Re: [asterisk-users] Asterisk Gurus - What is your best Asterisk Queue Analyzer and Asterisk Log Analyzer program out there?
On 31 July 2010 15:28, Leif Madsen leif.mad...@asteriskdocs.org wrote: On 7/30/2010 5:49 AM, Lenz Emilitri wrote: QueueMetrics is actually free (as in beer) for very small call centers and individual hackers. Oh really! I didn't know that! Very nice. What is considered a small call centre? Are we talking up to around 5 agents or something? Is there a limit on the number of queues? (I'm sure there is a page on the website that answers most of these questions, heh :)) Leif Madsen. See http://queuemetrics.com Peter -- _ -- 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] Distinctive ring for INTERNAL calls only? How to do it?
On 19 July 2010 00:35, Anthony Messina amess...@messinet.com wrote: On Wednesday, July 14, 2010 01:44:54 pm bruce bruce wrote: Using Elastix (FreePBX + Asterisk 1.4.2x combination) with Aastra phones, how can one receive distinctive ring tones for INTERNAL calls ONLY? Using Aastra 4801 CT phones... [external-context] ; Calls entering from outside the system exten = 1234,1,SIPAddHeader(Alert-Info: info=Bellcore-dr2) ; Double Ring same = n,Dial(SIP/... [internal-context] ; Calls routed from within the system exten = 1234,1,Dial(SIP/... ; No special ring One of the problems with Distinctive Ring tones is that its not consistent, between different phones so if you have a mix of phone types you have a problem. Quite a lot seam to follow the Bellcore stand says the rhythmn of the ring tone, but not the tune, so Bellcore-dr2 might be long long short and bellcore-dr3 might be short short. A type or Morse code I guess... But its hard work to notice the difference in a hurry when you need to answer the phone, hence its not normally enough. In an ideal world you should be able to send the ring tone with the call so sending a URL or embedding it in the sip header, but I've not heard any method to do this. Peter. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Softphone's
I'm looking for a good Linux Softphone that a has Consultation Transfer built in, I know you can do this by dialling what ever is in features.conf but this is not ideal. b has the ability to handle more than 2 lines eg calls at a time. c Works with Asterisk. d Has a feature where someone can dial in to the phone and listen in to everything, ie bit like call monitoring but include the bits between the calls Does anyone have any ideas, or is it going to be quickest to write my own. Peter. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Is a device a member of a queue?
I'm looking for a function I can put in my dial plan that tells me if
a device is a member of a queue, but I can seem to find one. Basically
I want to be able to dial to join a queue and if I'm already on the
queue, leave..
exten = 4,1,GotoIf(${is_queue_member(queuename,SIP/${ext})}?leave:join)
exten = 4,n(leave),RemoveQueueMember(queuename,SIP/${ext})
exten = 4,n,Hangup
exten = 4,n(join),AddQueueMember(queuename,SIP/${ext})
or simular, If such a function exists it would be very handy
The only way I can see of doing this is to use
queue_member_list(queue) and then loop through the returned list using
cut searching for the device.
So.
1. Is there a function I'm missing to do this say..
is_queue_member(queuename,channel)
or
2. Is there some way of creating such a function.
Thanks in advanced
Peter Childs
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Re: [asterisk-users] Is a device a member of a queue?
On 11 July 2010 14:19, Paul Belanger paul.belan...@polybeacon.com wrote: On Sun, Jul 11, 2010 at 5:40 AM, Peter Childs pchi...@bcs.org wrote: 1. Is there a function I'm missing to do this say.. is_queue_member(queuename,channel) *CLI core show function QUEUE_MEMBER No function by that name registered. also its not listed on voip-info, I'm using SARK/Asterisk 1.4.21 Peter -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.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] Is a device a member of a queue?
On 11 July 2010 15:56, Paul Belanger paul.belan...@polybeacon.com wrote: On Sun, Jul 11, 2010 at 9:28 AM, Peter Childs pchi...@bcs.org wrote: No function by that name registered. also its not listed on voip-info, I'm using SARK/Asterisk 1.4.21 The function is in 1.6.2. Best you could do in 1.4 is: *CLI core show function QUEUE_MEMBER_LIST FYI: voip-info is terribly out of date. Always best to look in your CLI. Hmm Yes but http://www.asterisk.org/docs/asterisk/trunk/functions/queue_member says that it just counts the number of members in the list, just like queue_member_count does. queue_member_penality might do what I want, depending on what it actually returns if the given interface is not a member But then I still need 1.6! Peter Childs -- _ -- 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] Agents
On 17 May 2010 08:40, Lenz Emilitri lenz.lo...@gmail.com wrote:
Use Addmember and removemeber instead :)
l.
Hmm I'm getting that kind of.
From What I can work out.
Agents have been deprecated and are going to be removed.
The replacement, is some complex dialplan using Local Channels which
the admin will have to dream up for themselves.
I'm quite happy to use some new method, but I don't really understand
how yet as all the docs I can find point to using agents
Ideally I need to be able to
a Log into a queue, both by dialing and using the management API
AgentCallbackLogin
b Log Out a que, both by dialing and using the management API
System(agent logoff agent/x) or agentlogoff in management api.
c If the SIP channel (Phone) is not working (Unavailable) remove it
from the queue.
autologoffunavail=yes in agents.conf (but it don't seam to work)
d If the phone is not answered within 10 secs log remove it from the que..
autologoff=10 in agent.conf
e Allow hotdesking extensions so that people don't always need to
login to the same extension.
dial(agent/${EXTEN})
f If the queue is empty or nobody is handling the que drop out, and
ring every phone.
joinempty=strict, leavewhenempty=strict
Using Asterisk 1.4 and a Sark 850.
Any help, or at least where to go
Peter.
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[asterisk-users] Agents
I've been trying to get the hang of Agents and Queues and I must say
its a little unclear as to how things work.
So maybe someone has some better idea
From what I can work out an Agent is meant to be nothing more than a
virtual device that can be moved around physical devices... by login
and logging out. Queues can contain any type of interface not a point
that is partially well put in the Sark we have just got nore in the
voi-info website It also seams to suggest that Agents are a
deprecated feature.
AgentLogin.
AgentCallbackLogin is depreciated but what has it been replaced by?
Not sure what AgentLogin is actually useful for.
AgentCallbackLogin in the Management API does not set
${AGENTBYCALLERID_${CALLERID(num)}} I guessing this is a error,
fortunatly I've worked out a way to get round it. (setvar)
The is no way to log an agent in from the Command Line Interface.
AgentLogoff
Easy so long as you know the agent id you need to logoff, which means
using ${AGENTBYCALLID_${CALLERID(num)}}
Queues really have very little to do with Agents as any type of device
can be statically on a queue or dynamically added when needed, but the
info I've found seams to heavily tie the two concepts together.
Peter
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Re: [asterisk-users] [asterisk-biz] Asterisk system for church call center
On 30 March 2010 02:04, Mark Phillips g7...@g7ltt.com wrote: They say confession is good for the soul. Perhaps they are offering a phone in confessional service? Unfortunately the business of the church often flies in the face of the business of the Church. I think you'll find a lot of Church Based Charities, Consoling and Advise Lines needing very high capacity phone systems out there. Asterisk due to it being Free is an ideal solution for this purpose. The Church primary commission is to tell the world. So use of modern technology is the ideal method of getting this job done. Its not about money its about evangelism. Peter. On 03/29/2010 07:48 PM, Alex Balashov wrote: Sounds like the church has strayed from its core competencies and invited the money-changers into the temple. Being the official asterisk-biz harbinger of God's wrath, I suggest an intensely commercial platform, for the meek shall inherit the Earth, not the 700 Club. Fight the power. -- _ -- 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 Management API
On 11 March 2010 21:09, Matt Riddell li...@venturevoip.com wrote: On 9/03/10 9:13 PM, Peter Childs wrote: Also is there some way to get the starting end to auto pickup, (or at least hit for this to happen (I'm using SIP if that helps)) When you make an originate request it works like this: 1. Call is made to the Channel parameter. 2. When the Channel answers it connects the other end to the application/context/extension. So, send the channel to the SIP device and then the other end won't start till the SIP device picks up. Yes I got that, and it seams to work quite well, It does mean that its more difficult to actually have a call going to a dead phone when it gets sent from the wrong channel in error. 2. Send DTMF to the far end, PlayDTMF looks like it should work but it seams to send the Play the DTMF to my end not the far end. I seam to be able to send it to the far end by finding far end channel's name and using that instead, but this does not work if the far end is not a channel, (eg the Answer phone) but I hope that will not really be a problem... Again, looks like you have the order of the channels round the wrong way. If you originated to a SIP device and sent the other end to the application PlayDTMF, then it would be sent to the SIP device (if that's what you want). I figured that out. It means that if you want to control your calls when in you own menus, you can't do it by send DTMF but need to use the underlining application/dial-plan. which makes things more complex than they should be. Peter. -- _ -- 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 Management API
On 8 March 2010 15:34, Olle E. Johansson o...@edvina.net wrote: 8 mar 2010 kl. 11.13 skrev Peter Childs: On 5 March 2010 13:48, Jim Dickenson dicken...@cfmc.com wrote: At an Asterisk CLI use the command manager show commands. Life is rarely that simple, and this does not really answer the question. Oh and Channel can mean different things in different contexts ie Channel in a PlayDTMF command means a Call to play the DTMF on, where as Channel in a Originate command means the Device to place the call on so you can't use the same input for both commands (or can you?) I agree that it's kind of stupid. I cleared up some of that mess in 1.6.x, but not all. And the changes hurted a lot of existing applications, so I'm careful not to mess around too much with AMI again. The most important part is that we don't allow reuse of existing headers for new things in new actions and events. I've been trying to watch over manager in order to disallow misuse, but development is fast and it's easy to miss a commit or a review... Ok, I'm not 100% sure if this is even possible (it should be) 1. Make a Call (Originate works fine but I can't seam to phone the voice mail using originate, or a que for that matter.) 2. Send DTMF to the far end, PlayDTMF looks like it should work but it seams to send the Play the DTMF to my end not the far end. Currently I'm not finding this any job any easier than the CSTA was on the Alcatel was. Peter. -- _ -- 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 Management API
On 9 March 2010 07:58, Peter Childs pchi...@bcs.org wrote: On 8 March 2010 15:34, Olle E. Johansson o...@edvina.net wrote: 8 mar 2010 kl. 11.13 skrev Peter Childs: On 5 March 2010 13:48, Jim Dickenson dicken...@cfmc.com wrote: At an Asterisk CLI use the command manager show commands. Life is rarely that simple, and this does not really answer the question. Oh and Channel can mean different things in different contexts ie Channel in a PlayDTMF command means a Call to play the DTMF on, where as Channel in a Originate command means the Device to place the call on so you can't use the same input for both commands (or can you?) I agree that it's kind of stupid. I cleared up some of that mess in 1.6.x, but not all. And the changes hurted a lot of existing applications, so I'm careful not to mess around too much with AMI again. The most important part is that we don't allow reuse of existing headers for new things in new actions and events. I've been trying to watch over manager in order to disallow misuse, but development is fast and it's easy to miss a commit or a review... Ok, I'm not 100% sure if this is even possible (it should be) 1. Make a Call (Originate works fine but I can't seam to phone the voice mail using originate, or a que for that matter.) Also is there some way to get the starting end to auto pickup, (or at least hit for this to happen (I'm using SIP if that helps)) 2. Send DTMF to the far end, PlayDTMF looks like it should work but it seams to send the Play the DTMF to my end not the far end. I seam to be able to send it to the far end by finding far end channel's name and using that instead, but this does not work if the far end is not a channel, (eg the Answer phone) but I hope that will not really be a problem... Currently I'm not finding this any job any easier than the CSTA was on the Alcatel was. Peter. Peter. -- _ -- 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 Management API
On 5 March 2010 13:48, Jim Dickenson dicken...@cfmc.com wrote: At an Asterisk CLI use the command manager show commands. Life is rarely that simple, and this does not really answer the question. Oh and Channel can mean different things in different contexts ie Channel in a PlayDTMF command means a Call to play the DTMF on, where as Channel in a Originate command means the Device to place the call on so you can't use the same input for both commands (or can you?) Peter -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Management API
Is there a list of input's / out puts from the management API together with there parameters, there meanings and which are required and what they do/mean. Its just all the docs I've found seam to be rather sketchy and gathered by trial and error, not really up to what I would call a protocol standard. Peter. -- _ -- 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] Beginners Guide to setting up a Call Centre
On 16 January 2010 06:04, Sean Brady sbr...@gtfservices.com wrote: Looking at all the docs I can find Asterisks looks like it should be able to do the job and a whole lot more. This is for a small call centre so ideally we want all the features of an average call centre, ACD, Call Recording, Queue's etc etc. Any pointers on how to get started would be most helpful. Peter. --- (sorry this is so long) Peter, I figured that I would chime in, as I run IT and am a managing partner of a small call center based on Asterisk and I think that my experience will be helpful (hate to beat a dead horse)... Asterisk can definitely do what you need, so I am not going to talk about that any further. I wouldn't waste my time with anything else. I would strongly recommend either of the two following methods to get started, with the deciding factors being time and money. There are lots of factors that will sway this argument, such as the complexity of your workflow, CTI needs, etc., but those time and money are the biggies. You also have to carefully weigh your support requirements, uptime, and your desire to manage a phone system. Asterisk doesn't have to take that much work once it's installed and tuned, but it will require some maintenance. You will need to evaluate whether or not you want to take on that maintenance role or whether you want to pay to have it done for you. Method 1: A professional installation by a Digium Certified Asterisk Professional. It will cost you some money, how much depends on your needs and how clearly you articulate them. There are lots of great people out there that can help you get EXACTLY what you want and design a system that will grow with your business. Call Digium for recommendations, or reply to this with your contact info and we can talk off list (I'm not trying to sell anything, but I have some people that I can recommend). This can be a great option for a solid Asterisk system with good support and reliable operation with little maintenance. There's a couple different approaches to this method- managed and developed with support. Managed is where the team that developed the dialplan and asterisk environment for you manages the system for you as well for a recurring support fee. Drawbacks to this method: A. You will have to find a good vendor that will charge fairly and deliver on their SLA (always get an SLA with enforceable penalties). This isn't that tough, but it's important. B. The recurring support costs can eat into your budget quickly C. This will take some time to develop properly, and for simple environments it may be overkill. D. Adds/changes/ and deletes can be costly as well. This can be mitigating by communicating the need to accommodate staff turnover with a user maintainable system. Does not sound much worse than what we have now :) Method 2: Get a distro, install it, be dialing in about 8 hours or less (the route that I took when we started). This method is by far and away the easiest, cheapest, get-it-up-and-running-consequences-be-damned method. You will take less time, effort and money to get going like this than any other way I know of. If your call flow is simple to moderately complex, this is the way to go in my opinion. The FreePBX distros (Trixbox, AsteriskNOW [I think], Elastix, etc) all are very well put together, and will do everything that you listed in your original message and then some. Of the distro's, I would probably either go with AsteriskNOW or, if you are up for a little more setup work, FreePBX on it's own. Drawbacks to this method: A. I can't speak for others, but I found that the configuration engines have their limitations when it comes to call centers. They simply weren't designed to do some of the specific things that we needed to do as we grew. This doesn't mean that they wont do everything you need though, each case is unique. They were fine for us in the beginning, but as our business grew so did our specific needs, and we outgrew these solutions. There is nothing wrong with that if you understand from the outset that you may have needs that aren't met in the future. These distros have to factor in the needs of their respective communities, and what may be good for one organization might not be good for others. B. Troubleshooting issues can be more complex as you start to understand Asterisk and increase your level of sophistication. I had a hard time troubleshooting FreePBX until I understood it's dialplan more, and it made troubleshooting complicated as I didn't fully understand the call flow through it's dialplan. The more you work with it, the easier it gets, but there can be a learning curve. C. Integration with other vendor's products can sometimes be a challenge if they don't already support your
[asterisk-users] Sip Trunk takes incomming calls for 2 minutes and then stops
Using sipgate.co.uk, Asterisk, FreePBX and Asterisk in a Flash I've managed to get a basic system set up. and can now take and make sip calls over the sip trunk I've got from sipgate.co.uk for testing purposes Anyway I can make calls fine (if only to the testing line and other sipgate lines as I have not set up any credit), and I can take calls but only if someone phones me within 2 minutes of doing a sip reload otherwise I just get a dead line. I'm thinking this is something to do with registration or Nat, but I've set my Nat up to forward everything, and it all works for 2minutes. Peter. -- _ -- 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] Sip Trunk takes incomming calls for 2 minutes and then stops
2010/1/26 Yves Arikoglu yves...@gmx.de: do you use the qualify=yes No, If I do it does not work at all. I've found if I set defaultexpiry to 30 it works fine. and was infact working for 30 seconds every two minutes before, It looks like sipgate.co.uk are expiring there registry attempts very quickly. Peter option for your endpoints? y. Peter Childs schrieb: Using sipgate.co.uk, Asterisk, FreePBX and Asterisk in a Flash I've managed to get a basic system set up. and can now take and make sip calls over the sip trunk I've got from sipgate.co.uk for testing purposes Anyway I can make calls fine (if only to the testing line and other sipgate lines as I have not set up any credit), and I can take calls but only if someone phones me within 2 minutes of doing a sip reload otherwise I just get a dead line. I'm thinking this is something to do with registration or Nat, but I've set my Nat up to forward everything, and it all works for 2minutes. Peter. -- _ -- 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] Sip Trunk takes incomming calls for 2 minutes and then stops
2010/1/26 Peter Childs pchi...@bcs.org: 2010/1/26 Yves Arikoglu yves...@gmx.de: do you use the qualify=yes No, If I do it does not work at all. I've found if I set defaultexpiry to 30 it works fine. and was infact working for 30 seconds every two minutes before, It looks like sipgate.co.uk are expiring there registry attempts very quickly. However I'm not totally sure this fixes the whole problem, as it still only works sometimes. Its just its works more often now than it did before. Peter. Peter option for your endpoints? y. Peter Childs schrieb: Using sipgate.co.uk, Asterisk, FreePBX and Asterisk in a Flash I've managed to get a basic system set up. and can now take and make sip calls over the sip trunk I've got from sipgate.co.uk for testing purposes Anyway I can make calls fine (if only to the testing line and other sipgate lines as I have not set up any credit), and I can take calls but only if someone phones me within 2 minutes of doing a sip reload otherwise I just get a dead line. I'm thinking this is something to do with registration or Nat, but I've set my Nat up to forward everything, and it all works for 2minutes. Peter. -- _ -- 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] Beginners Guide to setting up a Call Centre
Ok this has Probably been put to bed several time but never mind. Elastix, Trixbox, or AsterixNow, or DIY (ie Ubuntu or whatever installed with OpenPBX, Asterix etc by hand) I've got a new server to run Asterix on and want to get it working quickly and yet be configurable in the future with out having to reisntall and start again regally. Currently no VoIP hardware but that will come once I prove the concept. I guess Oh the machine does not have a CD Rom Drive so a network/USB install would be nice.. But I guess I can open the case and plug one in for installation if I must! (Says he who has just installed Ubuntu over the network to check the computer works!) Peter. -- _ -- 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] Beginners Guide to setting up a Call Centre
This is currently still at a proof of concept stage. After being mis-sold a Alcatel phone system, that does None of the things we asked for (Ok it takes calls but that's about it) We are looking at alternatives to try and bring some of the features we previously had on our old Analogue STS phone system. Looking at all the docs I can find Asterisks looks like it should be able to do the job and a whole lot more. This is for a small call centre so ideally we want all the features of an average call centre, ACD, Call Recording, Queue's etc etc. Any pointers on how to get started would be most helpful. Peter. -- _ -- 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] Beginners Guide to setting up a Call Centre
2010/1/12 James Mutuku listmut...@gmail.com: http://www.google.co.ke/search?q=asterisk+for+call+centersie=utf-8oe=utf-8aq=trls=org.mozilla:en-US:officialclient=firefox-a I can use Google just as well as the next guy, and if you'd bothered to look at the results you could see they were extremely bland and not partially useful. I'm thinking I want some up to date information and a beginners guide, But I'm finding it difficult to find much dated after 2003 I'm not an expert on phones, I'm just an IT guy who thinks he might have a solution to a problem, that is not really his problem but is trying to see if he can get it to work. That's how bad the Alcatel phone system is! Peter. -- _ -- 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] Beginners Guide to setting up a Call Centre
2010/1/12 Robert Lister r...@lentil.org: Do you have any idea of numbers of users, and number/type of external lines as this can be quite important when deciding what type of asterisk setup and hardware to go with. (For example, if your lines are already presented over ISDN PRI or BRI, or if they are provided over IP, by an IP telephony provider.) Up until a year ago when we got our new useless Alcatel system all our lines were analogue. When we got the Alcatel we got told Everything has gone digital so we now have a ISDN PRI. I'm now seeing this is actually far from the truth but never mind Also you will need to think if you want to support analogue devices such as modems/fax machines etc. If the system can take faxes then fine, We already use Hylafax on a separate analogue line currently, which is were it will remain, unless I can find a good reason to change. If Asteriks can identify a fax coming in on the main line and do the right thing then that would be a neat feature, but its in the end of the world if its not there. Do you have existing IP handsets that you want to integrate, and what are these? Or are you starting from scratch? Or are you going to use PCs with soft phones and headsets? (Often very suitable for a call centre setup) Starting from scratch, I'm not sure I trust soft phones enough, But it would be cheap and the project has very little budget currently! What sort of support do you require for the system / handsets? Do you need CTI integration / soft phones / headsets etc? Yes this is vital its the one of the big things we miss since we got the new Alcatel (The Alcatel crashes every 2 days if we switch the CTI on!) Headset vital. How many lines in total are coming in to the system? Currently 1 ISDN PRI I think but I can't see anything Asterks should not be able to handle. we used to have 48 Analogue lines but I've not seen the office having more than 5 calls at the same time in years. Do you need hotdesk users or are they all based at the same desks every day? Totally HotDesk 24x7 phones are always in use. We don't currently have personal extensions but this would be a nice feature What are the requirements for redundancy/failover? (ranging from 'none' to 'magic failover between two sites') Fallover would be nice again we don't have any currently. we would also like people to be able to log in and take calls from home from time to time when we get really busy I'm looking at AsterksNow/TrixBox but I'm a ubuntu guy (whole office is running on Ubuntu for our desktops) so if the phones run that too it would mean everything was the same, but if the simplest solution is different then fine. I do need a GUI that is easy to deal with, ie adding users, groups, queues etc. If you can answer this, then it will help work out what sort of hardware you will need (software can be changed about to suit, but choice of server setup/cards/media gateways is important in that decision as well.) I've got a basic idea what I need, I'm just trying to work out a demo to get the idea of the board past management (Without causing too much trouble) Software, There are many pre-built solutions that are based on asterisk which have GUIs to use/admin them. These may or may not do what you want out of the box. Hot desk support is particularly limited in many of these. It shows how good our old STS system really was! Or you can install just the base asterisk and roll your own. This is a bit more complex (and maybe unneeded if you are using on the most common features.) but it has its benefits, such as not being restricted by a particular GUI or management system, and being able to customise things a bit more. Peter. Rob On Tue, 2010-01-12 at 10:55 +, Peter Childs wrote: This is currently still at a proof of concept stage. After being mis-sold a Alcatel phone system, that does None of the things we asked for (Ok it takes calls but that's about it) We are looking at alternatives to try and bring some of the features we previously had on our old Analogue STS phone system. Looking at all the docs I can find Asterisks looks like it should be able to do the job and a whole lot more. This is for a small call centre so ideally we want all the features of an average call centre, ACD, Call Recording, Queue's etc etc. Any pointers on how to get started would be most helpful. -- _ -- 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 NEC NEAX 2000 IPS
On Tue, 2006-07-11 at 12:29 +1000, MBIT Technologies wrote: Hi Guys I am just looking for a bit of help here. I am trying to integrate the 2 of these together via a E1 link. The link has no signalling and is basically a dumb 2 meg link. I would have thought that you would have _some_ type of signalling. Perhaps it would be easier to have it configured as either ETSI ISDN (CPE) or QSIG, and then configure your digium as per the samples/wiki/etc... (as either QSIG or ISDN NET) Cheers, Peter ___ --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] Asterisk Genesys integration
What function would you be expecting asterisk to be performing? Genesys have a IP call server that basically functions as a call-control facility like a B2BUA. They also have a RTP process to do digit collection, play announcements etc... etc...etc... If you wanted to say use asterisk as a PBX, and have a 'Asterisk' T-Server then I think you might be out of luck as they no longer have third parties develop t-servers. As a media gateway, pre-treatment, etc it should be usable. Cheers, Peter -- Peter Childs NEC Business Solutions Ltd Ph:61-8-8301-4908 Mb:61-4-0819-7693 IM: pjcinaus (yahoo) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marakovic, Ivan Sent: Tuesday, 17 January 2006 1:36 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk Genesys integration Hi All, Would it be possible to integrate Asterisk with Genesys CTI? Has anyone done any work on Asterisk/Genesys integration? Regards Ivan Marakovic Operations Manager Link:Q Take the hassle out of paying your Link:Q account each month - set up a DIRECT DEBIT facility. It's easy - just call Link:Q customer service on 1300 650 840 ** The information in this email and any attachment is confidential and may be privileged. If you are not the intended recipient, please destroy this message, delete any copies held on your system and notify the sender immediately or telephone Link:Q on +613 9625 8000. You should not retain, copy or use this email for any purpose, nor disclose all or any part of its contents to any other person. Any views expressed in this message are those of the individual sender and, in the absence of express authority, should not be regarded as the views of Link:Q. ** ___ --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] Media gateway recommendations?
The couple of AS5300's that we have seem to work fine. We are using them for E1-SIP with SER and Asterisk. We also use PC/Digium for E1-SIP. With far end echo issues I found that adjusting the echo tail on the cisco 'fixed' things straight away, with the Digium. Well... Um Perhaps I lacked the skills. YMMV. Cheers, Peter -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dustin Wenz Sent: Wednesday, 16 November 2005 2:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Media gateway recommendations? I've been looking a little as the Cisco AS5000 series (specifically the AS5350) as a SIP gateway for our PRI T1. Does anybody know how well these work with Asterisk? - .Dustin Wenz On Nov 14, 2005, at 4:09 PM, Dustin Wenz wrote: Thanks for the info. Are you finding the Lucent gateway to play as nicely as people say it should with Asterisk? The data sheet claims that it can manage 720 concurrent calls. I think that piece of hardware is a little too extreme for our purposes. Even something that offered 1/10th the capacity would be more than enough. Does Lucent offer any sort of TNT Universal Gateway Mini? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] NEC IPS PABX
What interface do you have in the asterisk box? Digium? Make model? What does your /etc/zaptel look like? What is your signalling set to in /etc/asterisk/zapata.conf (ie pri_cpe or pri_net)? What does the output of 'cat /proc/zaptel/* | grep Span' look like? What is your PBX side interface (a 30PRTA or 30DTC+SC01?) -- just interested, it shouldn't matter unless you are trying a QSIG interface... Are you attempting calls from both directions, or just from the PBX? If you are just attempting from the PBX, perhaps attempt a call from the asterisk system. If you have no other devices you can use a softphone, or we usually just throw a 'call' description file into /var/spool/asterisk/outgoing that plays music on hold for a few minutes... Cheers, Peter -- Peter Childs NEC Business Solutions Ltd Ph:61-8-8301-4908 Mb:61-4-0819-7693 IM: pjcinaus (yahoo) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, 9 November 2005 11:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] NEC IPS PABX We are here at a site trying to get an NEC IPS pabx to talk to an Asterisk Box via E1. We get a green light on the E1, but we can't see any call data moving between the systems. I have turned on all of the debugging there is, and we still see nothing. Any ideas? PaulH ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [other] [Asterisk-Users] Dell Poweredge 1400
Called [EMAIL PROTECTED] ?? I assume it actually didn't dial 'myhomeno' but your number? Are you fully prefixing your number (ie 10 digits)... Does ringing other numbers (a engine test number etc..) work? Cheers, Peter -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of clint_in_sydney Sent: Saturday, 13 August 2005 8:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [other] [Asterisk-Users] Dell Poweredge 1400 Hi all, I get an engaged tone and the following when I dial out. -- Called [EMAIL PROTECTED] -- Got SIP Response 404 Not Found back from 202.61.13.40 -- SIP/byo.engin.com.au- is circuit-busy To me this suggests that the engin server can't find my PSTN... Does anyone have any ideas as to what I should look for? Cheers, Clint Send instant messages to your online friends http://au.messenger.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] HELP! X100P IRQ conflict w/ USB
Have you loaded the zaptel drivers yet? Until you load the wcfxo module you will see nothing in /proc/interrupts. Cheers, Peter -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of 163 Sent: Friday, 5 August 2005 12:35 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] HELP! X100P IRQ conflict w/ USB PC: HP Vetra VL400 Mainbood: Intel815 BIOS: Phonix 4.0 release 6.0 OS: REDHAT 9.0 I installed the X100P in PCI slot 2 and disable the USB port, serial-port and parallel-port in BIOS. I can't found the X100P card in cat interrupts But I can found the card in the cat ioports use lspci I found the X100Pcard use the interrupts 11 too. Who can help me to solve this problem? --- [EMAIL PROTECTED] proc]# cat interrupts CPU0 0: 28561 XT-PIC timer 1: 6 XT-PIC keyboard 2: 0 XT-PIC cascade 5: 2954 XT-PIC eth0 8: 1 XT-PIC rtc 11: 0 XT-PIC usb-uhci 12: 29 XT-PIC PS/2 Mouse 14: 7199 XT-PIC ide0 15: 1 XT-PIC ide1 NMI: 0 ERR: 0 - - [EMAIL PROTECTED] proc]# cat ioports -001f : dma1 0020-003f : pic1 0040-005f : timer 0060-006f : keyboard 0070-007f : rtc 0080-008f : dma page reg 00a0-00bf : pic2 00c0-00df : dma2 00f0-00ff : fpu 0170-0177 : ide1 01f0-01f7 : ide0 0376-0376 : ide1 0378-037a : parport0 037b-037f : parport0 03c0-03df : vga+ 03f6-03f6 : ide0 0cf8-0cff : PCI conf1 1800-180f : Intel Corp. 82801AA IDE 1800-1807 : ide0 1808-180f : ide1 1810-181f : Intel Corp. 82801AA SMBus 1820-183f : Intel Corp. 82801AA USB 1820-183f : usb-uhci 2000-20ff : Tiger Jet Network Inc. Model 300 128k 2400-247f : 3Com Corporation 3c905C-TX/TX-M [Tornado] 2400-247f : 01:04.0 -- [EMAIL PROTECTED] sbin]# ./lspci -vvv 00:00.0 Host bridge: Intel Corp. 82815 815 Chipset Host Bridge and Memory Contr) Subsystem: Intel Corp. 82815 815 Chipset Host Bridge and Memory Controlb Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Ste- Status: Cap+ 66Mhz- UDF- FastB2B+ ParErr- DEVSEL=fast TAbort- TAbort-- Latency: 0 Capabilities: [88] #09 [f104] 00:02.0 VGA compatible controller: Intel Corp. 82815 CGC [Chipset Graphics Cont) Subsystem: Hewlett-Packard Company: Unknown device 1245 Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Ste- Status: Cap+ 66Mhz+ UDF- FastB2B+ ParErr- DEVSEL=medium TAbort- TAbor- Latency: 0 Interrupt: pin A routed to IRQ 10 Region 0: Memory at f000 (32-bit, prefetchable) [size=64M] Region 1: Memory at ec00 (32-bit, non-prefetchable) [size=512K] Capabilities: [dc] Power Management version 2 Flags: PMEClk- DSI+ D1- D2- AuxCurrent=0mA PME(D0-,D1-,D2-,D3ho) Status: D0 PME-Enable- DSel=0 DScale=0 PME- 00:1e.0 PCI bridge: Intel Corp. 82801AA PCI Bridge (rev 02) (prog-if 00 [Normal) Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Ste- Status: Cap- 66Mhz- UDF- FastB2B+ ParErr- DEVSEL=fast TAbort- TAbort-- Latency: 0 Bus: primary=00, secondary=01, subordinate=01, sec-latency=32 I/O behind bridge: 2000-2fff Memory behind bridge: ec10-ec1f Prefetchable memory behind bridge: fff0-000f BridgeCtl: Parity- SERR- NoISA+ VGA- MAbort- Reset- FastB2B- 00:1f.0 ISA bridge: Intel Corp. 82801AA ISA Bridge (LPC) (rev 02) Control: I/O+ Mem+ BusMaster+ SpecCycle+ MemWINV- VGASnoop- ParErr- Ste- Status: Cap- 66Mhz- UDF- FastB2B+ ParErr- DEVSEL=medium TAbort- TAbor- Latency: 0 00:1f.1 IDE interface: Intel Corp. 82801AA IDE (rev 02) (prog-if 80 [Master]) Subsystem: Intel Corp. 82801AA IDE Control: I/O+ Mem- BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Ste- Status: Cap- 66Mhz- UDF- FastB2B+ ParErr- DEVSEL=medium TAbort- TAbor- Latency: 0 Region 4: I/O ports at 1800 [size=16] 00:1f.2 USB Controller: Intel Corp. 82801AA USB (rev 02) (prog-if 00 [UHCI]) Subsystem: Intel Corp. 82801AA USB Control: I/O+ Mem- BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Ste- Status: Cap- 66Mhz- UDF- FastB2B+ ParErr- DEVSEL=medium TAbort- TAbor- Latency: 0 Interrupt: pin D routed to IRQ 11 Region 4: I/O ports at 1820 [size=32] 00:1f.3 SMBus: Intel Corp. 82801AA SMBus (rev 02) Subsystem: Intel Corp. 82801AA SMBus Control: I/O+ Mem- BusMaster- SpecCycle- MemWINV- VGASnoop- ParErr- Ste- Status: Cap- 66Mhz- UDF- FastB2B+
RE: [Asterisk-Users] Klicking sounds in background
Your ISDN clocking is slipping (or not sync'd). Your digium PRI needs to clock off the ISDN. See zaptel.conf on the Wiki and set something like... Span=1,1,... (the second '1' is important.. Ie 'use as primary sync source') http://www.voip-info.org/tiki-index.php?page=Zaptel.conf+span+sintax -- Peter Childs NEC Business Solutions Ltd Ph:61-8-8301-4908 Mb:61-4-0819-7693 IM: pjcinaus (yahoo) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jochen Witte Sent: Wednesday, 27 July 2005 10:48 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Klicking sounds in background Hello, I have an IVR with Intel HMP SIP stack, which is a peer behind my Asterisk box (Asterisk 1.0.7, Digium PRI). When dialing in via PSTN, there are klicking sounds in the background, which do not appear, when dialing in via SIP (using Asterisk as pbx). The issue does not seem to be an alaw/µlaw problem. I tried trunking two Asterisk boxes via IAX and then call via two asterisks, but the same effect appears. Whenever there is PSTN involved, I have these klicking sounds, when there is no PSTN, everything works correctly. The setup works great with different SIP peers (others than the Intel...) Anyone has an idea? Best regards Jochen -- Jochen Witte email: [EMAIL PROTECTED] web: http://alpha-lab.net ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TE410P card in an HP-Compaq DL380 G4 server
What is your dmesg output when you fire up the card. There were some problems with TE410P and the intel chipset used in the DL380 G4's. You need firmware at least 'TE410P version c01a010b' Contact Digium and RMA if you have older firmware (basically the symptom will be everything is ok, but the never generates an interrupt under /proc/interrupts Cheers, Peter -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Roland Zagler Sent: Wednesday, 20 July 2005 4:52 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] TE410P card in an HP-Compaq DL380 G4 server Hello list, Did anyone already get the T410P card running in an HP-Compaq DL380 G4 server? If yes, how? I'm using Fedora Core 3 with 2.6.11-1.35_FC3smp Kernel package. Thanks in advance, Roland ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Fwd: RE: [Asterisk-Users] Re: TE410P card in an HP-Compaq DL380 G4 server]
We switched out our Cisco AS5300 with a new TE410P card, works a treat. We couldn't get the D channels up when we hooked up to a PBX ISDN-PRI when the zap was pri_net, but hooking up to a carrier ISDN-PRI as pri_cpe works a treat. Now we just have to figure out how to get the echocancelling tail a little longer without stuffing the line full of static (?!!) [=yes works =256 very noisy...]From testing with our Cisco we needed more than 16ms tail, 32ms tail on the cisco cleaned up incoming that originated from analog PSTN fine... Cheers, Peter -Original Message- From: Eric Bishop [mailto:[EMAIL PROTECTED] Sent: Saturday, 12 March 2005 11:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; Peter Childs Subject: Re: [Fwd: RE: [Asterisk-Users] Re: TE410P card in an HP-Compaq DL380 G4 server] How did you go? On Tue, 8 Mar 2005 11:28:59 +1030, Peter Childs [EMAIL PROTECTED] wrote: Digium shipped me a replacement card, but they sent the wrong one, so they fedex'd another and its just arrived. Should be testing in the next two days (the box is in another state...) The last I heard from Eric Bishop (on the 1st march) was that he had received an updated card from digium, but it didn't function in his DL380... I can let you know the outcome of the test if you'd like. [...] -Original Message- From: Mark F. Vickers [mailto:[EMAIL PROTECTED] Sent: Tuesday, 8 March 2005 11:20 AM To: [EMAIL PROTECTED]; asterisk-users@lists.digium.com Subject: [Fwd: RE: [Asterisk-Users] Re: TE410P card in an HP-Compaq DL380 G4 server] Was there any resolution on this I also have a TE410P in an box with an Intel E7501 chipset? -Vickers ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Fwd: RE: [Asterisk-Users] Re: TE410P card in an HP-Compaq DL380 G4 server]
A co-worker installed the card and when the driver was loaded the lights went red! (instead of just turning off) This is a big step forward, however I won't be testing asterisk with the card until tomorrow.Fingers crossed. :) Cheers, Peter -Original Message- From: Eric Bishop [mailto:[EMAIL PROTECTED] Sent: Saturday, 12 March 2005 11:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; Peter Childs Subject: Re: [Fwd: RE: [Asterisk-Users] Re: TE410P card in an HP-Compaq DL380 G4 server] How did you go? On Tue, 8 Mar 2005 11:28:59 +1030, Peter Childs [EMAIL PROTECTED] wrote: Digium shipped me a replacement card, but they sent the wrong one, so they fedex'd another and its just arrived. Should be testing in the next two days (the box is in another state...) The last I heard from Eric Bishop (on the 1st march) was that he had received an updated card from digium, but it didn't function in his DL380... I can let you know the outcome of the test if you'd like. Cheers, Peter -Original Message- From: Mark F. Vickers [mailto:[EMAIL PROTECTED] Sent: Tuesday, 8 March 2005 11:20 AM To: [EMAIL PROTECTED]; asterisk-users@lists.digium.com Subject: [Fwd: RE: [Asterisk-Users] Re: TE410P card in an HP-Compaq DL380 G4 server] Was there any resolution on this I also have a TE410P in an box with an Intel E7501 chipset? -Vickers Original Message Subject: RE: [Asterisk-Users] Re: TE410P card in an HP-Compaq DL380 G4 server Date: Tue, 8 Feb 2005 11:13:24 +1030 From: Peter Childs [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com CC: [EMAIL PROTECTED] RMA your non-functional card and get one with a new firmware they are trying that fixes the issues with the Intel E75xx chipsets. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Tony Mountifield Sent: Monday, 7 February 2005 6:53 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: TE410P card in an HP-Compaq DL380 G4 server In article [EMAIL PROTECTED], Peter Childs [EMAIL PROTECTED] wrote: Contact Digium Support. They have been very helpful with this issue (mention your using the G4 server with the Intel E7520 Chipset..) So do they have a solution? What is it? Cheers Tony ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Fwd: RE: [Asterisk-Users] Re: TE410P card in an HP-Compaq DL380 G4 server]
Digium shipped me a replacement card, but they sent the wrong one, so they fedex'd another and its just arrived. Should be testing in the next two days (the box is in another state...) The last I heard from Eric Bishop (on the 1st march) was that he had received an updated card from digium, but it didn't function in his DL380... I can let you know the outcome of the test if you'd like. Cheers, Peter -Original Message- From: Mark F. Vickers [mailto:[EMAIL PROTECTED] Sent: Tuesday, 8 March 2005 11:20 AM To: [EMAIL PROTECTED]; asterisk-users@lists.digium.com Subject: [Fwd: RE: [Asterisk-Users] Re: TE410P card in an HP-Compaq DL380 G4 server] Was there any resolution on this I also have a TE410P in an box with an Intel E7501 chipset? -Vickers Original Message Subject: RE: [Asterisk-Users] Re: TE410P card in an HP-Compaq DL380 G4 server Date: Tue, 8 Feb 2005 11:13:24 +1030 From: Peter Childs [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com CC: [EMAIL PROTECTED] RMA your non-functional card and get one with a new firmware they are trying that fixes the issues with the Intel E75xx chipsets. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Tony Mountifield Sent: Monday, 7 February 2005 6:53 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: TE410P card in an HP-Compaq DL380 G4 server In article [EMAIL PROTECTED], Peter Childs [EMAIL PROTECTED] wrote: Contact Digium Support. They have been very helpful with this issue (mention your using the G4 server with the Intel E7520 Chipset..) So do they have a solution? What is it? Cheers Tony ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Zyxel Prestige 2000W
Sort of. I worked sort of ok, but I found I really just thought it sucked, and carrying a wifi phone and mobile together just didn't impress me at all! I had some issues with WEP, but I was trying to run adhoc so it may not have been a problem with the device but my wifi... Good luck. Cheers, Peter -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rod Bacon Sent: Thursday, 3 March 2005 1:28 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Zyxel Prestige 2000W Does anyone have this piece of crap working properly with Asterisk? I thought my problems were all due to NAT, but even on my local LAN segment it's still flaky. Symptoms include periodically losing registration and/or being able to make/receive one call, then not another until rebooted (failing to hang-up). P.S. It's NOT a wireless problem. I'm sitting right on top of the access point, and have a strong, clean signal. == Rod Bacon Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600Fax: +613 99401650 == ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Anyone using SuperMicro SuperServer 6014P-8R?
From http://www.supermicro.com/products/system/1U/6014/SYS-6014P-8R.cfm you can see the board has the IntelR E7520 chipset. I would suggest you note this to Digium when purchasing your TE410p, as several people have had issues with this chipset in servers (see HP DL380-G4), and Digium have a newer firmware which may resolve this issue. Cheers, Peter -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Augustine Olaifa Sent: Tuesday, 22 February 2005 8:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Anyone using SuperMicro SuperServer 6014P-8R? Yes, one of the supermicro motherboards will work fine with both TE10P and TE05P (not sure of the specifics 6014P-8R) just make sure you look out for the slots parameters as follows: for TE10P the board must have a 32-bit PCI -slot with 66Mhz speed bus and importantly uses 3volts. (if the board has a 64-bit PCI-X slot the better because this will allow you switch from 33Mhz -100Mhz speed bus) (Note that the TE10P will not work with 5 volts board power rating) for TE05P according to the specs on the technical sheet the card will work with a 16-bit or 32 -bit PCI slot will work fine on the conventional 5Volts power ratings on boards. On Mon, 21 Feb 2005, Tony Mountifield wrote: Hi, Is anyone here using the SuperMicro SuperServer 6014P-8R with Asterisk? I'm especially interested if you've used it with a TE405P or TE410P. Cheers Tony -- Olaifa Augustine General Data Engineering Services Ltd 18b oshin road,kongi bodija p.o.box 29460, secretariate, ibadan. tel:- 234-2-8105156 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: TE410P card in an HP-Compaq DL380 G4 server
RMA your non-functional card and get one with a new firmware they are trying that fixes the issues with the Intel E75xx chipsets. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Tony Mountifield Sent: Monday, 7 February 2005 6:53 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: TE410P card in an HP-Compaq DL380 G4 server In article [EMAIL PROTECTED], Peter Childs [EMAIL PROTECTED] wrote: Contact Digium Support. They have been very helpful with this issue (mention your using the G4 server with the Intel E7520 Chipset..) So do they have a solution? What is it? Cheers Tony ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] HP ProLiant server for Asterisk
Digium support are trailing some new firmware with the TE410P for machines with the Intel E75xx Chipsets that are having issues (such as the DL380 G4). I believe they are confident they have resolved the issue that prevents the cards working, but you may need to specifically mention that you are running this type of machine when acquiring the cards to ensure you get the 'in-testing' firmware. I'm sure as soon as someone gets and tests the new firmware on one of these machines they will post their results to the list. Regards, Peter -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Dana Olson Sent: Saturday, 5 February 2005 1:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] HP ProLiant server for Asterisk I'm looking at ordering a server from HP. I checked around on Google and found in the Wiki that the ProLiant DL380 is supposed to be known to work with *. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TE410P card in an HP-Compaq DL380 G4 server
Contact Digium Support. They have been very helpful with this issue (mention your using the G4 server with the Intel E7520 Chipset..) Cheers, Peter -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Erick Perez Sent: Saturday, 5 February 2005 5:51 AM To: Eric Bishop; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] TE410P card in an HP-Compaq DL380 G4 server Has anyone on this list have a way to contact ServerWorks? they make the mobos for the G4. I dont have a G4 but i do know HP in the G line uses ServerWorks I have to make a full stop ordering on 2 G4 monsters because of this thread...However one friend is using a sangoma card without problems TE410P/ServerWork motherboard combo not working because of bus problems my less than 1 cent On Mon, 31 Jan 2005 20:42:47 +1100, Eric Bishop [EMAIL PROTECTED] wrote: Did anyone get anywhere with this thread? Any HP G4 series servers working? On Wed, 26 Jan 2005 09:46:31 +1100, Eric Bishop [EMAIL PROTECTED] wrote: Has anyone had any luck with this issue and new Asterisk/Zaptel releases (1.05/1.04)? I am still searching for a solution and waiting for that Eureka! moment.. On Thu, 20 Jan 2005 09:20:09 +0100, Tais M. Hansen [EMAIL PROTECTED] wrote: On Wednesday 19 January 2005 23:15, Eric Bishop wrote: Well guys this is truly bizarre. I managed to get a DL360 G3 to show interrupts with FC2 but not FC3. Exact same config and setup proceedure. Ofcourse neither FC2 or FC3 show interrupts with the DL360 G4. I think TE410P is just a flakey card. Anyone got a DL360 G3 going with a TE410P and FC3? I did manage to get a TE110P running on the DL380 G4. Still can't get the TE410P working in the G4 though. Supports your theory. Sadly we're now being forced to look elsewhere for PRI cards. -- Regards, Tais M. Hansen ComX Networks A/S Tel: +45-70257474 Fax: +45-70257374 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Some more hardware and E1 questions
Personally I would for the time being steer clear of anything with a Intel E7520 Chipset (newish) such as the HP DL380 G4 etc... if you are using a TE410P card (ie 3.3v). Just my 2 cents. But you can always give it a go :) Cheers, Peter -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Daniel Nyström Sent: Thursday, 20 January 2005 11:55 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Some more hardware and E1 questions Hi again folks! ;) As before, I will transform one E1 30 Channel PRI into 30 FXS channels using Adit 600. Now I'm into choosing server platform. And the two opponents are: * Dell PowerEdge 750 w/ SCSI RAID (or even SATA RAID1) * FujistuSiemens PRIMERGY RX100 S2 (SATA RAID1) As I've seen people having problem with HP server, I havn't looked at it at all. What experience do you have with the alternatives above? Which would you recommend? And another question at the same time; what's really E1? How is E1 devices connected? Seems like regular Cat5 cables, but it problably ian't? If anyone's using Adit 600, did they send all cables required for connecting to the FXS channels? Seems like a very unique plug on the side of Adit. Thanks! BR Daniel Nyström ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TE110P as E1
Sounds like you don't ever get Layer 1 up. I'd check your cabling (pins 1,2 and 4,5) Should span=1,1,0,ccs,hdb3,crc4 be the way to go (aren't you getting clocking from your carrier...) Good luck. Cheers, Peter -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Andre FAURE Sent: Tuesday, 18 January 2005 11:59 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] TE110P as E1 Hello, I'm having problem with a wildcard TE110P. As soon as I load the module (wcte11xp for kernel 2.6.10), it spawns a yellow error with or without an E1 plugged-in. Any one managed to set it up in France? Here are my files: zaptel.conf: span=1,0,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 zapata.conf: [channels] language=fr context=default switchtype=euroisdn pridialplan=unknown prilocaldialplan=unknown signalling=pri_cpe echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no group=1 channel = 1-15 channel = 17-31 I also tried without crc4 and using signaling=pri_net. Thanks for any idea. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TE410P problem (Looping UP Span 1...) [digium.com #13999]
The the 'common' factor here appers to be the Intel E7520 Chipset. I have a NEC 120Rg-2 here with this chipset with the same problem. This chipset exists in the HP DL380 G4 Server, and the machine mentioned below. Someone else mentioned the same issue on a new Dual Xeon EM64T capable Tatung server, and some searching on their website shows TSS-2552 also uses the Intel E7520 Chipset Perhaps its just coincidence? Cheers, Peter -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Sid Sent: Tuesday, 11 January 2005 7:35 AM To: Eric Bishop; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] TE410P problem (Looping UP Span 1...) Hi, This is the motherboard: SM X6DAE-XG2 Dual XEON 800FSB EMT64 w/2-Ch SATA-R 01,SVGA,2xGb LAN Dual Intel. Xeon EM64T Support up to 3.60 GHz Intel. E7520 (Lindenhurst) Chipset 1(x8) PCI-Express on (x16) Slot, 3 x 64-bit 133MHz PCI-X, 2 x 64-bit 100MHz PCI-X Slots ATI RageXL 8MB Graphics Actually it doesnt have 4 cpus as i mentioned in my earlier mail. It has 2 XEON cpus with hyper-threading technology. Anyone has seenexperienced any problems with this motherboard? -Sid --- Eric Bishop [EMAIL PROTECTED] wrote: Is it a HP-Compaq DLXXX G4 machine? Because there is a thread here about the TE410P not generating interrupts with these servers... On Sat, 8 Jan 2005 05:14:39 -0800 (PST), Sid [EMAIL PROTECTED] wrote: Hi Scott, and Jack, --- Scott Stingel [EMAIL PROTECTED] wrote: Sid- Try connecting one port to another. Note that one of the ports must be set up as cpe and the other as net in zapata.conf when you loop them together like this. A suitable crossover cable for testing can be constructed by cutting up a CAT 5 cable, and connecting: Pin 1 -- Pin 4 on the other end Pin 2 -- Pin 5 Pin 4 -- Pin 1 Pin 5 -- Pin 2 You should get green's on both the connected channels if your zaptel and zapata configurations are ok, and if you've run both modprobe and ztcfg as documented. Thanks for the valuable responses. We can only do the tests on monday, as the machine is in a data center. Other than that we have done every tests we can think of and found in the mailing list/wiki. The tests done at the NOC says that T1 is ok at their end. Please see the following information about the system: The machine has 4 Xeon 2.80GHz CPUs. This is from /proc/interrupts 16: 0 0 0 3 IO-APIC-level usb-uhci 19: 0 0 0 0 IO-APIC-level usb-uhci 23: 0 0 0 0 IO-APIC-level ehci-hcd 26: 129024 0 0 28 IO-APIC-level eth1 27: 0 0 21352 5 IO-APIC-level eth0 76: 0 0 0 0 IO-APIC-level t4xxp # dmesg Zapata Telephony Interface Registered on major 196 Specify address with base=0xN Registered Tormenta2 PCI Found TE410P at base address fc8ff800, remapped to f8a40800 TE410P version c01a009b FALC version: 0005, Board ID: 00 Reg 0: 0x371c9800 Reg 1: 0x371c9000 Reg 2: 0x07fc07fc Reg 3: 0x Reg 4: 0x Reg 5: 0x Reg 6: 0xc01a009b Reg 7: 0x1000 Reg 8: 0x Reg 9: 0x00ff Reg 10: 0x TE410P: Launching card: 0 TE410P: Setting up global serial parameters Found a Wildcard: Wildcard TE410P-Xilinx Registered tone zone 0 (United States / North America) TE410P: Span 1 configured for ESF/B8ZS SPAN 1: Primary Sync Source I am doubtful about the interrupts. Are those values ok? We have been after this problem for more than a week now, we have tested with 2 different cards to no success. I'll post the results of the crossover connection test, once we do that. Thanks again for the responses. BR, -Sid __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] NEC Univerge
I have one sitting around here, only the SV7000T module though (not the SV7000S sip handset server), and a few handsets and media gateways. The SV7000 is basically just the CPU module removed from their IPX hybrid tdm/voip models for customers that want a 'pure ip' solution. The software is pretty much identical to the IPX line of software, so its background is traditional PBX as opposed to VOIP switch. As such things like having two modules for redudancy with fast failover is 100%. Solid state storage. If you need to setup some sort of large distributed voip system with transparency there are quite a few options such as 'fusion' (each nodes as a part of a single-pbx-system, but is still capable of self operation), or the surviable remote gateway (like redudancy, but remote from the core platform) Very mature software, very stable. However its build from a voice perspective, not a data perspective, and its not a open source system. The dtermIP telephones are very nice units, however they are not SIP telephones, they run NEC's proprietory 'protims' protocol over IP, with a few other smarts. I managed to get some basic telephony working with a bit of back end java 'pretending' I was a protims server, and was thinking of writing an asterisk channel, but I think that would be the end of my job *grin* Horses for courses. Cheers, Peter NOTE the views of myself do not represent the views of my company, well its not really my company, but you know what I mean. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of el Flynn Sent: Thursday, 9 December 2004 1:15 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] NEC Univerge hi * users, anyone out there ever come across, worked with or heard anything about NEC's Univerge SV7000 Telephony Server? Link is at http://www.univerge.nec.com/products/list/sv7000/sv7000.html I'm just wondering whether it's as flexible, programmable and configurable as Asterisk. It also looks like they've got a whole range of IP phones (http://www.univerge.nec.com/products/list/ip_pho/ip_pho.html), anyone ever used one of these? I wonder what they're going for... Flynn ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re:RE: NEC Univerge
My communications was as a PH-D handler and DRS server for the ip telephones. I'm sure the CCIS/IP would be a _whole_ different kettle of fish, and I'm sure a NEC SIP-MG for the SV7000 would be around (perhaps only in the NEC/J marketplace, but soon to others...) My thoughts of attempting to get perhaps a Protims/IP channel in asterisk is that 1. asterisk could be a server for dtermIP handsets (nice handsets, pretty reasonably priced without a NEC Right-To-Use license..) 2. asterisk could be a server for voicemail/ivr/conference by 'registering' as a bunch of extensions on a IPX/SV7000 The proof-of-concept stuff was enought that I figure that I could do it, but its non-trival.If I didn't have kids I'd probably do it as a pet project, but since the munchkins are around I would rather spend my free time with them :) If my work paid for me to do it that would be great, but I don't think that will happen (you never know... perhaps I'll ask..) Cheers, Peter -Original Message- I managed to get some basic telephony working with a bit of back end java 'pretending' I was a protims server, and was thinking of writing an asterisk channel, but I think that would be the end of my job *grin* so are you ripping off an IP CCIS stream and simulating a CCIS node? I was thinking about trying to put a t-berd on a t-1 CCIS span and seeing if I could interpret the sigalling on the common channel to build a CCIS channel for * as well but why do that if you can do it over IP? whatever, IMHO the SV7000 Univerge is a great idea poorly executed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call ID WinPopup working one-line example for YAC
http://sunflowerhead.com/software/yac/index.html
You only need to run the client..
exten = s,4,System(/bin/echo -e '@CALL${CALLERIDNAME} ~${CALLERIDNUM}' |
nc -q 0 -w 1 pjcm400 10629 )
Does the trick for me... and YAC has a nice caller history log etc (and
I do like those nice windows ballons etc..!)
The 'nc' here is of course 'netcat' and the options are -q (quit after
EOF), and -w 1 (only wait for connect for 1 second...)
Thanks guys, this is cool!
Cheers,
Peter
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Thomas
Hutton
Sent: Thursday, 18 November 2004 2:54 PM
To: AsteriskUserMaillist
Subject: [Asterisk-Users] Call ID WinPopup working one-line example
withoutscratch file
Here's a tested example that works without any scratch file. I still
had to use a combination of single and double quote characters, as well
as a double backslash for the \n newline command.
; Extension 200 Call ID Popup Example
exten = 200,1,NoOp(${CALLERID} ${DATETIME})
exten = 200,2,System(/bin/echo -e 'Incoming Call From: ${CALLERID} \\r
Received: ${DATETIME}'|/usr/bin/smbclient -M target_netbiosname)
exten = 200,3,Dial,sip/tom|30|t; Ring, 30 secs max
exten = 200,4,Congestion
Note: line two wrapped - it needs to be all on one line.
Thanks to Duane and Adam for the ideas.
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RE: [Asterisk-Users] sip trunking works?
'SIP trunking' is something I've heard before mainly from traditional switch vendors that are having trouble with SIP as they are used to 'stations' and 'trunks'. You should find asterisk very capable as a 'toolkit' box, as it can 'register' like a traditional SIP client with a SIP Registrar, or act as one, or both, or neither! Its pretty flexible. I'd be interested how you go with the Alcatel. On the SER (www.iptel.org) site they have a couple of page intro on SIP which is worth the read and will put you miles ahead of 90% of the ICT world which has problems spelling SIP :) Good luck, Peter -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of David Hajek Sent: Tuesday, 9 November 2004 8:51 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] sip trunking works? Hello, I'm about to connect asterisk with Alcatel Enterprise PBX using SIP trunking, I can't find if Asterisk has this capability. Can you please advice? Thank you. -David ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] sip trunking works?
Appologies I should have put a link http://www.iptel.org/ser/sipintro.html Its only 20 pages, and they are pretty small pages :) Cheers, Peter -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Paul Rodan Sent: Wednesday, 10 November 2004 5:57 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] sip trunking works? Are you talking about the 187 page SIP tutorial? What couple of pages are you referring to? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peter Childs Sent: Tuesday, November 09, 2004 4:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] sip trunking works? 'SIP trunking' is something I've heard before mainly from traditional switch vendors that are having trouble with SIP as they are used to 'stations' and 'trunks'. You should find asterisk very capable as a 'toolkit' box, as it can 'register' like a traditional SIP client with a SIP Registrar, or act as one, or both, or neither! Its pretty flexible. I'd be interested how you go with the Alcatel. On the SER (www.iptel.org) site they have a couple of page intro on SIP which is worth the read and will put you miles ahead of 90% of the ICT world which has problems spelling SIP :) Good luck, Peter -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of David Hajek Sent: Tuesday, 9 November 2004 8:51 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] sip trunking works? Hello, I'm about to connect asterisk with Alcatel Enterprise PBX using SIP trunking, I can't find if Asterisk has this capability. Can you please advice? Thank you. -David ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] mwi over serial port
I may have missed something here but couldn't you just do this with a bit of bash/perl/etc using 'externnotify=' option in voicemail.conf file? I do this to set MWI via OAI (CTI) on a NEC switch without having to 'integrate' heavily. If you just need those bits you could probably just echo them out the port (?) Cheers, Peter -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Clay Zevely Sent: Wednesday, 13 October 2004 9:55 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] mwi over serial port I am trying to interface to a nortel dms100 and the only feature I have failed to figure out is the mwi. On the system being replaced they use the rs232 to activate and deactivate the mwi. Can I use teh serial as well on asterisk. An example I am looking for is as follows. at 9600 E 7 1 on the serial port (Activates indicator to station) OP:MWI_xx![Control D] (Deactivates indicator to station) RMV:MWI_xx![Control D] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP Phone - PBX Phone
What type of existing PABX do you have (Make and Model) What interfaces can you use to connect to your PABX, ie analog tie lines, E1/ISDN, anything else? Cheers, Peter -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of P J Sent: Friday, 17 September 2004 12:19 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SIP Phone - PBX Phone Hi, I'm new to Asterisk, and am researching information on linking Asterisk to an existing PBX. Could somebody please help me with what might be required for the following setup? - - We have an existing PBX. - I am going to setup Asterisk on our internal network along with some internal SIP phones. - I understand how Asterisk will act as the SIP Server, and SIP phones will be able to call each other, however - - I would also like to link Asterisk to our existing PBX so that SIP phones could call standard phones on our existing PBX system (and vice-versa). - I *do not* need to use Asterisk to call out via PSTN or ITSP. All outbound calls will be via the existing PBX. What hardware device is required to link the Asterisk box to the existing PBX? Could the SIP phones call the standard phones on our existing PBX system? If so, how does Asterisk do this? Thanks in advance. Ps. Even though I *do not* need to use Asterisk to call out via PSTN, what hardware device would be required to do this? And, how does this device differ from the device that links Asterisk to the existing PBX? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] * and Philips IS3090 PBX
For what its worth I think you'll find the Philips IV2000 voip switch is actually a NEC IVS2000. It does support IP telephony, and I have read that H323 trunking is possible (only _read_ it though).Handsets are NOT anything you would call 'standard'.I have no idea about the IS3090 though. Good luck. Cheers, Peter -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Terry Wade Sent: Wednesday, 15 September 2004 8:29 PM To: Asterisk Users Mailing List Subject: [Asterisk-Users] * and Philips IS3090 PBX Hi I have been playing with * for the last couple of weeks now. I am also speaking to one of my customers about installing a * server in addition to their Philips IS3090 switch. They are busy building a new office block and I have convinced them to go VoIP. Currently the client is thinking about a Philips IS2000 Voip switch. I am thinking * solution. 1) Would I use one the T1 cards to interconnect the two, does anybody have any experience with the Philips switches? 2) Would I be able to make calls to the local PSTN via the IS3090? 3) Do I create the extensions on the IS3090 or the *? 4) Would the IP phones be able to use the Kinesis voicemail on the IS3090? 5) Would the TMS still log all the activity from the IP phones? 6) Will the cell/mobile routers still make their allocated calls? 7) They are also running dect phones, so this would this be able to work as well? Sorry for all the questions, but this could be a big deal for me. Starting with 150 new ext's and can replace upto 750 Ext's. Thanks for advice in advanced Cheers Terry ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] digitnetworks card issues?
I have the same hardware (x2)
/etc/zaptel.conf file
fxsks=1-2
loadzone=au
defaultzone=au
/etc/asterisk/zapata.conf file
[channels]
language=en
context=inbound
group=1
musiconhold=default
; need these much shorter than defaults
flash=90
signalling=fxs_ks
threewaycalling=yes
transfer=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
;busydetect=no
busydetect=yes
;busycount=6
callprogress=no
channel = 1
channel = 2
from /etc/asterisk/extensions.conf
exten = _X.,1,Dial(Zap/g1/${EXTEN})
I had some noise issues at first, and then I used a decent shielded cable
between the cards and the wall socket and that cleared it up...
YMMV.
Cheers,
Peter
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Imran Akbar
Sent: Friday, 3 September 2004 4:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] digitnetworks card issues?
Hi,
I've purchased two x100p clones, and when I try accessing a line
from asterisk with something like this:
exten = _1NXXNXX,1,Dial(Zap/2/{$EXTEN})
(is that only supposed to put you on channel 2 or actually dial the #
for you?)
but I first hear noise, then a dial tone, but as soon as I start dialing
numbers I get feedback and noise, and the call doesn't go through.
Any suggestions?
___
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