question hadn't put the headset back on the charger properly. Make sure that your echo cancellation is well taken care of if you're contemplating using softphones - generally speaking there's a much longer delay (in the order of ~150-200ms) in softphones compared to any physical phone - IP or analogue, so any echo you have will be much more noticeable and far more distracting. Edward Gray wrote: Hi, we are looking to roll out PBX IN A Flash at our office. The first group will be using Soft Phones (X-Lite appears to be the best and works in Windows, Apple Linux). There are many types of USB Headsets to choose from and a fairly broad price range. Is there any USB headsets people would recommend? I'm specifically interested in acceptable audio (speaker and microphone) quality for business calls but am sensitive to price as well. In reading online, the Logitech Premium headset does get some good reviews but the reviews appear to be more from consumer based. I'd much prefer real experience from the good people who are operating their own Asterisk implementations. Any advice? Thanks! ___ -- 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] gpx 2000 Busy Lamp Field
Yes. Grandstreams suck. Oguzhan Kayhan wrote: Hello, I configured both asterisk and grandstream 2000 accourding to howtos on the web.. And everything seems working fin. But if i reload asterisk grandstream stops working with BLF. I need to restart the phone to enable BLF again. Any clues?? ___ -- 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] Simple(?) dialplan question.
Asterisk wrote:
You can simply:
exten = _0.,n,Goto(${DIALSTATUS})
(before the playback)
Use the labels as the destinations - eg.
exten = _0.,n(BUSY),Noop()
exten = _0.,n(CONGESTION),Noop()
I've never seen that before, does that definitely work in 1.4.x? If so,
cool...
That's been a part of the standard extension macro I've been using
forever, as follows...
[macro-stdexten]
exten = s,1,Set(__DYNAMIC_FEATURES=${FEATURES})
exten = s,2,GotoIf($[${FOLLOWME_${ARG1}} = 1]?5:3)
exten = s,3,Dial(${ARG2},${RINGTIME},${DIALOPTIONS})
exten = s,4,Goto(s-${DIALSTATUS},1)
exten = s,5,Macro(stdexten-followme,${ARG1},${ARG2})
exten = s-NOANSWER,1,Voicemail(${ARG1},u)
exten = s-NOANSWER,2,Goto(default,s,1)
exten = s-BUSY,1,Voicemail(${ARG1},b)
exten = s-BUSY,2,Goto(default,s,1)
exten = _s-.,1,Goto(s-NOANSWER,1)
exten = a,1,VoicemailMain(${ARG1})
(the syntax is marginally different, but not significantly. Note the
_s-. extension to catch any odd/unexpected return status codes)
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Re: [asterisk-users] Best way to get 60+ analogue extensions.
Duncan Turnbull wrote: Hi All I am looking at a replacement for a hotel PBX which requires at least 60 analogue extensions. I tend to use Sangoma equipment but haven't tried this many analogue extensions before. I am interested in anyone's experience of which server platform literally fits and copes well with multiple cards, and the choice of Digium vs Sangoma or something else. You have several options here, however due to the power requirements, I wouldn't recommend you use either the Sangoma or Digium analogue cards here - providing ring voltage to that many extensions is likely to over-tax the power supply in the server. I'd either be looking at three channel banks (3 24 channel channel banks would give you a total of 72 analogue channels) or two Xorcom Astribanks which would likewise give you up to 64 channels. The Astribanks are probably a cheaper way to go since they connect to your server via USB rather than T1/E1 ports. However, I haven't had any experience with multiple Astribanks connected to the same server, so there may be issues there that I'm not aware of. Channel banks are certainly the proven and reliable technology, but will be significantly more expensive since they connect to your Asterisk server via T1/E1 links. I may need an ISDN PRA inbound but am working hard to have the inbound lines via SIP, but if I do that means at least 4 slots on this plan. You'd need to be very sure of the bandwidth and quality of connection to your VoIP provider to go with SIP for more than half a dozen channels. This kind of connection can easily be far more expensive than a traditional T1/E1 line, so I wouldn't be pushing so hard for SIP. If you were to use channel banks, you would most likely end up with a four port T1/E1 card and would only be using three of those channels, leaving a spare one for an incoming T1/E1 line. If you were to use Astribanks, you would have plenty of space in the server to include a T1/E1 card. ___ -- 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] Faxing with asterisk
Fabio Mosti wrote: 2009/2/16 Steve Underwood ste...@coppice.org: You don't indicate the kind of setup you are using. I use asterisk (Spandsp) with a IAX2 trunk (ethernet connection) to another asterisk (zap). client-asterisk (Spandsp)-asterisk (zap)-fax To quote the Mythbusters, there's your problem. Fax over IP = forget it unless the connection between your two Asterisk machines is some form of LAN connection. This *may* change a little when the T.38 support in Asterisk includes a gateway mode, which I don't believe it does yet. (IIRC 1.6 includes much better support for T.38, but I don't think it includes this kind of gateway yet - anyone care to correct me?) ___ -- 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] Siemens Hipath PRI to Asterisk Call Routing?
Which line of code is generating this log entry? [Feb 11 17:45:31] VERBOSE[5764] logger.c: -- Executing [91...@from-pstn:3] Goto(Zap/31-1, s|a2) in new stack ...because this appears to be where your problem lies. joek...@gmail.com wrote: Hi all, I have a connect between a siemens hipath Asterisk system over PRI The connection works perfectly I can call from the Hipath to an Asterisk Extension. I want to allow the hipath extensions to dial out over a SIP trunk on asterisk but I keep getting The number you have dialed is not in service In this CLI I'm dialing from hipath extension 9 (prefix for sip trunk) then the number 1905 (Freefone number in Ireland) Please help I cant figure this one out. Thanks, Joe CLI - [Feb 11 17:45:25] VERBOSE[4526] logger.c: -- Accepting overlap call from '0339' to 'unspecified' on channel 0/31, span 1 [Feb 11 17:45:25] VERBOSE[5764] logger.c: -- Starting simple switch on 'Zap/31-1' [Feb 11 17:45:31] VERBOSE[5764] logger.c: -- Executing [91...@from-pstn:1] Set(Zap/31-1, __FROM_DID=91905) in new stack [Feb 11 17:45:31] VERBOSE[5764] logger.c: -- Executing [91...@from-pstn:2] NoOp(Zap/31-1, Received an unknown call with DID set to 91905) in new stack [Feb 11 17:45:31] VERBOSE[5764] logger.c: -- Executing [91...@from-pstn:3] Goto(Zap/31-1, s|a2) in new stack [Feb 11 17:45:31] VERBOSE[5764] logger.c: -- Goto (from-pstn,s,2) [Feb 11 17:45:31] VERBOSE[5764] logger.c: -- Executing [...@from-pstn:2] Answer(Zap/31-1, ) in new stack [Feb 11 17:45:31] VERBOSE[5764] logger.c: -- Executing [...@from-pstn:3] Wait(Zap/31-1, 2) in new stack [Feb 11 17:45:33] VERBOSE[5764] logger.c: -- Executing [...@from-pstn:4] Playback(Zap/31-1, ss-noservice) in new stack [Feb 11 17:45:33] VERBOSE[5764] logger.c: -- Zap/31-1 Playing 'ss-noservice' (language 'en') [Feb 11 17:45:38] VERBOSE[5764] logger.c: -- Executing [...@from-pstn:5] SayAlpha(Zap/31-1, 91905) in new stack [Feb 11 17:45:38] VERBOSE[5764] logger.c: -- Zap/31-1 Playing 'digits/9' (language 'en') [Feb 11 17:45:39] VERBOSE[4526] logger.c: -- Channel 0/31, span 1 got hangup request, cause 16 [Feb 11 17:45:39] WARNING[5764] file.c: Failed to write frame [Feb 11 17:45:39] VERBOSE[5764] logger.c: == Spawn extension (from-pstn, s, 5) exited non-zero on 'Zap/31-1' [Feb 11 17:45:39] VERBOSE[5764] logger.c: -- Executing [...@from-pstn:1] Hangup(Zap/31-1, ) in new stack [Feb 11 17:45:39] VERBOSE[5764] logger.c: == Spawn extension (from-pstn, h, 1) exited non-zero on 'Zap/31-1' [Feb 11 17:45:39] DEBUG[5764] chan_dahdi.c: Set option AUDIO MODE, value: ON(1) on Zap/31-1 [Feb 11 17:45:39] DEBUG[5764] chan_dahdi.c: Not yet hungup... Calling hangup once with icause, and clearing call [Feb 11 17:45:39] DEBUG[5764] chan_dahdi.c: Set option AUDIO MODE, value: OFF(0) on Zap/31-1 [Feb 11 17:45:39] VERBOSE[5764] logger.c: -- Hungup 'Zap/31-1' ___ -- 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] AgentCallBackLogin no longer works after installing asterisk 1.6
...except that Macros are now deprecated and will most likely be removed
in 1.8.
Robert Broyles wrote:
Hmm, this is all very interesting.
Looks like using a Macro and the 'M' Dial() option is the way to go for
now if you need the answer confirmation.
http://www.voip-info.org/wiki-Asterisk+cmd+Dial
Look at example #2, and adapt it for your needs.
--
Regards,
Robert Broyles
Philipp Kempgen wrote:
Anthony Francis schrieb:
Robert Broyles wrote:
Check out this alternative:
http://hostseries.com/agentcallbacklogin-alternative/
---cut---
[agents]
exten = 1050,1,Set(AGENT_SIP=${DB(agent_sip/1050)})
exten = 1050,n,Dial(SIP/${AGENT_SIP})
---cut---
I like what he came up ,with however it doesn't replace the agent
callback login systems use of being able to make an agent press a key to
accept a call, very important when people are logging in via cell phone
and you don't their voice mail answering the call. In fact none of the
replacements do that. FAIL
First of all: The voicemail acts on behalf of the subscriber.
If they configure their voicemail to answer the call it's their
fault.
But I understand the problem.
http://www.voip-info.org/wiki/view/Asterisk+ZAP+channels
talks about answer confirmation:
---cut---
If the letter c follows, then Answer Confirmation is requested,
in which the call is not considered answered until the called
user presses #.
---cut---
So Dial(Zap/G1c/${phone_number}) might work.
Not sure why that is implemented for Zap channels only.
It should be an option to Dial() instead.
Philipp Kempgen
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Re: [asterisk-users] sip peer permit/deny - Need some explanation
Administrator TOOTAI wrote: [MyPeer] host=xxx.xxx.xxx.139 deny=0.0.0.0/0.0.0.0 permit=xxx.xxx.xxx.136/255.255.255.248 ;IP address from range 138 to 142 permit=yyy.yyy.yyy.yyy/255.255.255.255 On incoming calls, when the peer address is the one terminating with .139 everything is OK. If I change the external IP from the peer *ON* the peer machine to let's say .140 (or any other permitted address from this peer), incoming calls are not recognized despite the deny/permit stanza. If I modify the host to .140 in my peer definition, it's again working normally. Question is: why even by allowing in the permit stuff the allowed IPs from a peer, Asterisk does only accept calls from those peers if the peer machine has the IP address from the host definition in my peer sip.conf Since you are including a specific IP address in the host line, Asterisk will not accept calls from any other IP address. If you want to accept calls from multiple IP addresses, you *must* set host to dynamic and then use the permit/deny lines to restrict calls accordingly. Of course, since your sip peer is now set to dynamic, it will now need to register with 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
Re: [asterisk-users] Problem: no such extension 'xx' in context 'default'
Michael wrote: Change it to the following: exten = _10,1,Dial(SIP/10,10) exten =_10,n,Background(vm-nobodyavail) exten = _11,1,Dial(SIP/11,5) exten =_11,n,Background(vm-nobodyavail) The only time I am aware of that you can leave out the prefix underscore is for exten = s and exten = i No, you can leave out the leading underscore when you are using explicit extension numbers. (such as those above) As soon as you introduce /any/ pattern matching characters, you /must/ include the leading underscore. ___ -- 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] Good comparisons on cheaper VOIP phones
forums - sigma wrote: having deployed a fair amount of phones I have the following observation (and these observations are worth what you paid for them :-) ) 1. Linksys 942, my preferred mainstream desk phone, a bit more expensive than the Polycom IP330. Be careful as there are two SKUs with and without power supply (which is true of the ip330). The 942 has a nice large backlit screen, nice big MWI light, takes a 3.5mm headphone. With latest firmware, now supports BLF and LDAP. Finally BLF support! I've not done a great deal of Asterisk work in the last six months, but the lack of BLF support in the 942 was one of the few irritants with these phones. Not that the SPA-942 really has enough buttons to make proper use of the feature, but one or two is a whole hell of a lot better than none! ___ -- 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] Good comparisons on cheaper VOIP phones
Michael wrote: My experience with Grandstream is that are one of the better 'cheap' ones, but cheap non the less. I am yet to run into a worse IP phone than the Grandstreams - although having said that, I should say that I've always steered clear of most of the Chinese no-name brand phones. They're unstable, temperamental and upgrading the firmware is a crapshoot half the time since you never know what new bugs will be introduced and quite often you can't downgrade the firmware if you don't like the newer firmware. My suggestion would be to look at the Snom 300 (although they are very simplistic phones), the Polycom IP330 (I have a feeling the 320s don't support PoE) or the Linksys phones. I noted an earlier post saying that these phones were overpriced and designed to lock you in to Linksys gear - my experience has been completely different. The SPA-942 is quite cheap and integrates nicely with Asterisk. The SPA-962 is considerably more expensive - but considering the size of the colour LCD screen, they're not that badly priced. (as an aside, the button banks for the SPA-962 are one of the /cheapest/ available!) ___ -- 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] Good comparisons on cheaper VOIP phones
Michael wrote: I bought it. The SPA962 went on ebay within 3 months of me buying it. I have a few grandstream 286's I like to use for traveling and placing in remote areas of an installation. 3 months... that long? Again I'm surprised. I've had no problems at all with the Linksys phones connected to an Asterisk system. My list of irritants with the phone is pretty low - you can't use the line buttons as BLF buttons and localising tones is rather painful. They're not in the same class as Polycoms when it comes to hands-free (but then again, basically nothing else is) but the hands-free is quite usable. I have a Linksys SPA9000 IP PBX I want to quit. Mint condition with all packing etc. Nothing 'wrong' with it (except that it's a Linksys) I just hate proprietary stuff which is what the Linksys is. Ahh... that's a bit different. Yes, the SPA-9000s are an overpriced pain in the ass, but the phones certainly don't fall into the same category. ___ -- 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] Linksys SPA922 - hangup problem
dubravko caric wrote: Hi all, I'm testing Linksys SPA922 phone and I have strange issue. when call is finished on the phone I see CallEnded and normal silence for cca. 5 seconds and then I get fast busy for cca. 20 sec. So, this isn't automatic hangup as on other phones I have tried (Cisco 7940, grandstream, XLite,... ) and I have to manually hangup handset to finish a call. Is this normal behavior for this phones or have I misconfigured something (maybe on Asterisk, but other types of phones are working OK)? This is normal behaviour for the Linksys phones - it's either a carry over from their ATAs or a slightly misguided attempt to make phones behave similarly to analogue phones. It's one of the few, admittedly minor irritations these phones have. ___ -- 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] callcenter supervisor system
David fire wrote: hi i need an open source callcenter manager system like queuemetrics but opensource any one know any? i prefer to search before start a new one You'll be pushing to find something even close to QueueMetrics' quality available in open source. The closest I'm aware of is Vicidial, though if you only want a call centre statistics package, Vicidial doesn't really meet the requirements since it's focus is on being a predictive dialler. There are a couple of unfinished, unpolished packages that are around that don't even come /close/ to what is available through QueueMetrics. ___ -- 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] SendImage()
Philipp Kempgen wrote: SendImage() in 1.4: ---cut--- SendImage(filename): Sends an image on a channel. If the channel supports image transport but the image send fails, the channel will be hung up. Otherwise, the dialplan continues execution. The option string may contain the following character: 'j' -- jump to priority n+101 if the channel doesn't support image transport This application sets the following channel variable upon completion: SENDIMAGESTATUSThe status is the result of the attempt as a text string, one of OK | NOSUPPORT ---cut--- in 1.6: ---cut--- SendImage(filename): Sends an image on a channel. Result of transmission will be stored in SENDIMAGESTATUS channel variable: SUCCESS Transmission succeeded FAILURE Transmission failed UNSUPPORTED Image transmission not supported by channel ---cut--- Is there any reason to break backwards compatibility? Why is SUCCESS better than OK and UNSUPPORTED better than NOSUPPORT? IMHO there was no need to change anything except for adding the FAILURE return status. Might be a -dev question though This is typical of the criticism that has been levelled at Digium time and time and time again - making changes that don't really add any functionality, but break compatibility. I had a hell of a time migrating a couple of systems from 1.2 to 1.4 - so much that I have no plans at all in the near future of migrating them from 1.4 to 1.6. Even the comments made at the time suggesting a parsing tool be provided to point out where changes to dialplan code would be required got a nice idea response, but nothing has been forthcoming. This habit of breaking functionality for limited or no reason, plus making the results from functions far /less/ useful (note my previous complaints about the REALTIME() function) and more difficult to use is the biggest problem with Asterisk bar none. ___ -- 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] Upgrading Asterisk and FreePBX from 1.2 to 1.4
Carlos Chavez wrote: I have a new customer that wants to upgrade their Asterisk installation from 1.2.27 to 1.4.22. They use FreePBX for administration. Since there are many syntax and command changes from those versions of Asterisk, is there an easy way to convert the FreePBX configuration so it will work with the newer Asterisk? Unless you have a lot of custom dialplan components in there, the only thing you need to be sure of is that you are running FreePBX 2.3 (I believe - possibly 2.2) or later. If you are running a very old version of FreePBX, then you will need to upgrade it /before/ you upgrade to Asterisk 1.4. ___ -- 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] IAX2 client for eee pc 1000
Alex Balashov wrote: The solution for the problem of an IAX client is a SIP client. That's not a particularly good solution if you have a NAT between your client and Asterisk. IAX is still *much* easier to get working through a firewall. ___ -- 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] tired of midget packet received warnings
Russell Bryant wrote: On Nov 8, 2008, at 1:30 PM, Atis Lezdins wrote: Asterisk offers very much the same flexibility. You can disable specific log levels (for example warnings) in logger.conf or you can log everything to syslog, where filter out this specific message. Of course, there is always this method, which is an even easier way to disable this specific message: I would have thought logging to syslog and using the filter functions there would have been considerably easier than recompiling a patched version of Asterisk - particularly if precompiled versions are in use. ___ -- 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] tired of midget packet received warnings
Tzafrir Cohen wrote: On Sat, Nov 08, 2008 at 02:33:18PM +1100, Rob Hillis wrote: Maybe it's me, but I think that warning should be regarding a problem I can fix. Malformed network content does not neceserily fall under that definition. notice? Absolutely it does. Warnings of malformed packets are often (as mentioned above) symptomatic of network problems. Fix the network problem, fix the warning. As you saw in this case, this is a monitoring program that checks if somebody still listens on the UDP port. Would you teach nmap to try a valid IAX packet on every UDP port? How can you tell in advance that the port is IAX and not SIP? Or whatever UDP protocol? Why should the monitoring program care? Depends on how thorough you want the monitoring program to be. Personally if I were monitoring a service, I'd want to know that the service was responding the way you were expecting it to rather than blindly checking whether the port was open. However, one of my previous job was to monitor a large network that was running software that I would consider to be pretty badly broken and the fact that a port was open meant nothing more than the executable was still running - it was quite common for the software behind it to have gone into an infinite loop that promptly ignored all other data. I learnt to be incredibly paranoid if I wanted to be sure that everything was working the way it was supposed to be. UDP presents it's own challenges when it comes to monitoring anyway since there's no guarantee you'll get a reply from the other end. However, in the case of a program such as nmap, I take your point. Nmap is more interested in whether a port is open than whether the software is fully functional or not. ___ -- 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] tired of midget packet received warnings
Louis-David Mitterrand wrote: On Thu, Nov 06, 2008 at 08:42:52AM -0600, Kevin P. Fleming wrote: Your monitoring app is not sending valid IAX2 packets to the server. If it was sending a true IAX2 POKE, it would be a valid packet and wouldn't generate this warning. Could asterisk at least _not_ report this harmless, below-warning event when using a zero-verbose (asterisk -r) level? That would be nice and logical. Actually, I would have said that corrupt/bad IAX packsets *should* be reported and are *not* harmless. They're harmless in your instance because your monitoring application isn't functioning properly, but to anyone else they're likely to indicate either (a) a hacking attempt or (b) a fairly serious network problem. How about you fix your monitoring application to send a correct IAX2 POKE request? ___ -- 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] tired of midget packet received warnings
Tzafrir Cohen wrote: On Fri, Nov 07, 2008 at 09:29:20AM +, Tim Panton wrote: I'd take this warning seriously. It means that your monitoring app isn't monitoring what you think it is. I always want to know when I get malformed protocol packets in. It is always bad news, mostly either a misconfiguration (your case), an attack, (ie my firewall is not protecting this service) or a sign of a switch port going bad. Fix the cause not the symptom. Maybe it's me, but I think that warning should be regarding a problem I can fix. Malformed network content does not neceserily fall under that definition. notice? Absolutely it does. Warnings of malformed packets are often (as mentioned above) symptomatic of network problems. Fix the network problem, fix the warning. ___ -- 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] SPA-962 Time on Asterisk
Steve Anness wrote: Good Day, I have been tasked with fixing the time on our asterisk server. I am having a hard time finding documentation to tell my what asterisk uses to get its time information to push to phones (or a better question, where does the SPA-962 get its time information)? Asterisk does not push time to the phones. The phones themselves will contact an NTP server to obtain the correct time. Basically, I can go under the settings of the phone and change the offset to set the correct hour, but it is still about 4 minutes fast. So the SPA-962 has an offset option, but to offset it from what? The time on the asterisk server? That isn’t right because my asterisk server has the correct time. To offset from GMT? No because I am +6 from GMT not +2. I can physically set the time, but that isn’t the best solution when you have many phones, shouldn’t the phone be syncing with something? Any thoughts? I am not finding anything conclusive. Best practice is to configure ntpd as both a server and client on your Asterisk server and point your phones to it. That way your phones will have exactly the same time as your Asterisk server. ___ -- 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] Question regarding keywords in sip.conf/users.conf
Unfortunately RealTime isn't going to be an option - it's another level of configuration I want to avoid, but more importantly since I'm planning on being able to run these scripts on an Astlinux install, there won't always be a MySQL database available. If worst comes to worst, and the extra configuration included in sip.conf becomes a problem, I'll move it to another text-based config file - not my preferred option (since I'd like to keep everything close-to-hand) but not a major problem since I'm likely to need a separate config file for global configuration options anyway. Paul Hales wrote: It should ignore the keywords, but you will get lots of errors in the CLI. My guess is that if you put it all in a DB (and use realtime) you can probably do whatever you want. PaulH Rob Hillis wrote: Hi guys, I'm about to embark on a small (undoubtedly to get much larger) project to write a set of scripts to handle provisioning of phones - Snom to begin with, possibly with others (most likely Polycom and Linksys) to follow later. Since I want this script to handle *all* aspects of phone provisioning (such as BLF buttons and so on) I need a place to store data. My preference is to keep all phone related configuration in the one place - such as sip.conf or users.conf. How would having additional keywords (most likely with a prefix of some type to reduce the likelihood of conflicts with real keywords) in Asterisk's .conf files affect Asterisk? I would expect that Asterisk should ignore unknown keywords, but I'd rather check on this with those in the know first. Any insights? ___ -- 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 ___ -- 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] Question regarding keywords in sip.conf/users.conf
Barry L. Kline wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Rob Hillis wrote: Unfortunately RealTime isn't going to be an option - it's another level of configuration I want to avoid, but more importantly since I'm planning on being able to run these scripts on an Astlinux install, there won't always be a MySQL database available. If worst comes to worst, and the extra configuration included in sip.conf becomes a problem, I'll move it to another text-based config file - not my preferred option (since I'd like to keep everything close-to-hand) but not a major problem since I'm likely to need a separate config file for global configuration options anyway. You could still store them in sip.conf, just make each line a comment. e.g.: ;[myentry]keyword=value ;[myentry]keyword=value You could then search for ;[myentry] for your keywords and strip them off when you write out the real entries. I did think of that, but the idea of using something that's actually a comment as configuration seems fraught with danger - not to mention it being an awful hack. ___ -- 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] Question regarding keywords in sip.conf/users.conf
Hi guys, I'm about to embark on a small (undoubtedly to get much larger) project to write a set of scripts to handle provisioning of phones - Snom to begin with, possibly with others (most likely Polycom and Linksys) to follow later. Since I want this script to handle *all* aspects of phone provisioning (such as BLF buttons and so on) I need a place to store data. My preference is to keep all phone related configuration in the one place - such as sip.conf or users.conf. How would having additional keywords (most likely with a prefix of some type to reduce the likelihood of conflicts with real keywords) in Asterisk's .conf files affect Asterisk? I would expect that Asterisk should ignore unknown keywords, but I'd rather check on this with those in the know first. Any insights? ___ -- 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] Call problems
Emmanuel Pascal Bruno wrote: I have turned off firewall on the linux box, I have turned off firewall on the router I still have the same problem :-( Disabling firewalls is almost certainly going to ensure the problem persists. You need to ensure that all SIP and RTP ports are port-forwarded from your firewall to your Asterisk box. ___ -- 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] Call problems
Emmanuel Pascal Bruno wrote: I have a DID from IPKall.com which is forwarded to my asterisk box. Then this extension should call my ip phone using Dial application. Everything works fine, except when I pickup the phone, I can talk, the other party can hear me, but I cannot hear anything the person says on the ip phone. Then after a couple of seconds, the call hangs up. I don't know why. Here is the message I get: SIP/ipphone-09401f10 answered SIP/XX.XX.XXX.XX-09400918 -- Native bridging SIP/XX.XX.XXX.XX-09400918 and SIP/ipphone-09401f10 [Oct 30 13:38:12] WARNING[7290]: chan_sip.c:2787 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 102 (Critical Response) -- See doc/sip-retransmit.txt. [Oct 30 13:38:12] WARNING[7290]: chan_sip.c:2814 retrans_pkt: Hanging up call [EMAIL PROTECTED] - no reply to our critical packet (see doc/sip-retransmit.txt). == Spawn extension (ipkall, ipphone, 1) exited non-zero on 'SIP/XX.XX.XXX.XX-09400918' I am running asterisk 1.6 on CentOS Please help me fix this You likely have firewall issues since it appears that you are not receiving a response from the other end. Make sure you have *both* your SIP and RTP ports forwarded to your Asterisk box. ___ -- 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] Copy protection issues with G.729 codec in Solaris
Peter Galiovsky wrote: Does anyone have any idea what should I try next? Either contact Digium support directly or the people you bought the G729 license from. You're more likely to get the assistance you need in a shorter period from these people than this list. ___ -- 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] Decent Voip Phones for enterprise
Kev Szaszvari wrote: Hi there Our company is using the Linksys SPA-942 Phones, and they are pretty useless. They dont have any central management or provisioning, as well as a pretty bad interface. This is completely incorrect. Linksys SPA-942s *do* have the ability for central management. It's done via an XML file - the details on which you can get by opening your web browser and going to http://ip.of.your.phone/admin/spacfg.xml which will return the full configuration of your phone as it currently stands. As for the interface, whilst it's not the best I've seen, it's far from the worst. Can anyone reccomend any voip phones( Cisco, Polycom, SNOM ) that have * Central Management for all the phones (We dont mind if we have to buy the software to manage them) * Programable shortcut buttons, So i can program in on certian phones quick dials to queues. * Optional but bonus, The ability to have a shared address book accross the phones. Snom phones (320 and above) have all the features you ask for here. ___ -- 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] WebCall application
Tim Panton wrote: Does anybody know any free WebCall solution to let our customer call us directly via our web site? Any clue will be welcomed. Yep, take a look at our offering on www.phonefromhere.com A per-minute charge does not constitute a free solution. Please read requests before spouting off about your own products. ___ -- 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] Is there a way to test SIP credentials withoutmaking a call?
Eric Chamberlain wrote: I should have clarified, we're only making outbound calls, not inbound, so there is no registration. Is there a particular reason you /can't/ register? It would seem that registration would provide the functionality you require, even if you're only making outbound calls. ___ -- 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] Is there a way to test SIP credentials withoutmaking a call?
Eric Chamberlain wrote: Is there a particular reason you /can't/ register? It would seem that registration would provide the functionality you require, even if you're only making outbound calls. In the case of a server like Asterisk, wouldn't sending a register disrupt the flow of inbound calls until the UA that normally handles inbound calls re-registers? Are you using the same credentials as existing extensions to make calls from different extensions? That would seem to be a particularly bad idea. You should be configuring /one/ sip extension per SIP phone. Those extensions that handle outgoing calls only could be put in a different number range, or have a letter prefixed or suffixed to the extension, but you should /not/ be using one configured extension for two different purposes. ___ -- 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] Matching *, + and # in the dialplan
Tilghman Lesher wrote: Can someone suggest the best way to deal with this without resoring to a highly repetitive/iterative dialplan? Leif and I discussed something like this at Astricon 2008, and we came up with this patch: http://bugs.digium.com/view.php?id=13632 Nice! For those of us still using the old dialplan code versus AEL, this could potentially make the readability of extensions.conf much easier! Are there any plans to include this in a future release of Asterisk? ___ -- 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] make func_realtime work like app_realtime (1.6)
Wesley Haut wrote: Yell at me if you will, but I hate func_realtime - it's not very usable nor is it change-friendly (update your database and your dialplan completely breaks). I agree completely. As it stands, the REALTIME() function is nearly completely useless. If Asterisk had better string manipulation functionality, it would be /marginally/ better, though still not much good. A far better approach would be to allow you to specify the specific field you want to retrieve - the same way that you do for a write. /That/ would make the function many times more useful than it currently is. ___ -- 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] make func_realtime work like app_realtime (1.6)
Tilghman Lesher wrote: On Wednesday 08 October 2008 13:22:25 Rob Hillis wrote: Wesley Haut wrote: Yell at me if you will, but I hate func_realtime - it's not very usable nor is it change-friendly (update your database and your dialplan completely breaks). I agree completely. As it stands, the REALTIME() function is nearly completely useless. If Asterisk had better string manipulation functionality, it would be /marginally/ better, though still not much good. A far better approach would be to allow you to specify the specific field you want to retrieve - the same way that you do for a write. /That/ would make the function many times more useful than it currently is. What if I made it work with the HASH() dialplan function, similar to how func_odbc works? Keys are column names, values are the associated field value I can't say I'm familiar with this method, but a quick look at core show function HASH would seem to indicate that it would be much better than what we have at the moment. However, for those occasions where you only want to pull one variable out of realtime, I'd still like to have the option of specifying one field name to retrieve. Possibly the most useful method would be to return the single field if a field name was specified, otherwise return the array with all values in it. That would allow people to pick the method most suitable for them. ___ -- 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] How to add Callee's name into Dial command ?
Olivier wrote:
2008/10/3 Olivier [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
Hi,
When dialing a number, I use :
exten = _123X, 1, Dial (SIP/${EXTEN})
Then, I get TRYING and RINGING SIP messages which both include
this kind of line :
To: sip [EMAIL PROTECTED] mailto:[EMAIL PROTECTED];user=phone
Is it possible, configuring Asterisk 1.4, to get something like
this instead ?
To: John Doe sip [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED];user=phone
This way, I'm hoping to display callee's name beside (or instead
of) callee's number which would offer a double check for caller
which might be confusing extensions, for instance.
I tried this :
exten = _123X, 1, SIPAddHeader(To: Doe \sip [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]\;user=phone\)
but I still got :
To: sip [EMAIL PROTECTED] mailto:[EMAIL PROTECTED];user=phone
I must add I also tried without success :
exten = _123X, 1, Set(SIALPEERNAME=Doe)
Have you tried Set(CALLERID(name)=Doe)? This the normal method for
setting caller ID names in Asterisk.
___
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Re: [asterisk-users] sip clients for smart phones?
Babcock, Michael Alex wrote: windows smart phone v 6.0 example htc shadow is what i have. It has wifi abilitys. Googling for windows mobile sip yeilds a multitude of results. I'm sure one of them will point you in the right direction. ___ -- 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 Documentation now on voip-info.org Wiki
Josiah Bryan wrote: The script design supports plugin formatting as it stands. E.g. I can insert any formatting algorithm if anyone has any suggestions. Right now, the formatter script just does: #!/usr/bin/perl use strict; my $file = $ARGV[0]; print ~pp~\n; print `cat $file`; print ~/pp~\n; Any formatting can be added as desired - this was just a quick way to get the content online. Might I suggest including... print -=NOTE: These pages are automatically updated once per day/week/month/year/decade from the Asterisk subversion repository. Any changes made to this page will be automatically overwritten with the latest version from insert URL here.\n; ...at the beginning? May stop some nutters whining that you're continually overwriting their changes. ___ -- 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] How can Block a pri channel
Giorgio Incantalupo wrote: Hi, why do not you simply delete them from zapata.conf and restart your PBX? Because that simply doesn't acheive what he's wanting to achieve. On PRI circuits you can dynamically enable and disable circuits at the data-link level. Whether this can be achieved with Asterisk or not, I don't know. ___ -- 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 or 1.6
Joseph wrote: I need to upgrade my Asterisk, currently I'm using 1.2.27 from Gentoo portage but I think this version has a problem with RFC2833 DTMF signaling and I don't think there will be any newer version available anytime soon on portage. I need stable version, I'm using Asterisk mostly with ATA adapter (Linksys and Sipura); should I go to 1.6 or 1.4? By saying you need a stable version, you've answered your own question. 1.6 has not yet been released and therefore should not be considered to be production ready yet. Upgrade to version 1.6 at your own risk only. Some people have reported 1.6 to be very stable, others (such as myself) are still having occasional problems with it. For me, these issues aren't a great concern since it's a home system, but at least for the moment, I wouldn't consider running 1.6 in a production environment. ___ -- 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] How to check mailbox exists (Received SIP subscribe for peer without mailbox)
Olivier wrote: Hi, I'm receiving this : [Aug 28 02:25:16] NOTICE[5895] chan_sip.c: Received SIP subscribe for peer without mailbox: 9163 I've read this : http://lists.digium.com/pipermail/asterisk-users/2008-May/211701.html I typed this: asterisk -rx reload asterisk -rx voicemail show users ... and got : default9163 john doe 0 This default ... line is 100% similar to others. Chances are that the mailbox line in sip.conf for this extension doesn't include the correct mailbox context. Make sure that [EMAIL PROTECTED] is in the section of sip.conf for this extension and reload. ___ -- 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] How to check mailbox exists (Received SIP subscribe for peer without mailbox)
Olivier wrote: Now that root cause is found, would you say that warnings or CLI should have been different ? Obviously, MWI subscriptions must come from SIP hardphones (at least those supporting MWI feature). So in this case, Received SIP subscribe for peer without mailbox: 9163 rather means Asterisk is receiving from SIP/9163, subscriptions to MWI, but nothing in peer SIP/9163 settings is describing which mailbox should be the scope of such subscriptions. Do you agree ? Actually if you read the log entry, it can easily be interpreted that way. Log entries really are a delicate balancing act - you need to provide enough information to determine what the problem is without becoming too wordy. The only real way I could see to improve that log entry would be to say Received SIP subscribe for peer without *configured* mailbox: 9163 The log correctly identifies it as a notice since it's not an error condition. Any log entry longer than this would bloat log files even further than they already are - and Asterisk log files on even a moderately busy system are already about as easy to follow as your average plate of spaghetti. ___ -- 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 1.6 beta
VoIP Cyprus wrote: Can you share with me your experiences with Asterisk 1.6? Is it stable enough for commercial service? No. No matter how good some people may tell you it is, 1.6 is still beta software and software is rarely beta for no good reason. Don't even THINK about running 1.6 until it leaves beta and RC stage unless you are truly desperate for the features and are willing to accept random crashes, unusual behaviour and the possibility of things changing before the final release. The company I worked for up until June this year was still selling 1.2 systems until late April because we hadn't worked through all the changes and tested things fully. If your company will depend on your phone system for customer service, don't take the risk. ___ -- 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 Realtime pounds MySQL
Tilghman Lesher wrote: Given that this is the case, we may want to do one of the following: a) document that qualify=yes is incompatible with realtime, unless rtcachefriends is turned on, b) automatically disallow qualify=yes if the peer is realtime and caching is not turned on, or c) automatically cache realtime peers whose qualify field is set to yes. I am open to discussion and suggestions. Option a is certainly the least invasive, which doesn't change any behavior, so it's the default, but I think that enough people would consider this behavior a bug that we might change it. The only question that I have is, change it to what? My vote is option (c) with a note clearly documenting this. The other option, of course is only to cache the required fields for qualify to be set to yes - IP address and port? ___ -- 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] Problems with D-channel (PRI)
Jakub Arkon Syrek wrote: Hello, we have strange problem, till now everything was working fine, there where no problems with dial and answer calls. Yesterday our system crashed and we notice strange behavior. What type of event caused the box to crash? Given the fact that you've also mentioned a degraded RAID array, this is sounding very much like a power spike that may have damaged hardware. What type of card is installed in your machine? Have you spoken to the relevant support people for that card? ___ -- 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] OT - Which rackable case for mini-ITX boards ?
Olivier wrote: Hi, Though this is a bit off-topic on this list, I think this might interest those looking to build Asterisk appliances out of mini-ITX boards such as http://www.pcengines.ch/alix1c.htm. If you're interested in the ALIX type boards, there is a reseller in Australia that offers a rack-mount option. It's designed to house two of the boards, though it's not compulsorary. See http://www.yawarra.com.au. The products that would probably be of interest would be http://www.yawarra.com.au/product.php?productCode=HW-NT48-R and http://www.yawarra.com.au/product.php?productCode=HW-AX22-R. Most Asterisk users would be more interested in the first option I'm guessing. It's not the ALIX board - it's the net5501. The reason it would most likely be of more interest is that it has a PCI slot available to it. I'll add that I'm not affiliated with Yawarra in any way other than having been a satisfied customer in the past. I must also add that whilst I've got an ALIX box myself, I /haven't/ used it as an Asterisk server - mine is a router. ___ -- 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] Semi-OT Satellite?
Ken Williams wrote: We're entertaining moving our intranet to Hughes satelite for our remote locations. I'm curious if anyone with Asterisk servers has used satellite, and if so, is the latency an issue. My understanding is that you immediately introduce 250ms latency for travel time up and back down, however it is a much more direct connection then offered by traditional land lines. Everyone I'm aware of that has tried using VoIP over a satellite based internet connection has been /completely/ dissatisfied. A base ping time of 250ms is simply too high for calls to proceed naturally - you'll end up with people talking over the top of each other. The problem with be exacerbated if both ends are satellite based. ___ -- 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] BLF functionality
Dan Peters wrote: We have had Asterisk up and running for a while now and it works very well. Recently we tried to integrate a Linsys SPA962 with the associated SPA932 console. We can get the BLF lights to blink when a phone is ringing and we can get the BLF lights to go solid when that call is picked up. My question is about the BLF for the phone that placed the call. Is the BLF supposed to light up when the handset is picked up and a dial tone is heard? Right now that is not happening. The BLF lights only seem to operate for phones that are RECEIVING calls and not MAKING them. Asterisk can only detect when a call has been placed. Once the call has been made and the other end is ringing, is the calling extension showing up? If not, I've seen this problem before with Asterisk 1.4. You need to ensure that the following configuration items are set in sip.conf:- * call-limit must be set for every extension. Usually set this to a large arbitrary number such as 100. (or 1 if you don't want another call to be received when the extension is on the phone) * notifyringing should be set to yes. * If you're using RealTime, ensure you have both rtcachefriends and rtupdate set to yes. * limitonpeers must be set to yes if you're using type=peer. (this is the most likely culprit since it's missing from the default config files) ___ -- 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 Realtime Unregister
If a phone is unplugged, it's not likely to have time to send
notification of this to Asterisk before it powers off. There's nothing
you can add to your dialplan to overcome this, however you *can* set the
qualify parameter within sip.conf (or it's equivalent realtime table)
to overcome this.
See http://www.voip-info.org/wiki/view/Asterisk+sip+qualify for more
information. Short version is that configuring a qualify interval is
the equivalent of setting up a heartbeat between Asterisk and registered
devices configured with a qualify interval. If the heartbeat fails, the
phone's registration is suspended.
Nhadie wrote:
Hi,
I'm running asterisk realtime, i had prob when a user does not
unregister properly.
I tested with SPA942 and a PAP2, when phone is registered, i call using
the SPA using x-lite no problem, but when i unplugged the power, it does
not unregister properly, so asterisk think SPA942 is still registered,
when i call using x-lite, asterisk tries to call it.so it gets stuck at
[Aug 11 21:37:31] -- Called 102104
until it reached the timed out i set in the dialplan which is 30 secs
Dial(SIP/${EXTEN}|30|t|M(setmusiconhold,moh-${EXTEN}))
is there something i can add on my dialplan to first detect that the
user is not available, or maybe force unregister, anything that would
not make my dialplan to wait for 30 secs.
also i'm not using rtcachefriends, how would i know in the CLI which
user is registered? i tried sip prune but it shows me nothing
sip prune realtime peer all
No peers found to prune.
anyone experienced this?
thank you
regards.
nhadie
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