Re: [asterisk-users] DTMF not being heard correctly by far end conference system
Hi Thorsten Thanks very much, at this point my preference is rfc2833 but I will try some other options. The system is generating audible tones (that I can hear), although I think the audio is generated by the last sip device in the network so if thats so I don't have any control of it. Probably then I have to go to inband to get some control back, I am not sure what I lose from this, or change upstream provider (although the current provider works from a different system) Cheers Duncan On 12/01/2011, at 11:42 PM, Thorsten Göllner wrote: As far as I can remember you should take a look at the used codec and this here: http://www.voip-info.org/wiki/view/Asterisk+sip+dtmfmode Some codecs do not feel happy with some seetings for dtmfmode. Perhaps you may comapre these on your 2 boxes. -- _ -- 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] Usage Reports
Freepbx really needs its own list but it doesn't seem to have one But - if you have mysql setup and records being logged then the reports should show you usage on a daily, weekly, monthly level. Make sure you built asterisk with cdrs logged into mysql - its in the addons Cheers Duncan On 30/12/2010, at 8:36 PM, Ben Schorr wrote: We’re using FreePBX 2.8 and there is a Reports tab but it doesn’t seem to actually do anything. Is there some secret/trick to getting a report out of it that will tell us which extensions are placing calls? I’ve tried every query on the form that I can think of. Is the reporting disabled by default or ??? Any tips/pointers appreciated. Ben M. Schorr Chief Executive Officer __ Roland Schorr Tower www.rolandschorr.com b...@rolandschorr.com -- _ -- 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] Pop-up for MS Outlook 2007 recommended
I think there is a new version of Outcall, the pop up was pretty good, but the dialout wasn't ideal in Win 7 , and I believe thats fixed now with good integration with 2007 and 2010 http://code.google.com/p/outcall/ You can buy commercial options from Biocom - who make Outcall http://www.bicomsystems.com/products/C/P/319/288/ Cheers Duncan On 26/10/2010, at 8:24 AM, unsero...@aol.com wrote: Did you already check Bria? I have not tested it yet but it seems to be very powerful. Unfortunately there is no trial version available. If you will give it a try I would be very interested in your opinion. http://www.counterpath.com/bria-for-microsoft-outlook.html Oliver -Original Message- From: Bruce B bruceb...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Mon, Oct 25, 2010 9:10 pm Subject: Re: [asterisk-users] Pop-up for MS Outlook 2007 recommended Great suggestion but unfortunately for this client a proven technology is needed and we don't mind paying a bit for it so once the time is available we might do this the way you suggested. Thanks On Mon, Oct 25, 2010 at 2:20 PM, Danny Nicholas da...@debsinc.com wrote: From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce B Sent: Monday, October 25, 2010 1:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Pop-up for MS Outlook 2007 recommended Hi Everyone, Which paid or unpaid commercial plugin is available out there for Asterisk that would do Outlook contacts pop-up that is proven to work great with MS Outlook 2007 and Asterisk 1.6. It would be a bonus to do Dialout as well through the Outlook. Thanks, Bruce Not specifically what you are looking for, but it is very simple to use Apache/Ajax to make AMI links to launch calls from anywhere. I would invest 30-240 minutes into this method before bothering with the other stuff that is out there. Also, will make it easier when you eventually jump to 1.8/1.10. -- _ -- 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 -- _ -- 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] .call files with application/data are not generating correct CDR
You often don't get cdrs or at least useful ones unless you run the call files through a Local channel You maybe already doing this Can you check the Master.csv and see if it also is recorded incorrectly there. Is this just an issue with mysql cdrs or something else. In my setups which use freepbx I haven't had an issue with cdrs and call files if using Local channels to call Cheers Duncan On 23/08/2010, at 2:11 AM, Andy Beak wrote: Hi, The exact problem that I'm experiencing is described at http://www.spinics.net/lists/asterisk/msg122364.html in an earlier posting to the mailing list, but I could find no replies to it. I installed Asterisk using Ubuntu's apt-get and then fixed the mysql conf (which doesn't load if you use the default apt-get install asterisk-mysql) by building it from scratch. I'm using Asterisk as an automated voice messaging system so need to be able to dynamically make .call files which point to different mp3 files. My calls are now being logged to the mysql database but even if I answer a call it still logs as Not Answered with a duration of zero. Setting unanswered to either yes or no makes no difference in cdr.conf - the call is still logged as Not Answered if I pick it up. Really the only way around this I can see is to check the lastapp field instead of the disposition. Lastapp is set to Dial if the call was really not answered and MP3Player if the call was answered. I see that there is a known bug in Asterisk and it is suggested to use extension.conf to set up a context rather than using call files. The problem is that I need to be able to change the MP3 that is played. Has anybody managed to solve this problem? Thanks, Andy -- _ -- 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] Opensource Speech recognition for Asterisk
The Lumenvox works fine in my limited use, easy to setup, good dictionary options but it always depends on your circumstance. http://www.lumenvox.com/partners/digium/Asterisk.aspx Most of it is being really careful in planning the customer experience. The technology is secondary to the business analysis focussing on why and what the caller wants and making the most easy and efficient method of getting them there. Voice recognition is a pain for people with accents and poor lines and when people have written bad call flows but by making sure you get someone to an operator really quickly if you can't work out what they said then you can alleviate a few issues. The primary advantage of voice recognition is to give more choice to the caller and route them through more quickly. If you can't do that or don't need that complexity then don't use it Cheers Duncan On 22/08/2010, at 11:09 AM, Zeeshan Zakaria wrote: Then may be these big multi-billion dollar corporations should use one of them, with whom we all deal regarding various services, and who put us through these voice recognition time-wasting activity in a hope that the poor caller will eventually give up, or will wait painfully long until one of their agent will get time to attend call in person. Your experience could be different and better then most, and you have certainly complete right of your own opinion. Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-08-21 6:57 PM, Paul Belanger paul.belan...@polybeacon.com wrote: On Sat, Aug 21, 2010 at 6:21 PM, Zeeshan Zakaria zisha...@gmail.com wrote: I yet have to see ANY... I disagree, while not Open Source like the OP requested, both Nuance and Microsoft Speech Server are nothing to laugh at. -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Pr... -- _ -- 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] Call agent when queue is empty and there is a voicemail left
So in broad terms You need to know when the queue is empty, and when there is voicemail (in a generic queue mailbox presumably) and also that you haven't already delivered the voicemail, and probably that when you deliver the mail its been successfully been heard and actioned. Are you also emailing the voicemail as what happens once you have played it - do you delete it assuming its actioned or leave it there, So you need a trigger when the queue is empty, could be an AGI checking on hangup or an event from the manager to check the queue count. I haven't done much with AGIs but would think that using a DeadAGI on hangup (in 'C' as Steve suggests for speed) you could check queue size, check voicemail then trigger an event that triggers the calls. You may not want the call direct in the AGI as you want that to finish but you could either use an async agi or write a call file or use the manager interface Do you actually want them to get the voicemail prompts or do you just want to find the voicemail messages and play them directly. So I don't know if it helped or added more questions but good luck Cheers Duncan On 12/08/2010, at 8:42 PM, Jonas Kellens wrote: Anyone has an idea of implementation ?! Jonas. On 08/10/2010 09:04 AM, Jonas Kellens wrote: Hello list, situation : 1. incoming calls come into a queue 2. there is 1 agent logged in into the queue (not always the same agent) 3. when the caller is in the queue, he has the option to quit the queue and leave a voicemail message what I want : when there are no more callers in the queue with who the agent has to be connected, Asterisk needs to call the agent and connect him with his voicemail IF there is a voicemail left. The caller can press 8 to exit the queue and go to the part in the dialplan that does voicemail. I thought about creating a call file when a voicemail is left. But how will I know when to connect the agent with the voicemail ? How will I know the queue is empty and so the call file may be executed ?! I'm not asking for dialplan examples, just a description of how you guys see the above implementation. Kind regards, Jonas. -- _ -- 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] AMD setup in Astersik
You can include the label of the context in the custom area instead of including a different context i.e. [ext-queues](+) http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf Not sure if it affects the order of processing or if that matters Cheers Duncan On 8/08/2010, at 7:43 AM, Rushikesh wrote: Hi, You can use /etc/asterisk/extensions_override_freepbx.conf file if you dont want your dialplan to get overridden. Regards, Rishi On Saturday 07 August 2010 07:36 PM, Tino wrote: In my Asterisk server following things have been done to detect answering machines before the answered call connects to the agents in queue. In extension_additional.conf == [ext-queues] include = ext-queues-custom exten = 5000,20,Macro(user-callerid,); changed the priority to 20 ... == In extension_custom.conf added following amd dialplan === [ext-queues-custom] exten = 5000,1,Answer() exten = 5000,n,AMD(2500|1500|300|5000|120|50|4|384) exten = 5000,n,GotoIf($[${AMDSTATUS} = MACHINE]?machine:human) exten = 5000,n(machine),Verbose(3, We found an answring machine) exten = 5000,n,Set(AMP=${CALLERID(num)}) exten = 5000,n,Set(date=${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}) exten = 5000,n,System(not showing the actual command) exten = 5000,n,Goto(ext-queues,5000,20) exten = 5000,n(human),Verbose(3, We've got a human on the line!) exten = 5000,n,Goto(ext-queues,5000,20) === This setup is working fine but the problem is that when i reload freepbx, extension_additional.conf will go to its original form and the changes made will be lost. Is there any way to make the changes in extension_additional.conf conf permanent . Or is there any alternative method for this ? -- _ -- 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] FYI: Seen the 2600Hz announcement?
No its a split FreePBX is still the same, V3 is still the same, this is a fork from some guys who had got involved (or maybe paid some money) Cheers Duncan On 4/08/2010, at 2:56 AM, Tzafrir Cohen wrote: On Tue, Aug 03, 2010 at 02:28:15PM +0100, Alan Lord (News) wrote: http://gigaom.com/2010/08/03/2600hz-project/ So practically FreePBX V3 was renmed 2600Hz / BlueBox ? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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] dial extension and play sound file from shell on asterisk server?
Have a look at the call files examples of voipinfo http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out Its not too hard to do what you want Cheers Duncan On 8/04/2010, at 11:00 PM, Brian J. Murrell wrote: I want to use Asterisk as a general message delivery system here. That is, I want to be able to have a (shell, perl, etc.) script on my Asterisk server dial an extension, wait for it to be answered and then play a sound file and then hang up, or even wait for a response or reactions to some IVR. Certainly if I had a SIP library, I could have the script simply look like a SIP extension but that seems like it should be an unnecessary added complication. Instead I am looking for a more direct API to Asterisk which allows a process to interact like a phone but not actually be one. Is this at all possible? Anyone done anything like this? b. -- _ -- 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] Asterisk system for church call center
Hi Frank I have found Freepbx on top of Asterisk a good solution for the church I look after and the rest of my customers, the callcentre functions you need are built in it and if they have someone technical then they can expand what they are doing It has both queues and ring groups (which are often all they need) I would imagine you just send them to an IVR or mailbox to ask for their name and details then move them into a higher priority second queue Elastix has Sugar in it I believe and looks okay but I like debian/ubuntu as a base distribution rather than Centos so haven't gone that way. However most of my customers still struggle with the concepts involved in telephony so while they are happy to look at it while I am there they forget quickly how to drive it or lose their nerve, especially the church which is faith focussed rather than tech focussed ;-) Because of that I think you want the easiest system for you to maintain remotely and elastix or freepbx is pretty easy. It also allows you to say this is available, this is not which is useful in narrowing down requirements. Cheers Duncan On 30/03/2010, at 9:58 AM, Frank Church wrote: I have been asked by my church to recommend a VoIP system which can do the following. They do internet radio shows which are sometimes broadcast on radio. They are looking for a system which does the following for about 5 agents, exactly as they have described it. 1. Take incoming calls 2. Put them on hold if there is no one to handle the call immediately, or transfer them to an available agent 3. Take down their details, and number, (if this can be retrieved and saved from the caller id, thats better) 4. Get them to hold on after taking their details if they still want to hold 5. Call them back when the backlog is cleared up. I have a fairly good grasp of the hardware and programming part of Asterisk, having compiled it more than a few times and implemented A2Billing phone card and call shop system with it. But the type of software suited to the Call Center side is where my knowledge gap lies. I am looking for solutions based on the usual Asterisk distributions like AsteriskNow, trixbox, elastix etc, whether ready packaged or requiring additional customization. The matter of whether they will use soft phones, or regular phones with headsets is also something to consider. Soft phones with good GUI's may be preferred if more cost effective for them, although my personal preferences are with hard phones. Any recommendations - the ease of software for the end users is the main thing for me, and integration with the database for taking customers details is the main thing for me. One of the distributions with SugarCRM comes to mind here. Sorry for cross-posting, but ready made and commercially supported systems are not ruled out, if they come within their budget. Regards Frank Church -- _ -- 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] No RTP from asterisk?
On 1/03/2010, at 2:41 PM, Peter Serwe wrote: I checked the firewall, iptables -L showed no rules whatsoever. No other traffic has indicated it was blocked, iptables was set in allow all everywhere mode. I went ahead and turned it off, still don't have RTP. No audio either direction via lines registered. G729 is completely disabled from all trunk groups and users, only using G711 at this point. Peter The asterisk rtp debug should show if asterisk is sending audio or receiving packets but its not nearly as useful as tcpdump. tcpdump udp port 5060 -s0 -A will give you all the SIP. But just dumping all traffic between asterisk and the host will give you a view on RTP - you should see it take off when a call is setup if its not blocked You should see a SIP Invite to setup a call with the audio destination - this should be your asterisk box and the far end depending on who is doing what. You should look at the address and also whether both sides are providing a mutually acceptable audio formats. If there are no agreed audio formats you won't get rtp. The c= in the session setup indicates the addresses each site is using for media. Cheers Duncan -- _ -- 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] Unable to open pid file '/var/run/asterisk/asterisk.pid': No such file or directory
The other way on Debian/Ubuntu is just to test the existence of the dir and create it if needed If you add this to the /etc/init.d/asterisk near the start you should be fine if ! [ -d /var/run/asterisk ] ; then mkdir /var/run/asterisk chown $AST_USER.$AST_GROUP /var/run/asterisk exit 0 fi Set the ownership as required Cheers Duncan On 11/02/2010, at 7:50 AM, Brian wrote: On Wed, 2010-02-10 at 11:24 -0600, Jason Parker wrote: Brian wrote: Each time the server is rebooted Asterisk duly deletes the manually created /var/run/asterisk directory - quite why it does this I just don't know - perhaps it is a bug? Your assumption is incorrect. Some Linux distributions will empty /var/run/ on boot, just as they do with /tmp/. Thanks Jason - that had never dawned on me, but I've just tested it and indeed it does. I do believe you're right, however, in suggesting that there is a bug in Asterisk. It appears that Asterisk creates /var/run/asterisk/ during install and assumes that it will always exist. Agreed - that would make sense that by default it thinks the directory is there. The workaround / fix is to take out the (!) from /etc/asterisk/asterisk.conf and allowing the default setting of: astrundir = /var/run to come into play. It then puts the .pid and .ctl in the root of /var/run Some of the sample init scripts (Debian) create that directory before starting Asterisk. This should be done in all of them (or in Asterisk itself, maybe?). The one I had didn't - but I could have added it. I just wanted to be sure I was doing the right thing. Please report an issue on http://issues.asterisk.org/ Done - but I'm a bit embarrassed as it seems so trivial. Thank you for your help. -- _ -- 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] Mitel integration
Having looked at the outputs into PMS they are very simple stop start records. Line by line text that can easily be recreated. They have about 4-5 fields, origin number, destination, time of call, duration, or similar things Usually they go out via a serial port or TCP port expecting a terminal to receive them so plugging into them will quickly show you what you need. Its not that you need to match the Mitel, you need to match the PMS. Best to talk to them but I have looked at it for a couple of customers who are still deciding and helped them fix their PBXs when they broke and its pretty straight forward. You just need to be able to output to either a serial or TCP port . Cheers Duncan On 28/01/2010, at 6:01 AM, Jeff LaCoursiere wrote: On Wed, 27 Jan 2010, Mark Wiater wrote: the mitel 3300 sends SMDR on TCP 1752. It spews software and hardware logs in the same manner, different ports. This particular model (need to get the model number) has a serial connection. I'm all for putting a serial sniffer between them (if they let me!), but was really hoping someone had already done this and could give me a headstart. I'll investigate the ethernet options, though, as that would make more sense anyway! If the PMS will talk over ethernet I'll try to pretend to be a 3300. Cheers, j On 1/27/2010 11:00 AM, Steve Howes said: On 27 Jan 2010, at 15:48, Jeff LaCoursiere wrote: Sounds good to me, but without the spec I'm stuck in a catch 22! tcpdump? (assuming IP). Bet its fairly simple plain text or something. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Upgrading an AA50 and finding zap drivers - lack of Digium support
Hi there I have a client who has an AA50 from DIgium. I am really challenged getting any support as the client doesn't have any of the original registration or subscription info, someone did the install and left without any records. I thought okay we can ask Digium, but you can't get help wthout registering your product and you can't register your product without your subscription info. I did get one response which was to email customer services and eventually found an email address for them but that seems to have fallen on deaf ears. Perhaps my expectations are too high but it was an email a week ago and no response, not even to say go away. Its really disappointing and amazingly hard to believe such a system exists. Perhaps no one else in the world loses their registration info. I need to upgrade the AA50 to the latest firmware, and just get some general support for the setup, as it doesn't seem to have picked up Zap and I think perhaps the CF card has died. Does anyone have any pointers as who to talk to. I see lots of digium people on the list so I am hoping someone can help me. Thanks very much Merry Xmas Cheers Duncan ___ -- 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 an AA50 and finding zap drivers - lack of Digium support
Thanks very much Kevin But its not that clear - in fact the support email address isn't listed on the support page - everything leads to logging in with your registered product. Its incredibly frustrating and I recommend you try looking and seeing how it works for yourself if you haven't got your registration details. Having some method to handle missing details would have sorted this all out but I get stuck in a big loop. You are probably right I should have called but I would have thought email would be possible at least. It wasn't a rush for a while. Also I am not in the US so I don't tend to call overseas as standard as most companies have ways of getting to support and need emails with details anyway. I agree a 4 day holiday is approaching but a week ago it wasn't and apart from an automated response not even a whiff of a reply Cheers Duncan On 24/12/2009, at 9:16 AM, Kevin P. Fleming wrote: Duncan Turnbull wrote: I did get one response which was to email customer services and eventually found an email address for them but that seems to have fallen on deaf ears. Perhaps my expectations are too high but it was an email a week ago and no response, not even to say go away. The contact information for Digium's support department is clearly listed on our web site. There are multiple methods, including just using the plain old PSTN to call us :-) Its really disappointing and amazingly hard to believe such a system exists. Perhaps no one else in the world loses their registration info. I need to upgrade the AA50 to the latest firmware, and just get some general support for the setup, as it doesn't seem to have picked up Zap and I think perhaps the CF card has died. Does anyone have any pointers as who to talk to. I see lots of digium people on the list so I am hoping someone can help me. Given the serial number of the unit the support department will be able to research the registration and get you squared away; the only issue is that we are rapidly approaching a four-day holiday weekend at Digium so you would likely not get any response until early next week. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- 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] GUI for hunt groups?
Freepbx comes with setup of ring groups and queues with different hunt strategies Also it has Flash Operator Panel which gives you the state of the system in real time graphical format No money - just a small bit of installation time and learning how to use it Cheers Duncan Ken D'Ambrosio wrote: Hi, all. I've got an Asterisk box installed that I'd really like to leverage -- and installing a GUI for hunt groups would be awesome. So long as I can have a trial copy, I could even pay money. It would have to be able to make use of both SIP and ZAP extensions. Suggestions? (Note: I wouldn't much care about the GUI, myself, but my boss is all over one.) Thanks! -Ken ___ -- 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] Today's problem: Inbound call routing
Usually that message comes up because the caller is anonymous and freepbx doesn't like anonymous calls by default. There is an option to accept anonymous calls, or set the incoming trunk to accept calls from the specific IP address Of course it could be something else Cheers Duncan Ben Schorr wrote: Sorry, I'm brand new at Asterisk (and/or FreePBX). I'm going to have to figure out what all those things are before I can show them. I'll have to get back to you. Ben M. Schorr Chief Executive Officer __ Roland Schorr Tower www.rolandschorr.com b...@rolandschorr.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Tzafrir Cohen Sent: Friday, October 09, 2009 9:54 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Today's problem: Inbound call routing On Fri, Oct 09, 2009 at 06:15:43AM -1000, Ben Schorr wrote: -Original Message- Too simple, apparently, when I dial the number the caller gets a recording that it's a non-working number and this is what I see in the CLI: Extension '8085255935' in context 'default' from '808xxx' does not exist. Rejecting call on channel 0/1, span 1 That is a pretty clear error message. Yes, I thought so. But how do I fix it? So...other than creating the inbound route and assigning it to an extension I apparently have to do something else. Any suggestions as to what that might be? You manage your dialplan with FreePBX. This mailing list supports Asterisk. I have no problem with questions about FreePBX systems. But they should also be phrased as Asterisk questions. This is a FreePBX question. I see, so this isn't an Asterisk problem it's a FreePBX problem? Creating an inbound route is FreePBX speak. This is a FreePBX question. Please ask an Asterisk question. For instance, show a dialplan trace, show the respective dialplan, show the respective channel configuration. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- 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 ___ -- 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 ___ -- 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] Upgrading Asterisk in Trixbox installation
I am a big fan of ubuntu LTS and freepbx and recently I saw mention of a custom module to add auto configuring endpoints for linksys (but i cna't find it again right now) Trixbox had too much stuff whereas the source install of just what you want is nice and clean Cheers Duncan Jeff LaCoursiere wrote: On Mon, 31 Aug 2009, ilker Aktuna wrote: Thank you. That was quick and helpful :) Then I'll just make and make install What should I backup, in case of rollback requirement ? That's a bit tougher. At the least /usr/lib/asterisk/modules, /etc/asterisk, and /usr/sbin/asterisk... someone else may need to chime in here... I've always been a fan of trixbox, and I have done a lot of installations, but when it comes down to it all I really want it for is for a quick installations of asterisk and FreePBX. I don't think I actually use any of the trixbox-only features. I've also been enamored with Ubuntu of late, and have dumped CentOS. YMMV, but you might consider starting over with a clean build of the linux of your choice, and doing asterisk + addons + FreePBX from source. j Thanks. - Original Message - From: Jeff LaCoursiere j...@jeff.net To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, August 31, 2009 11:15 PM Subject: Re: [asterisk-users] Upgrading Asterisk in Trixbox installation On Mon, 31 Aug 2009, ilker Aktuna wrote: Hi, My Trixbox 2.8.0.1 installation includes the following Asterik version: 1.6.0.9-samy-r27 I am having some problems with it and I think they might be solved if I use the latest Asterisk version. Is it a good idea to update Asterisk in Trixbox externally ? I've done it in the 1.4 branch. Is it safe ? Should be, as long as you stay within the same branch. That being the case, I would stick with 1.6.0.14 if I were you. Make sure you don't make samples :) j If so, which version should I prefer ? 1.6.1.5 or 1.6.0.14 ? Thanks, ilker ___ -- 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 ___ -- 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 ___ -- 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 ___ -- 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] outbound calls not ringing
Generally with FreePBX the ring options are set in the General Options - you can set the Dial options which are normally tr, but I guess that isn't working for you. The SIP files you could edit would have custom in their name, otherwise your changes will be overwritten when you reload freepbx You could put this in sip_general_custom.conf which will be included Cheers Duncan John A. Sullivan III wrote: Oops! - You're using FreePBX - someone who knows more about FreePBX will have to help you as I don't. May I also suggest that you bottom post in future responses rather than top post; that makes it a little easier to follow. Good luck - John On Wed, 2009-08-19 at 16:59 +, Ott Rose wrote: here is my sip.conf. i don't see it. ;; ; Do NOT edit this file as it is auto-generated by FreePBX. All modifications to ; ; this file must be done via the web gui. There are alternative files to make; ; custom modifications, details at: http://freepbx.org/configuration_files ; ;; ; [general] ; These files will all be included in the [general] context ; #include sip_general_additional.conf ;sip_general_custom.conf is the proper file location for placing any sip general ;options that you might need set. For example: enable and force the sip jitterbuffer. ;If these settings are desired they should be set the sip_general_custom.conf file. ; ; jbenable=yes ; jbforce=yes ; ;It is also the proper place to add the lines needed for sip nat'ing when going ;through a firewall. For nat'ing you'd need to add the following lines: ; nat=yes , externip= , localhost= , and optionally fromdomain= . ; #include sip_general_custom.conf ;sip_nat.conf is here for legacy support reasons and for those that upgrade ;from previous versions. If you have this file with lines in it please make ;sure they are not duplicated in sip_general_custom.conf, if so remove them ;from sip_nat.conf as sip_general_custom.conf will have precedence. #include sip_nat.conf ;sip_registrations_custom.conf is for any customizations you might need to do to ;the automatically generated registrations that FreePBX makes. ; #include sip_registrations_custom.conf #include sip_registrations.conf ; These files should all be expected to come after the [general] context ; #include sip_custom.conf #include sip_additional.conf ;sip_custom_post.conf If you have extra parameters that are needed for a ;extension to work to for example, those go here. So you have extension ;1000 defined in your system you start by creating a line [1000](+) in this ;file. Then on the next line add the extra parameter that is needed. ;When the sip.conf is loaded it will append your additions to the end of ;that extension. ; #include sip_custom_post.conf From: jsulli...@opensourcedevel.com To: asterisk-users@lists.digium.com Date: Wed, 19 Aug 2009 12:17:15 -0400 Subject: Re: [asterisk-users] outbound calls not ringing sip.conf On Wed, 2009-08-19 at 15:55 +, Ott Rose wrote: we are using Aastra 57i i don't see that setting. where is it at? From: jsulli...@opensourcedevel.com To: asterisk-users@lists.digium.com Date: Wed, 19 Aug 2009 11:07:21 -0400 Subject: Re: [asterisk-users] outbound calls not ringing On Wed, 2009-08-19 at 13:54 +, Ott Rose wrote: I put a post on here about my issues with outbound calls not ringing but i haven't resolved it. so i am trying again. When i dial any outside number i dont get a ring tone at all. when the person picks up and starts to talk i can hear them fine. it sounds great. How do I start to troubleshot this? snip What type of phones are giving you the problem? If I recall correctly, our SIP phones had this problem depending on how the destination handled signaling. We resolved it by adding progressinband=no (as opposed to the default never - at least I think it is the default) but this produces the problem of duplicate ring tones at times. Hope this helps - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- 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] Skype for Asterisk???
I am using the beta and its pretty good for remote access for clients It would help if they had some discount structure for volume Cheers Duncan Pascal Bruno wrote: Not sure if anybody noticed, but it seems like Skype For Asterisk is out. $66 per channels, pretty pricey http://store.digium.com/productview.php?product_code=1SFA0001 ___ -- 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 ___ -- 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] Call File Channel
If you use a Local channel to dial it then it will fall under the same rules Channel: Local/numbertod...@the-context-you-want This gets a CDR produced, it does pay to check everything works the same but it should be fine Cheers Duncan David Gibbons wrote: Context: is what the call is dumped into after it is answered, at extension Extension:. I don’t think it’s related to how the call is placed. I can dial the local extension SIP/170 but I’m not sure where that gets me. Basically I want to have the same failover that I have for all other outgoing calls on these automatic calls… Thanks Dave *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Danny Nicholas *Sent:* Wednesday, August 12, 2009 5:17 PM *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* Re: [asterisk-users] Call File Channel Ok. Here’s how you would do that: Channel: SIP/170 (some local extension) CallerID: SIP/104 (another local extension) MaxRetries: 1 WaitTime: 60 retryTime: 5 Context: your_context Extension: s This should create an extension call using your context. The context can then dial out as you write it. *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *David Gibbons *Sent:* Wednesday, August 12, 2009 4:10 PM *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* Re: [asterisk-users] Call File Channel Thanks Danny, I do have a dial cmd with multiple arguments in my normal outgoing context. I guess my question really is: How do I tell the call file using “Channel: XXX” to use my outgoing context instead of Zap/g1/xx or sip/trunk_x/xx directly? -Dave *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Danny Nicholas *Sent:* Wednesday, August 12, 2009 5:05 PM *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* Re: [asterisk-users] Call File Channel Exten = s,1,Dial(SIP/trunk_x/#1SIP/trunk_y/#2ZAP/g1/#3,60) *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *David Gibbons *Sent:* Wednesday, August 12, 2009 3:59 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Call File Channel I know I’m missing something here (been a long day)… How can I specify more than one channel in a call file? I want to dial SIP/trunk_x and fail to SIP/trunk_y and fail to ZAP/g1… Thanks Dave ___ -- 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 ___ -- 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] Anonymous Connection form IP to use specific Context
If you create a peer definition and put the host address in it and the context you want it to go to you should be fine Cheers Duncan David Klaverstyn wrote: Hi All, I never saw a reply to this question. Is anyone able to assist? Regards David. *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *David Klaverstyn *Sent:* Friday, 19 June 2009 2:28 PM *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' *Subject:* [asterisk-users] Anonymous Connection form IP to use specific Context Hi All, How can I force an anonymous SIP connection from a certain IP address to use a specific context rather than the default one defined in sip.conf. I am using Asterisk 1.6.0.9 Regards *David Klaverstyn* ___ -- 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] about monitored calls storing
Trixbox I think uses FreePBX FreePbx has an option for each extension to set it to record all calls. It will record the extension in the file name and you can view it through the recordings app if you want a web view. There are all stored in a common dir /var/spool/asterisk/monitor - you can probably mod the code for the recording if you want more info in the filename Cheers Duncan peace keeper wrote: Hello all, how can I possibly make the monitoring for all calls through the asterisk, and for those file to be stored with the name of the initiator, in additional to know to whom this call is going, could this functionality be implemented via configurations! in other words, could I configure the asterisk so that the administrator to be able to hear calls coming from who going to whom, as a having a record for each call, I am using trixbox v2.6.2.1 should that functionality be implemented by an external application , such as one written using asterisk-java !!! any help is appreciated? 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] how to sniff RTP and SIP traffic only
For Linux use tcpdump on the host you are after tcpdump udp and port 5060 or portrange 1-16000 -s0 -i eth0 where 5060 is your SIP port and 1-16000 are your rtp ranges -s0 means snap length of 0 so capture all the packet rather than cutting off at a point And refine it by adding the host you are targetting and -w to write to a file. Then you can import the file in wireshark and use the voip utlities to listen to it fairly easily or use tcpdump -r to read it back and clean it out a bit more Cheers Duncan Xavier Cardil wrote: Hi, do somebody knows how to sniff RTP and SIP traffic only for a faster debugging ? 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 ___ -- 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] GSM mobile trunks
Yip the VoiceBlue SIP units are very good but a bit pricey Gordon Henderson wrote: On Tue, 23 Jun 2009, Sasa Bobek wrote: Hi all, We have been planing for a long time to set up GSM mobile trunks for termination, and were planing on going with analog GSM adapters connected to a VoIP gateway. Should we be concerned with such a set-up as far as voice quality and other issues are concerned? Any experiences with GSM terminal chipsets? Why not SIP based GSM devices? e.g. Portech? Gordon ___ -- 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 and openvpn and sip
Usually this is a routing error with openvpn setup and asterisk thinking it needs to route someway other than the vpn. If the originating packets have an external ip address asterisk might send them back out another route Have a look using tcpdump on the server to see where the returned packets are destined Cheers Duncan Giorgio Incantalupo wrote: Hi all, I'm trying to connect one phone to a remote asterisk server via openvpn. First of all, I put the vpn server on the box hosting asterisk and the vpn client on another box, both with public ips. Then I set the client ip as my phone IP gateway and the remote pbx ip as the registrar and outbound proxy. I see in the phone log register packets are sent but nothing in return. Asterisk console shows it tries to give back the packets but they seem to be lost somewhere. I made some tests with my pc setting its gateway with the vpn client IP and I can reach the pbx machine (ping, ssh,...) but sipsak gets no response. It seems ping and ssh response packets are correctly routed but sip packets aren't. I tried to set nat=yes in sip.conf but without result. Is there any asterisk parameter to set to make it work with openvpn? Any help really appreciated. Thank you. Giorgio ___ -- 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] Automatic Calling Feature?
Not too hard to do, you can have a script generate a list of call files which automatically ring the callers in the list and play a message http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out Cheers Duncan Christopher Stamper wrote: Right now, my organization is using a commercial service (OneCallNow.com), that gives telephone notifications to all numbers in a predefined list. Example: -Admin records a voice message -Service calls each number in the list, and plays the message back to them It's a pretty handy service, albeit a bit pricey. I've been wondering if Asterisk could do this for me? I don't really want to have to write scripts, but it would be great if it's already a feature. I don't have an Asterisk PBX running yet, but when I do it will probably have multiple T1 PRI lines, making it possible to dial all these numbers (100+) in a reasonable amount of time. Anyone know of a way to do this? -- Christopher Stamper Email: christopherstam...@gmail.com mailto:christopherstam...@gmail.com Web: http://tinyurl.com/2ooncg gTalk: http://tinyurl.com/6e359r Skype: cdstamper ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Best way to get 60+ analogue extensions.
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. I can see the Digium AEX2400 with 24 lines, physically they are all very deep, if I had 3 of these in a server it would seem straight forward assuming the motherboard doesn't haven't anything get in the way Equally the Digium TDM2400P supports 24 lines and physically requires similar space The Sangoma A400 provides 24 ports but uses two slots, having 3 of these in a server looks like I need to pick the server carefully. 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. I am just interested in any recommendations for server hardware and card combinations that are currently in use. Also if anyone has provided call data out to the RMS system ( http://www.rms-global.com/Our-Products/RMS-Hotel/ ) I would be keen to hear how it worked. Thanks very much Cheers Duncan ___ -- 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] Best way to get 60+ analogue extensions.
Thanks very much Rob Stephen The channel banks look good. I am not sure if they are easily availble in NZ but we can get some in I am sure. Xorom make very positive comments about their astribanks and that you can have multiple channel banks on a server so they look pretty good (if they are honest). I can't tell the manufacturer of the other channel banks you were referring to. In Wellington, NZ, PRAs are pretty expensive and a 25Mbit/sec symmetrical fibre connection to a SIP provider is a better deal. On some of my other customers we have 15 SIP lines without issue using G711 and consuming about 80-100k per line if that. But I take the point so will revisit it in the design. Another reason for SIP is the Telepermited options available are limited over here, so to connect you really want to have an approved device in case you have any issues. But with SIP via a Provider you abstract that layer which is cleaner. If we need to have one E1 then having more for the Astribanks sounds fine. Cheers Duncan Rob Hillis wrote: 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 ___ -- 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] top posting again [was: Re: CDR Design]
I like the discussion, I doubt it will end. I prefer top posting because I reply to all my customers that way, my mail client isn't that smart and I think technology should meet the needs rather than force you to adopt work arounds. I can fully understand though others preferring it, but I don't. All the presented evidence so far suggest bottom posting is a work around to a list archive function that is less than ideal or a politeness to get around a way of doing things that doesn't really apply so much anymore. I would have thought someone could make a better list archive model, I don't believe bottom posting is intuitive and therefore being picked up by many newcomers to the game. An alternate is to get a filter that sorts the whole thing out depending on preferences ;-), but who can be bothered. I haven't seen a signup requirement to this list requiring bottom posting, and neither have I on the many other lists I am on. In fact if I look at most of my lists the majority of posters over time have tended to top posting. Doesn't mean its right but it appears to be happening. Cheers Duncan ___ -- 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] Cisco 7906g SIP
Hi Salvatore Have you checked the tftp logs in any event? Its important to check the tftp logs and see if anything is being requested. I have had this before but usually its still trying to grab its first couple of files, and from that you can get an idea of where its getting stuck. If it says upgrading it means its trying to change from one version to another and failing, so you need to go backwards to a version it can cope with. If its not asking for any files then usually what I have done is to go to the lowest SIP version 2 or 3 for changing from the call manager to SIP and reset the phone to factory defaults and try and get it to start the change again Cheers Duncan Sasa wrote: Hi Duncan, yes I have a tftp server (I use also Cisco 7941G that use tftp server for upload configuration) and I know this function, but now my problem is that the phone is stopped on the initial screen that show 'upgrading' and MAC address and the process not continued. Thanks. -- Salvatore. - Original Message - From: Duncan Turnbull [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, October 14, 2008 8:52 PM Subject: Re: [asterisk-users] Cisco 7906g SIP Hi Salvatore Do you have a TFTP server that serves the phone configuration files? This is very separate to the phone, i.e. on a server/pc somewhere, and will log all the file requests it receives. You can check this irrespective of the phone Have you checked whether tftp requests are being made, usually they come before the system goes into the upgrading state. I have had that before and it was caused by having different load files from that specified in the OS79XX.TXT file which for my phones usually have P003-08-6-00 but for upgrading I start from P0S30202 For SIPDefault.cnf you also need the image version to match #Image Version image_version:P0S3-08-6-00 ; But for conversion I first go to this image image_version:P0S30202 ; And I go from that to this image_version:P0S3-06-2-00 ; then to the current version And I have these files on my tftpserver which are the respective firmwares -rwxr-xr-x 1 root root 753560 2007-04-23 14:36 P0S3-08-6-00.sb2 -rwxr-xr-x 1 root root459 2007-04-23 14:36 P0S3-08-6-00.loads -rwxr-xr-x 1 root root 130228 2007-04-23 14:36 P003-08-6-00.sbn -rwxr-xr-x 1 root root 129824 2007-04-23 14:36 P003-08-6-00.bin -rwxr-xr-x 1 root root 486974 2007-04-27 14:51 P0S3-06-2-00.sbn -rwxr-xr-x 1 root root 486570 2007-04-27 14:51 P0S3-06-2-00.bin -rwxr-xr-x 1 root root 392214 2007-04-27 14:51 P0S30202.bin I can't recall if I need all the 08-6 versions Cheers Duncan Sasa wrote: Hi Duncan, I have tried more times to make the reset phone but is displays always and only 'upgrading' and MAC address and I cann't access the phone configuration. Thanks. -- Salvatore. - Original Message - From: Duncan Turnbull [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, October 14, 2008 11:41 AM Subject: Re: [asterisk-users] Cisco 7906g SIP Hi Salvatore You need to look at the logs of the tftp server, not the phone. Hopefully you can see the ip address of the phone asking for files If there is nothing at all being requested from the tftp server then you probably want to reset the phone to defaults again. Usually it stalls when you have some mismatches in the config files. But it almost always asks for the default files. From the files requested you can determine whether its asking for SIP or SCCP files, and if SIP which version of firmware for the phone Cheers Duncan Sasa wrote: Hi Dave, I don't view nothing in tftp server because the phone is stopped on start screen with displayed 'upgrading' and MAC address..I don't understand what happened after the reset. phone Regards. -- Salvatore. - Original Message - From: David Gibbons [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, October 13, 2008 4:29 PM Subject: Re: [asterisk-users] Cisco 7906g SIP Hi Salvatore, I'm talking about the tftp logs on the tftp server: Something like 'tail -f /var/log/tftp' or 'tail -f /var/log/messages' should do the trick. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sasa Sent: Monday, October 13, 2008 9:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7906g SIP I cann't view phone log files because, after reboot, the phone is stopped on this screen ( 'upgrading' with MAC address) ! Regards. -- Salvatore. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list
Re: [asterisk-users] Cisco 7906g SIP
Hi Salvatore Do you have a TFTP server that serves the phone configuration files? This is very separate to the phone, i.e. on a server/pc somewhere, and will log all the file requests it receives. You can check this irrespective of the phone Have you checked whether tftp requests are being made, usually they come before the system goes into the upgrading state. I have had that before and it was caused by having different load files from that specified in the OS79XX.TXT file which for my phones usually have P003-08-6-00 but for upgrading I start from P0S30202 For SIPDefault.cnf you also need the image version to match #Image Version image_version:P0S3-08-6-00 ; But for conversion I first go to this image image_version:P0S30202 ; And I go from that to this image_version:P0S3-06-2-00 ; then to the current version And I have these files on my tftpserver which are the respective firmwares -rwxr-xr-x 1 root root 753560 2007-04-23 14:36 P0S3-08-6-00.sb2 -rwxr-xr-x 1 root root459 2007-04-23 14:36 P0S3-08-6-00.loads -rwxr-xr-x 1 root root 130228 2007-04-23 14:36 P003-08-6-00.sbn -rwxr-xr-x 1 root root 129824 2007-04-23 14:36 P003-08-6-00.bin -rwxr-xr-x 1 root root 486974 2007-04-27 14:51 P0S3-06-2-00.sbn -rwxr-xr-x 1 root root 486570 2007-04-27 14:51 P0S3-06-2-00.bin -rwxr-xr-x 1 root root 392214 2007-04-27 14:51 P0S30202.bin I can't recall if I need all the 08-6 versions Cheers Duncan Sasa wrote: Hi Duncan, I have tried more times to make the reset phone but is displays always and only 'upgrading' and MAC address and I cann't access the phone configuration. Thanks. -- Salvatore. - Original Message - From: Duncan Turnbull [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, October 14, 2008 11:41 AM Subject: Re: [asterisk-users] Cisco 7906g SIP Hi Salvatore You need to look at the logs of the tftp server, not the phone. Hopefully you can see the ip address of the phone asking for files If there is nothing at all being requested from the tftp server then you probably want to reset the phone to defaults again. Usually it stalls when you have some mismatches in the config files. But it almost always asks for the default files. From the files requested you can determine whether its asking for SIP or SCCP files, and if SIP which version of firmware for the phone Cheers Duncan Sasa wrote: Hi Dave, I don't view nothing in tftp server because the phone is stopped on start screen with displayed 'upgrading' and MAC address..I don't understand what happened after the reset. phone Regards. -- Salvatore. - Original Message - From: David Gibbons [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, October 13, 2008 4:29 PM Subject: Re: [asterisk-users] Cisco 7906g SIP Hi Salvatore, I'm talking about the tftp logs on the tftp server: Something like 'tail -f /var/log/tftp' or 'tail -f /var/log/messages' should do the trick. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sasa Sent: Monday, October 13, 2008 9:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7906g SIP I cann't view phone log files because, after reboot, the phone is stopped on this screen ( 'upgrading' with MAC address) ! Regards. -- Salvatore. ___ -- 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] Cisco 7906g SIP
Hi Salvatore You need to look at the logs of the tftp server, not the phone. Hopefully you can see the ip address of the phone asking for files If there is nothing at all being requested from the tftp server then you probably want to reset the phone to defaults again. Usually it stalls when you have some mismatches in the config files. But it almost always asks for the default files. From the files requested you can determine whether its asking for SIP or SCCP files, and if SIP which version of firmware for the phone Cheers Duncan Sasa wrote: Hi Dave, I don't view nothing in tftp server because the phone is stopped on start screen with displayed 'upgrading' and MAC address..I don't understand what happened after the reset. phone Regards. -- Salvatore. - Original Message - From: David Gibbons [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, October 13, 2008 4:29 PM Subject: Re: [asterisk-users] Cisco 7906g SIP Hi Salvatore, I'm talking about the tftp logs on the tftp server: Something like 'tail -f /var/log/tftp' or 'tail -f /var/log/messages' should do the trick. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sasa Sent: Monday, October 13, 2008 9:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7906g SIP I cann't view phone log files because, after reboot, the phone is stopped on this screen ( 'upgrading' with MAC address) ! Regards. -- Salvatore. ___ -- 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] Cisco 7906g SIP
Are you sure you have set the 7960 to SIP? By default they use SCCP, so you need to go through the process of changing them over, which ideally would just be done with the edits you have already in the load files but generally means going back to an early version of the SIP code then working upwards from there. You can check the current hardware in the status, if its SIP it will be something like POS-0806... (I haven't got a phone handy to check) but there is a reasonable amount of info on voipinfo about the process Cheers Duncan Sasa wrote: Hi, I have a problem with Cisco 7906G and SIP protocol use with Asterisk 1.2.26. I have uploaded in my tftp server the firmware 'cmterm-7911_7906-sip.8-0-4SR1' that use 'SIP11.8-0-4SR1S.loads' and in SEPmacaddress.cnf.xml I have: loadInformationSIP11.8-0-4SR1S/loadInformation ..but in tftp log server I have: Oct 07 11:56:22 asterisk1.local atftpd[6230.-1208161360]: Serving CTLSEPmacaddress.tlv to 192.168.0.155:49152 Oct 07 11:56:22 asterisk1.local atftpd[6230.-1208161360]: Serving SEPmacaddress.cnf.xml to 192.168.0.155:49153 ..and in asterisk CLI I have: -- Starting Skinny session from 192.168.0.155 Device SEPmacaddress is attempting to register Now when 7906G started is loaded: load file: sccp11.8-3-2s boot load id: tnp06.3-0-1-31.bin ..why isn't loaded sip firmware ?? Thanks in advance. -- Salvatore. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Verbosity best practice
Its a good question I have lots of disk space so leave it high, I would rather have the detail if I need it It probably would seem sensible to revisit stable systems after a year and lower the verbosity, but then since I can afford the space I am not too fussed. Cheers Duncan Olivier wrote: Hello, When managing a stable system, which verbosity level do you adopt ? Leaving a higher level helps to catch root cause, if for any reason, things go wrong. Leaving a lower level saves resources if you need (have) to backup logs. What are current best practices ? Do you change verbosity level during system lifecycle ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wi-SIP 802.11f - Inter Access Point Protocol HANDOFF
Its not so hard if the APs are purely just converting ethernet to wireless. If there is any authing on the AP then it would be tougher. And a centralised DHCP issuer is important i.e. just one address range across all APs so when moving APs there is no dhcp change, no auth change, just a client reconnecting to the SSID. I guess this is not exactly what you are talking about but organising such protocols to work properly does become much more complex, and no we didn't have so much joy that way. We have a wireless ISP in Wellington, New Zealand called CafeNET and thats all the APs do i.e. wireless to ethernet for large zones of the city homed back to a central controller. I have walked along our main central city street, Lambton Quay and carried on a conversation moving between at least 3 APs. I usually walked about 200 - 350m depending my destination so could be using quite a few APs The asterisk box in question is not blocked by the ISP. Driving is different, but walking is okay, and the street noise masks the other occasional glitches, so I may think its doing better than it is, it can be noisy and hard to hear in any event with a mobile as well. I don't do it so much any more because the cellphone charges got lower and I got tired of two devices, especially one that ran out of batteries. But I did for a few months to prove the point you are asking about. Cheers Duncan Michael Graves wrote: On Sat, 30 Aug 2008 11:51:49 -0500, Karl Fife wrote: Has anyone ever really, truly, actually held on to a Wi-SIP call while moving from the range of one AP to the range of another AP in the same network? Let's say a 'YES' only counts if you had a bona-fide handoff. In other words, you began in place 'A' (within range of AP#1 but OUTSIDE the range of AP#2), AND THEN MOVED to place 'B' (in range of AP#2, but completely outside the range of AP#1) WITYOUT dropping the call. Supposedly it's possible with compliant hardware using 802.11f - Inter-Access Point Protocol (IAPP), but given how ALL standards ALWAYS work together PERFECTLY, 100% of the time :-), I'm guessing that it doesn't work. Can anyone speak to this from experience? -Karl Karl, I'm guessing that it was not common. 802.11f handoffs reportedly take 100ms which is considered too long for streaming applications like voice and video. The 802.11r standard was only agreed upon and released days ago. This specifies FAST BSS transition specifically to saisfy such applications. Not sure if any hardware supports this as yet. http://en.wikipedia.org/wiki/IEEE_802.11r Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AA50 using multiple outbound routes
Hi All I have an AA50 without inbound DDIs but each line has a separate number so based on analogue port it can be routed to different people. The challenge with this method is it appears to only allow the dial plan to use 1 outbound route so if all the analogue ports are split into individual lines it can only use 1 line for outbound, it won't allow it to step to other ports. This seems a less than ideal design and it could be I am missing something. If anyone knows how to make the outbound calls on an Digium AA50 appliance step through all its available ports (or at least a selected subset of them) I would love to know. Thanks very much Cheers Duncan ___ -- 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] TDMoE with Telco
You can use TDMoE to get an E1 running but its really designed to replicate an E1 end to end Its a standard and there is equipment out there that does it, e.g. from RAD and a few others. I didn't have any joy using the Asterisk code to get it going but it should in theory work. Its completely different to Dundi The challenge it is a protocol and needs two boxes talking TDMoE at each end. Telco's do not have this as an option, or at least none do that I have found Cheers Duncan Michael Graves wrote: --Original Message Text--- *From:* Yacine Boukaba *Date:* Sun, 3 Aug 2008 18:54:08 +0100 Hello, is it possible with TDMoE to replace classic digital T1/E1 interfaces like digium and sangoma cards connected to a telco. Or TDMoE is only possible for connecting two asterisk boxes using their NIC interfaces. if TDMoE can work with an T1/E1 connected with telco how we can get the remote mac address of the telco interface ? ThanksNo virus found in this incoming message. Checked by AVG - http://www.avg.com Version: 8.0.138 / Virus Database: 270.5.10/1586 - Release Date: 8/1/2008 6:59 PM I thought that TDMoE was largely depricated in the wake of DUNDi? Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] gui issue in asterisk aa50
I had an issue where I put a comma in the prepend digits string pn call plans and then the call plan menu would no longer load. It parses the menu from the text file so I used the file editor to clear the offending line and my menu came back. Not sure if thats your issue but I was surprised I could enter text that broke the menus Cheers Duncan On 16/07/2008, at 10:27 AM, Sydney Web Hosting [EMAIL PROTECTED] wrote: HI all, I am having issues with the gui on my AA50. under Voice Menus Add new Step Go to Time based rule. It allows me to select “Go to Time based rule” from the menu but no options come up when selected. I’ve tried all browsers but no luck. Thanks David. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] astrundir not used
Are you using ubuntu? Usually I have to edit the Makefile in the else section of Global variable declaration based on architecture # ASTVARRUNDIR=$(localstatedir)/run ASTVARRUNDIR=$(localstatedir)/run/asterisk This seems to do it Cheers Duncan on 07/09/08 04:53 Cyril SCETBON said the following: hi, I'im using asterisk 4.1.21 and astrundir is configured as followed in /etc/asterisk/asterisk.conf : [global] astetcdir = /etc/asterisk astmoddir = /usr/lib/asterisk/modules astvarlibdir = /var/lib/asterisk astagidir = /usr/share/asterisk/agi-bin astspooldir = /var/spool/asterisk astrundir = /var/run/asterisk astlogdir = /var/log/asterisk when I start asterisk it creates his pid file and the ctl socket in /var/run and not in /var/run/asterisk How can I fix it ? Is it a known issue ? I did not get this error with asterisk 1.4.10 Thanks ___ -- 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] sip extension compromised, need help blocking brute force attempts
Try some of the shell scripts in the asteriskcookbook recipe heap http://asteriskcookbook.com/wiki/index.php/RecipeHeap Specifically http://asteriskcookbook.com/wiki/index.php/Asterisk_Brute_Force_Prevention Cheers Duncan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton Sent: Tuesday, 1 July 2008 07:33 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] sip extension compromised,need help blocking brute force attempts iptables -A INPUT -p tcp -s 74.52.112.162 -j DROP Good luck. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of spectro Sent: June 30, 2008 12:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] sip extension compromised, need help blocking brute force attempts Hello, yesterday one of the extensions on my asterisk server got compromised by brute-force attack. The attacker used it to try pull an identity theft scam playing a recording from a bank your account has been blocked due to unusual activity, please call this number... Attacker managed to make lots of calls for around 8 hours before I detected it and changed the password for that extension. As of this morning it is still attempting to brute force the password for that extension again. I need a way to block that IP from connecting to my asterisk server, please advice. --- sip debug --- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 74.52.112.162 : 5060 (NAT) Found user '211' Reliably Transmitting (NAT) to 74.52.112.162:5060: SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 74.52.112.162:5060;branch=z9hG4bK3b28fa36;received=74.52.112.162;rport=5060 From: ASLPLS sip:[EMAIL PROTECTED];tag=as130a4d39 To: sip:[EMAIL PROTECTED];tag=as0c69057b Call-ID: [EMAIL PROTECTED] CSeq: 103 INVITE User-Agent: Asterisk PBX llow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- sip debug --- That box is currently running Trixbox 1.2.3. I have iptables disabled. If anybody can give me a simple ruleset that allows all traffic except ip 74.52.112.162 to port 5060 I will really appreciate it. Are there mechanisms in Asterisk to detect and automatically block these brute force attempts? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] one way audio after call transfer
I had a similar issue in 1.2 after transfer and we were using SIP only but an upgrade cured it We are now on 1.4.18 still without issues Cheers Duncan Rilawich Ango wrote: Hi all, Recently, I experienced one way audio after call transfer. incalling call (PSTN) A -- GXP2000 thro' zap --blind transfer-- destination B Finally A and B reach each others, but there is only one way audio. Anyone get the same experience before? How to solve the problem? Asterisk vesion: Asterisk 1.4.15 zaptel 1.4.7 asteriks-addon 1.4.5 ___ -- 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] Cisco to Asterisk migration
Hi Femi We have about 50 Cisco 7960s on one site off Asterisk 1.4.18 Its all SIP and it doesn't stress a P3 system much at all. I am not sure what phones you are using - the 7960s are not hard to configure, a bit of process to convert from the Cisco Skinny to SIP (using SIP v8.6) but everything seems to work well. The 7961s or 7971s use an XML config which is probably Everything loads off the TFTP server. We are using the Linksys POE Switches SFE2000P which seem okay but don't always like to be fully loaded Things I would work on are automating or simplifying the provisioning (doesn't change that much once its done), firmware upgrades, and getting to know the config files well. Cheers Duncan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Femi Sent: Friday, 25 April 2008 21:34 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Cisco to Asterisk migration Hi Guys, I have client with a Cisco 2690 call manager solution that wants to upgrade but cannot stomach the costs of continuing with Cisco The installation will go up to 100 users The client currently has about 40 Cisco phones and would like to continue with these phones with the odd Polycom I'm looking at plugging in an Asterisk box and using the existing Cisco box as a PSTN gateway only Has anyone on the list done this? Any pitfalls or tips you would like to share? Thanks Femi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Best practice security for internet access to Asterisk
Hi All For the scenario of a single asterisk server that needs to serve clients on the net, as well as local office clients, I would be very interested in people's views of the best method to handle security to prevent net based attacks while still allowing the client access. Some of the challenges I see are: - preventing brute force and bot type attacks - monitoring for unusual events and notifying and acting appropriately - limiting damage if someone does get in - avoiding a Denial or degradation of service on your asterisk platform - making it easy for staff to use Some of this can be done with - firewall control - but its hard to limit where your clients will come from, besides restricting ports - scripts monitoring logs, I saw a recipe for checking password failures then blocking that ip after x failures, I imagine this could get quite sophisticated - using separate restrictions for offnet users but this kind of makes it harder for the staff members. - using a proxy in front of asterisk for SIP, to limit the available extensions and minimise the scanning impact on the asterisk box. I am hoping this could detect and prevent illegitimate or poorly formed requests or unknown user agents. Staff should be using a standard set. - using iax softclients to shift the attack requirements - I don't know much about how well these work - running all clients over a vpn e.g open vpn, but this is not so good for wireless handsets or other devices that can't do a vpn I am interested in all views and recommendations Thanks very much Cheers Duncan ___ -- 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] Upgrade to Asterisk 1.4 - it's one year's old!
We build and maintain 7 Asterisk boxes for our customers, I have recently moved 3 to 1.4. I also use iaxmodem and on the last one 1.4.14 I was getting iax thread errors - which was reported as a bug in much earlier versions but apparently fixed. When 1.4.15 came out (two days later) it solved this problem, for me at least. I didn't dig any further but it did moderate my confidence somewhat. We run everything on ubuntu server 6.06 LTS and also use freepbx as the interface with some minor customisations. It works very well and we are now shifting some others to 1.4 but the issue is if anything goes wrong its too costly to fix, as part of maintenance we keep them uptodate. The main blocker for 1.4 was freepbx but now it supports 1.4 and seems to manage the transition really well. However being a small self employed group of two the main reason to stick with what works is the risk of cost. We don't generally do major upgrades without charging but there isn't any seriously missing functionality yet, and the effort involved to be sure it will be hassle free is significant. The clients have to see value in the upgrade. We also work with people still on version 1.0, because the risk of change to a working system is too high This seems to be the same issue already mentioned but my take on it is most people can't cope with any risk on production machines unless there is some significant gain. Its been a year now, generally I would think that means its probably starting to become stable but a year isn't very long really. Give it another year and the new installs will mostly be 1.4 and the migration process will be a lot more trusted. I don't think a year is really long enough to expect much more than where you are at. The debian stable, unstable, and testing model would be useful here, debian stable is so reliable it just rocks, if there was a version like that it would be fantastic (of course you trade access to the latest features for it) . We find ubuntu server a great balance between debian stability and getting the latest options. Is there a performance analysis of 1.2 vs 1.4 around or a clear business analysis of the distinctions in value for each? Cheers Duncan Lyle Giese wrote: Olle E Johansson wrote: All I can say is with 1.6, if a change is made that causes something that worked in 1.4 not to work in 1.6, please think twice, three times or four times before making the change, or making the change in such a way that it won't break dialplan stuff from 1.4. Our policy is to never remove any functionality between two versions. We replace the functionality with new functionality and print out warnings whenever you use the deprecated functions. We also add this to the documenation in the software and the UPGRADE.TXT file. So the functionality that you lost in 1.4 was old 1.0 functions that was marked as deprecated in 1.2 and removed in 1.4. We might want to be more informative about those changes. We need to make a clear list of things you need to start changing as a user of 1.4 to prepare for lost functionality in 1.6. This information already exist, but should maybe be a bit more public. In some cases we do have to change in a dramatic way and can't preserve the old functionality to solve a bug in the software. This requires thorough discussion in the developer group and is something we really want to avoid at all costs. If this happens, it's clearly documented in the software. Thank you for your feedback, it's important to us. /O Along that this same line, I ran 1.0.something for a long time and it was working just fine for my SOHO. I had a channel bank to interface pots lines from the local Telco and feed the analog phones in the house. Over time, I replaced most of those analog phones with SIP phones. An unfortunate incident caused us to lose that server and several sip phones. When I recovered enough to rebuild *, I tried 1.4 and it would not compile completely and zaptel did not load properly. I download 1.2 and it worked with the same configs as 1.0, but the quality was poor. That was due to hardware issues. I purchased a new motherboard and rebuilt using a newer Asterisk 1.4 with the then current libpri and zaptel and the call quality came back. But I had a hard time with syntax changes. Basically I was jumping from 1.0.x to 1.4.x in one leap. My biggest gripe is that everything loaded and seemed to work. A day later we found this did not work and discovered a syntax change. A day later something else did not work, an other syntax change. Why isn't there some pre-processor to check the syntax of the config files? Would have saved me a whole bunch of time I didn't have to spare and still don't. Lyle As it is syntax problems or changes are not noticed or logged until Asterisk tries to execute them. If there is a chunk of code that is only hit once a week??? It almost
Re: [asterisk-users] 7960 Queue Issue
The freepbx system has a primary number option in its ring group dialing which if selected as a ring strategy means it won't ring any further if the primary number is engaged. This is useful in follow me setups. I haven't dug into how its implemented but it works for ring groups and follow me on freepbx (asterisk 1.2 and 1.4) An article on the concepts. http://freepbx.org/2007/06/03/ring-group-and-follow-me-ring-strategies-1-of-2 It maybe useful to help figure out a way around your issue. Cheers Duncan on 11/05/07 14:09 Nick Brown said the following: Thanks Eric, this is the case. A bit of a shame that it removes the functionality for the member to see calls that have not come from a queue however there is not much choice in the matter. FWIW to get this option a firmware upgrade was required (Now running POS3-08-8-00). Cheers. On 5/11/07 11:57 AM, Eric Merkel [EMAIL PROTECTED] wrote: On 11/4/07, Nick Brown [EMAIL PROTECTED] wrote: Morning All, Quick question that has me stumped. Have a queue with several members (Statically defined in queues.conf at this stage for testing) who use Cisco 7960's. The queue is configured to use rrmemory and generally this works correctly. However if a member is already on a call their phone will still ring (The 7960 can show multiple incoming calls for one line). I really don't want members who are on calls to get more calls. Especially when we start logging out members who don't answer. Asterisk shows; -- Called 1014 -- SIP/1014-08f2e4d0 is ringing -- Local/[EMAIL PROTECTED];1 is ringing -- Nobody picked up in 15000 ms Short of disabling the feature to show multiple incoming calls on the 7960's (Which I don't know if it can be done anyway), has anyone got any suggestions? Yes, you can turn off this in the phone. Go into call preferences on the phone and turn off call waiting. Not optimal but can be done. -Eric ___ --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] Dell PowerEdge 860, Sangoma A108
Hi Helen Sounds good, I think Troy will need me to setup the notification list to the winners though so it might pay to send me those details directly Should be better rugby this weekend for one of us ;-) Cheers Duncan on 10/09/07 14:20 Paul Hales said the following: We have used a quite a few dell 860's in our installs with Digium cards (Te120's) without any issues. PaulH On Mon, 2007-10-08 at 15:28 -0700, Girts Graudins wrote: Hello everyone, I'm considering getting me a quad-core Dell PowerEdge 860 to run Asterisk. Anyone else using this model? Any tales of woe and sorrow I should know about? Then, in a couple of weeks, I'm thinking of getting a Sangoma A108 and giving that a try. Same question with that one - any quirks I should be aware of? Girts ___ --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] FreePBX (2.3) - Good? Bad? Ugly?
I am yet to use 2.3 but have 2.2 on 8 ubuntu based installations with Asterisk 1.2.18 or greater FreePbx is really useful as an interface to all the config files, stats etc, its also really great if your customers need some control The documentation has recently been updated and there is a lot of life in the project so I would recommend it Just note, that like everything you still need to put some time into understanding what you are doing and how to get around the systems Cheers Duncan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jay R. Ashworth Sent: Friday, 14 September 2007 8:57 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] FreePBX (2.3) - Good? Bad? Ugly? On Thu, Sep 13, 2007 at 04:32:27PM -0400, Jay R. Ashworth wrote: I'm about to (finally) do my first Asterisk install; SMB, 4 FXO, 4-6 stations, mostly IP (I'm looking at the Grandstream 201, to start), and maybe X-lite on a couple of laptops via VPN. We've got a 4xFXO box we bought off eBay, which unfortunately I can't find to quote a model number off of, but I *think* it's a Grandstream as well. I've looked at several of the packages that turn Asterisk from a PBX construction kit into an *actual* PBX, and so far FreePBX looks like the one that matches my mental model of a small phone system best. Anyone have any first hand experiences with it that they'd like to share? And I inadvertantly thread-jacked someone. Sorry. Fixed. Cheers -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Slow list
This message arrived today 18 July NZ time Full headers below but most of my mail is like this - the offending bit seems to be: INXS.digium.internal which took 4 days to deliver it Cheers Duncan Return-path: [EMAIL PROTECTED] Envelope-to: [EMAIL PROTECTED] Delivery-date: Wed, 18 Jul 2007 09:33:50 +1200 Received: from pluto.e-simple.co.nz ([203.163.98.172]) by quotient.onesquared.net with esmtp (Exim 4.50) id 1IAug2-0006xz-17 for [EMAIL PROTECTED]; Wed, 18 Jul 2007 09:33:50 +1200 Received: by pluto.e-simple.co.nz (Postfix, from userid 1014) id 7C20823927F; Wed, 18 Jul 2007 09:33:48 +1200 (NZST) X-Spam-Checker-Version: SpamAssassin 3.1.7-deb (2006-10-05) on pluto X-Spam-Level: X-Spam-Status: No, score=-2.2 required=5.0 tests=AWL,BAYES_00, FORGED_RCVD_HELO autolearn=ham version=3.1.7-deb Received: from lists.digium.com (lists.digium.com [216.207.245.17]) by pluto.e-simple.co.nz (Postfix) with ESMTP id 2CF0223928A for [EMAIL PROTECTED]; Wed, 18 Jul 2007 09:33:44 +1200 (NZST) Received: from localhost ([127.0.0.1] helo=INXS.digium.internal) by lists.digium.com with esmtp (Exim 4.63) (envelope-from [EMAIL PROTECTED]) id 1I9oPd-0002fh-Rm; Sat, 14 Jul 2007 15:40:21 -0500 Received: from exprod8mx52.postini.com ([64.18.3.152] helo=psmtp.com) by lists.digium.com with smtp (Exim 4.63) (envelope-from [EMAIL PROTECTED]) id 1I9oPU-0002fV-Ja for asterisk-users@lists.digium.com; Sat, 14 Jul 2007 15:40:12 -0500 Received: from source ([82.103.132.43]) by exprod8mx52.postini.com ([64.18.7.10]) with SMTP; Sat, 14 Jul 2007 13:40:11 PDT Received: from localhost (threed001.three-dimensional.net [127.0.0.1]) by three-dimensional.net (Postfix) with ESMTP id 628F651C50 for asterisk-users@lists.digium.com; Sat, 14 Jul 2007 22:40:10 +0200 (CEST) X-Virus-Scanned: Debian amavisd-new at three-dimensional.net Received: from three-dimensional.net ([127.0.0.1]) by localhost (mail.three-dimensional.net [127.0.0.1]) (amavisd-new, port 10024) with ESMTP id I8DqCNjuf0hk for asterisk-users@lists.digium.com; Sat, 14 Jul 2007 22:40:04 +0200 (CEST) Received: from lunteren.vanbaak.info (vanbaak.xs4all.nl [82.95.250.75]) by three-dimensional.net (Postfix) with ESMTP id EF00D51C4F for asterisk-users@lists.digium.com; Sat, 14 Jul 2007 22:40:03 +0200 (CEST) Received: by lunteren.vanbaak.info (Postfix, from userid 1000) id 5814A50BB; Sat, 14 Jul 2007 22:39:57 +0200 (CEST) Date: Sat, 14 Jul 2007 22:39:56 +0200 From: Michiel van Baak [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Mail-Followup-To: asterisk-users@lists.digium.com References: [EMAIL PROTECTED] MIME-Version: 1.0 Content-Disposition: inline In-Reply-To: [EMAIL PROTECTED] X-PGP-Key: http://michiel.vanbaak.info/Files/pubkey.asc User-Agent: Mutt/1.5.12-2006-07-14 X-pstn-neptune: 0/0/0.00/0 X-pstn-levels: (S:91.99845/99.9 FC:95.5390 LC:95.5390 R:95.9108 P:95.9108 M:97.0282 C:98.6951 ) X-pstn-settings: 3 (1.:1.) s fc lc gt3 gt2 gt1 r p m c X-pstn-addresses: from [EMAIL PROTECTED] [db-null] Subject: Re: [asterisk-users] Slow list X-BeenThere: asterisk-users@lists.digium.com X-Mailman-Version: 2.1.9 Precedence: list Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com List-Id: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users.lists.digium.com List-Unsubscribe: http://lists.digium.com/mailman/listinfo/asterisk-users, mailto:[EMAIL PROTECTED] List-Archive: http://lists.digium.com/pipermail/asterisk-users List-Post: mailto:asterisk-users@lists.digium.com List-Help: mailto:[EMAIL PROTECTED] List-Subscribe: http://lists.digium.com/mailman/listinfo/asterisk-users, mailto:[EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Content-Transfer-Encoding: 7bit Sender: [EMAIL PROTECTED] Errors-To: [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michiel van Baak Sent: Sunday, 15 July 2007 8:40 a.m. To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Slow list On 16:28, Thu 05 Jul 07, Philipp Kempgen wrote: Since the list was switched over to API-Digital almost every message I get is older than a week. Coincidence? Is anyone else having trouble? Regards, Philipp I got this message today July 14 Yes, I have the same. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called 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] Slow list
I thought initially it was a pretty poor generalization about postgrey and our capabilities until I realized that this was sent a few weeks ago when this probably wasn't an as obvious issue. But it clearly is an issue now. I have checked my mail servers for failures, implicitly greylisting is working as the mails are coming from digium constantly - just a long time delayed, if postgrey was an issue there would still be retries and there have been none in over a week - as long as my logs go back, any decent mail server should have retried in much less than a week. Anyway - a discussion and investigation of issues is made pretty hard with 4 -10 day gaps in it. Since every other list works on time (+- a few hours) its looking like Digium from my view. I imagine someone will have sorted it before I see my own post, fingers crossed. Cheers Duncan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Walt Reed Sent: Friday, 6 July 2007 6:33 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Slow list On Thu, Jul 05, 2007 at 01:40:50PM -0400, Doug Lytle said: Well, this is now the third active thread on this subject, but I guess you won't see this message for a while. Has anyone dissected the headers of a delayed message yet? We should be able to tell for sure where the holdup is. All of the messages are coming through on time for me, so it won't do much good for me to look. Looks like mail is getting held up between INXS.digium.internal and lists.digium.com INXS.digium.internal received it the first of July, lists.digium.com received it on the 4th. drdos.info (ME) received it from lists.digium.com on that same day (Today). What you can't see without looking at the mail server logs on both ends is delivery attempts. Greylisting for example can totally hose you over depending on the implementation. Greylisting without whitelisting is irresponsible. How many tries did the digium server make before the message finally got through??? That's what we need to know. Only Digium can say. Before poking Digium too much, I would look at exactly what YOUR mail servers are doing that may potentially be the real cause of the delays. ___ --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] SugarCRM Integration
If you look through the Trixbox without Tears by Ben Sharif - google for it, it's a good read for things you can do for asterisk Ch 31 has this below I would search the tribox and sugar forums for more info - really its just using click to dial from sugar, and potentially CID lookup - I am not sure if it is using Cheers Duncan SugarCRM is a contact management software that comes bundled with Trixbox. To set up SugarCRM, first, you need to open the SugarCRM application http://your.trixboxip.address/crm using the default username of Admin and the password of password. For security reason you should change the Admin password. To do this, click on 'My Account' in the upper right-hand corner, then click on the 'Change Password' button underneath 'Users: Administrator (Admin) in the center-left of the screen. Change it to a new password and confirm your new password and click 'Save.' Now it's time to set up your contacts. I will start off setting up a couple of my internal extensions. Click on 'My Account' again and then click the 'Edit' button. Change 'Asterisk Phone Extension' to your Asterisk extension. My extension is 2001. While you are at it, change your time zone and date format as well. Click 'Save' to save that information. Let's add another one. Click on the contacts tab and then select 'Create Contact' from the left hand Shortcuts menu. Add another extension, in my case I chose my daughter's extension 2002: Firstname: Norsurya Last name: Sharif Home: 2002 Click 'Save' to save that information. Add another and another if you want to, using the method above. At this point, you may find that you are unable to make a phone call through SugarCRM. This is due to a little bug in the popup_picker.php (this bug may have been fixed by the time you read this, but at the time of writing, this bug exists). To fix this bug, you need to edit popup_picker.php by doing the following: From your Linux CLI, log in as root. cd /var/www/html/crm/modules/Contacts nano Popup_picker.php Browse down to line 121 and change it from: $number = preg_replace ( /[^\d\*]/, , $number ); To $number = preg_replace ( /[^\d\*]/, , $display_number ); TRIXBOX Without Tears Page 136 of 209 You should now be able to dial from SugarCRM to your other internal extensions and to the outside world. Note: You can add multiple users who will each have their own settings/contacts/etc. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Crazy Boy Sent: Saturday, 2 June 2007 7:25 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SugarCRM Integration Yes. You are right. You can integrate Sugar with Trixbox very easily. You can customize it also. Thanks, Chandra Joseph Bajin [EMAIL PROTECTED] wrote: I'd like to know as well about this. On 6/1/07, Diego Quintana Cruz wrote: Hi folks, I was wondering if there's a guide on how to configure sugarCRM Integration with Asterisk. I was looking in google and all i found was about Trixbox, which has sugarcrm integrated by default. Regards, -- Diego Quintana a.k.a. RouterMaN Ingeniero de las Telecomunicaciones Linux Registered User #382615 - http://counter.li.org/ SIP # 1-747-633-6676 Ext. 1011 FWD # 764839 Ext. 1011 http://routerman.blogsome.com http://gst.telecom.pucp.edu.pe ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Park yourself in front of a world of choices in alternative vehicles. Visit http://us.rd.yahoo.com/evt=48246/*http:/autos.yahoo.com/green_center/;_ylc=X3oDMTE5cDF2bXZzBF9TAzk3MTA3MDc2BHNlYwNtYWlsdGFncwRzbGsD Z3JlZW4tY2VudGVy the Yahoo! Auto Green Center. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Delays on E1 Delivered via SHDSL
I doubt it's the PRI itself SHDSL isn't part of the internet per se, its just an access technology. SHDSL is just synchronous DSL which can be used to deliver E1s over. ISDN PRI's are delivered in a 2Mbit/sec G703/G704 frame and will give you lots of alarms if they are having any issues It could be your toll provider at the end of it is routing calls in ways that cause delays, but less likely to be the PRI Cheers duncan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Sent: Thursday, 31 May 2007 12:18 p.m. To: asterisk-users@lists.digium.com Subject: [asterisk-users] Delays on E1 Delivered via SHDSL I have an Asterisk system with a TE110P installed and connected to an ISDN E1 PRI that is delivered via a 2mb SHDSL connection. I am experiencing delays (the type of delay you would get on an international call) during calls. I am wondering if anyone could advise, would the problem be with any part of the Asterisk system or is the problem with the fact that the ISDN is delivered over the internet? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Bottom line on fax reception
I use Asterisk and Hylafax and iaxmodem to send 600+ faxes per week with no baby sitting, I receive about 20 and it requires no baby sitting Hylafax and iaxmodem with Asterisk works very well - check out iaxmodem and hylafax lists for much bigger examples Cheers Duncan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of shadowym Sent: Tuesday, 29 May 2007 7:34 a.m. To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Bottom line on fax reception Sorry but I must have missed it if someone else responded. If the built in fax reception doesn't work very well what about the 3rd party stuff mentioned on the Asterisk Wiki? -Original Message- From: Steve Totaro [mailto:[EMAIL PROTECTED] Sent: Monday, May 28, 2007 8:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Bottom line on fax reception Someone already answered this question. The answer is no, it does not work by your definition of production ready. Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of shadowym Sent: Monday, May 28, 2007 11:20 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Bottom line on fax reception Anybody?? -Original Message- From: shadowym [mailto:[EMAIL PROTECTED] Sent: Thursday, May 24, 2007 9:35 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Bottom line on fax reception So what is the bottom line? Does it work or not. I've heard stories it works, it doesn't work, it kinda sorta works when it's not raining out side. Everything under the rainbow. What's the bottom line with recent updates on 1.2.x? Is it production ready for fax? By production ready I mean that it just works all the time and doesn't need any babysitting. Do I have to worry about dropped lines, sometimes not detecting incoming fax toneyada yada. I know I don't have to use fax on Asterisk but I really want to for various reasons. Mostly incoming but outgoing is a nice to have. Should I use an addon package and if so which one? Any help would be appreciated. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] WiFi SIP phones
I have a recent dual gsm /wifi from e28 via Skyvoice. (http://myskyvoice.com/) Its built to use voip or gsm and is about the same price as existing wifi phones. My main hassle is it doesn' yet do WPA - WEP's okay and they say WPA is only a firmware load away ;-) , and it has a browser to login if you need to. So far so good and then to some degree I am not sure I would use a wifi only phone again That said wifi voip is still occasionally flaky but I much prefer it to soft clients on the laptop. Cheers Duncan _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Graves Sent: Thursday, 24 May 2007 2:50 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] WiFi SIP phones I travel a lot for work. I frequently find hotels that have wifi, free or otherwise available. But I've yet to find it anywhere near sufficient to support voip applications. At least not good enough to compel me to not use my cell phone. If you have control of the host LAN then you can ensure it meets the needs of a wifi SIP phone, otherwise why bother. Has anyone ever seen anyone making a voip call on a wif handset ata public hotspot? While that would score many geek points I doubt it would work in many places. About 18 mo ago I bought the Hitachi Cable WIP5000 handset. It was seriously flawed so I resold it after a few months and settled on the Aastra desk phone. I do wish the cordless handsets were a little more like a Panasonic cordless phone...more buttons...easier to program, etc. Michael On Wed, 23 May 2007 21:59:03 -0400, Justin Moore wrote: On 5/23/07, Michael Graves [EMAIL PROTECTED] wrote: I must say that I've VERY happy with my Aastra 4801 CT phones. I think that they're DECT. Each can have up to six cordless handsets. Technically its a 9 line phone, but if you use G.729 you can only sustain two calls at once. I can have a call on the portable and easily take another on the base. I am also an extremely happy user of an Aastra 480i CT. Awesome phone. However, I was under the impression that the OP was looking for a WiFi phone that could be carried from place to place, but I may be wrong... -- Justin Moore aka wantmoore --- www.wantmoore.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] HUDlite Server on Debian Etch/Asterisk 1.4
I have the same challenge and issue, the server dies shortly after being fired up, although I am using Asterisk 1.2 Even with strace its very trying to work out whether the messages are errors or importance or just run of the mill All advice and options appreciated Cheers Duncan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Thursday, 17 May 2007 11:37 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] HUDlite Server on Debian Etch/Asterisk 1.4 On Wed, May 16, 2007 at 03:22:35PM +0200, Jack wrote: Hi, has anyone managed to get hudlite server working on a Debian Etch based installation of Asterisk 1.4? So far I managed to eliminate all error messages, but the process is killed directly after starting the hudlite server without showing any error messages. I would be very happy if anyone can give me some hints or point me to a installation guide. What I would do in such a situation, is run everything under strace. However, recall that you're dealing with a proprietary program here. The only ones who have the full information to help you are Fonality. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] DTMF not working using *98, but OK on inbound routes?
I have this happening with a Cisco 7960 - I can't see what the difference is, I have asterisk 1.2.13 and a number of 7960s which happily work, as well as some 7961s which also work. However one 7960 doesn't, although it dials quite happily but that's probably due to dtmf being put into SIP rather than inband. Why one works and the other doesn't I don't yet know. Cheers Duncan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Sent: Thursday, 17 May 2007 2:40 p.m. To: asterisk-users@lists.digium.com Subject: [asterisk-users] DTMF not working using *98,but OK on inbound routes? Has anyone seen anything like this: I dial *98. Asterisk says Password? I punch in the password, and the system doesn't recognize the tones. However, if I dial my own number and ignore the incoming call, it goes to voicemail, and then I can get into voicemail. I have a sneaking suspicion that Asterisk is somehow not recognizing the DTMF tones somewhere along the way. This happens intermittently with Linksys ATAs and Polycom phones. Using a Cisco 3640 VOIP router. Any ideas on what to check? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users