[asterisk-users] Need provider recommendations for the UK
Hi, I realise this is probably the wrong list for such a question, but I need a pointer to somewhere I can get some feedback on experience of (business class) voip providers for the UK? Situation is that we are currently with Gradwell and use them for an inbound/outbound single line for a business and their quality has gone from excellent to abysmal in the last few weeks. I'm sure they will work it out, but right now I just need a reliable provider that I can port a number to. I'm not especially price sensitive, reliability is the main requirement. IAX preferred, but not fussy. Possibly multiple incoming numbers in future, single incoming at the moment - in general we rarely have more than 1 line in use, but occasionally hit 2-3 simultaneous calls Note, it's going to be important that we can port our number across from Gradwell... Grateful if anyone can offer some really solid recommendations, or point me towards a more appropriate forum to request the same? Thanks Ed W ___ -- 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
[asterisk-users] Sangoma and BT single lines
Hi, got a Sangoma A200 with a bunch of extension cards and having real problems getting it to deal with a normal single BT line Symptoms are that incoming calls are fine. Outgoing calls ring the far end, BUT asterisk never sees that the call is answered (ie no message in the logs files saying so), as a result the remove end can hear the PBX side talking, but there is no audio back from the remote side to us. When we hangup the log files show messages thave suggest it thinks the line is still ringing Comparing with another line which works fine (this is a BT multi-line system with what they call PBX signalling on it) I see that as soon as the remote end answers then asterisk gets a log message stating the same and audio is fine on this line Have now spent nearly 4 months trying to get the signalling sorted on this line. Most recently we requested dual signalling on the line - the end result is now that outbound calls work and asterisk reports that the phone answers, however, when you hangup the call then asterisk obviously gets a bunch of extra line reversals and things there is an immediate incoming call on the back of that outgoing call... Please - any suggestions on how to configure a Sangoma card for use with a normal BT single line? Thanks Ed W ___ -- 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] Recommend Wireless IP Phone
Wilton Helm wrote: Wi-Fi SIP phones aren't limited to hot spots. I am in the process of setting up asterisk for SOHO. At present I'm not even using VoIP trunking, only LAN to stns and I intend to use Wi-Fi instead of analog cordless phone. I got the Engenius one, and it works, but I haven't played with it much. I was disappointed that it only has a single line appearance, as part of my reason for going SIP was to allow the same features like say my 941. I also got their 600 mw access point, but haven't had time to try it. My goal is to cover out 3 acre property and the 1/2 mile road to the mailbox, including mountainous terrain. Maybe I'll share more when I actually get it all put together. I think you will get better range and longer battery life from a DECT phone though... Probably more features and better quality also! Any of the panasonic DECT phones seem to work very nicely (speaker phone, features, R key works for call transfer, handset intercom etc - mine lasts up to a week on a single charge and light use) and there are several Siemens DECT phones with a builtin SIP gateway which avoids the need for an external adaptor box It's definitely possible to make wifi work for half a mile and you don't even need a 600mw transmitter to do that - however, wifi is all about receive strength, and so you are unlikely to get a significantly better coverage with a high power hotspot which is suboptimally placed. If you do go that route then getting the antennas into a location where 90% of the signal isn't already killed going through walls before it has to travel some distance is the trick. Probably also consider a repeater of some sort rather than just one high power device Good luck though! Ed W ___ -- 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] Polycom's lose BLF after Asterisk restart
Mr Shunz wrote: Hi, We have an issue where Polycom's lose BLF functionality after a reboot. The only way to fix it is to reboot the Polycoms. Anyone else have this issue? We are using 1.4.18. If I run 'sip show subscriptions' all the subscriptions come back after the restart but the lights on the phones do not work. Any help would be appreciated. have the same issue with grandstreams and thomson (at least on st20XX) if we restart asterisk, phones don't renew subscriptions ... didn't search too hard, but i haven't found neither an option in asterisk nor on the phone to force resubscriptions ... Can you reboot the phones remotely? With snom it's quite easy to write a script to reboot all phones - you can put that in your boot scripts Ed ___ -- 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] Hang up detection with TDM400P and Telewest/Virgin Media line
robb wrote: I have a TDM400 working quite well, Digium dialled in and recompiled chan_zap with some changes , to get BT Callerid working and I have set hangup on polarity in the zaptel.conf which seems to work well this is a BT home line, not business, if you have a business line you should get the DCT set to 800ms and the disconnect clear should work Would you be kind enough to share the changes you made to get callerID working please? Any chance of posting the relevant bits of your zap config also? My situation is that I have callerid working most of the time on a home BT line. Hangup is fairly reliably detected. TDM400P However, at a customers site on a bunch of business BT lines and the same model of TDM400P we see unreliable hangups (not frequent, but occasional times that lines are getting stuck off hook). Also callerId is working about 50-60% of the time and when it doesn't work (or genuinely that the callerId is witheld) there is a long pause for about 2-3 rings before Asterisk answers the zap line. It would be desirable to limit this pause because it makes it look like they are being slow to answer all the calls! Just wondering what changes you made? Also, anyone understand why DCT is different between home and business lines? Can the Zap code be changed to avoid needing something tweaking on the exchange? Thanks Ed W ___ -- 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 Openfire Asterisk-IM Plugin Performance Observation
Is it better for production to run Openfire on a separate server than the PBX? Since discovering linux vservers I put every service into its own install. Each install can be very lightweight and vservers only add about 1MB to ram usage (I don't run a separate init process), so very lightweight. The advantage is that it's super simple to backup each server and you can test upgrades by simply copying the image, fire up a new instance, test your upgrade, then burn it down again... Piece of cake to shuffle services between real machines also (preserving IP addresses also if that's required). Backups can be done very easily (make the /vserver dir an LVM disk) Good luck Ed W ___ -- 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] Phantom Rings
I'm fairly certain the problem is with the phone line. I have all callerID settings disabled as the Telco is unable to provide it along with our rollover line setup due to limitations in their antiquated switch. The CLI and Logs all plainly show the calls as if they were normal calls with the exception of a message about Failed to write frame and no DTMF attempts, then the call is routed into the operator queue. The calls always came in on Zap1-1 so I tried swapping the 2 lines to see if it stayed on port 1 or if the phantom followed the line. As expected, the phantom rings followed the line and began showing up on Zap2-1. So it pretty has to be something in the telco, but I'm not sure what. Putting WaitForRing(3) before the Answer command in my IVR menu eliminates most of them, but sometimes more of them slip through. I get a similar problem with a domestic analogue line in the UK. I *speculate* that there is a short half ring being sent for some reason (line test or similar), but my card (Digium) seems to need about 5 seconds to detect hangup on the remote end, so I get a phantom 2 rings at my end and then it stops... No solution, but thought it might give you something to consider... Ed W ___ -- 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 hookup to cell phone for outbound calls?
Sam Tam wrote: Well I think you need a GSM Gateway You can find some info on cyber-telecom.net For a cheap option you can try a CT-G1000 or CT-G2000 and then plug it in a X100P or something similar then it would be very economical. Yep, this is the kind of thing I am after, except my hardware PBX has limited connectivity and ideally I want a USB or ethernet hookup to the box..? The scenario is basically a small commercial PBX (small form factor) which can be supplied with IP phones and will talk out via a cell phone channel (or via a satellite phone if that's the only option available, but this is out of scope of this question). So basically I want to figure out some options to hookup a GSM cellphone channel to a small form factor asterisk PBX which has limited expansion options (ethernet, USB and mini-PCI - although prefer to use the later for a wifi card...) I only need a single channel of GSM right now (and a single SIM) Any thoughts? Remember this needs to be production quality and priced sensible for a commodity market Thanks for pointers to hardware Ed W ___ -- 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] Matching + characters in dial plan
Can someone please explain how to match a + character in a dial plan (so that I can swap it for the 00 country escape code). In Europe at least the + is a common shortcut for the international prefix (which is 00 in my country). However, my trunk chokes on the + character and all my speed-dials are setup with a + at the start of them... Trying to fix the phone rather than the addressbook... Thanks Ed W ___ -- 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] How to hookup to cell phone for outbound calls?
Hi I need a small PBX for use on the move. This means that outbound calls will need to be made over the cell phone network. Assuming a small hardware PBX with a spare mini-PCI slot or a USB slot then what hardware options do I have to get an outbound cellular channel? Options need to be rock solid, so no bluetooth to a cell phone kind of solutions need apply. Can any of the 3G usb devices out there offer outbound analogue calls (ie other than via voip)? Cheers Ed W ___ -- 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] Poll: Asterisk IMAP feedback (was: Is anyonesuccessfully using IMAP storage)
Anthony Rodgers wrote: We tried with MS Exchange but couldn't get it to work (MS Exchange doesn't support a master account). It used to? Not out the box though... Ed W ___ --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] Flash IDE
Juan Sandro wrote: Hi We have a number offices accommodating 4-6 people each hence it is very important for PBX to be fanless and silent. We have been looking at using IDE flash disks also called DOM. The performance tests we have done so far satisfy our requirements, however we are concerned with DOM durability. We have installed debian and vanilla asterisk on 1GB DOM. All seems to work fine at the moment however will DOM last? How long it will last? Is anyone able to share similar experience? Any other information/tips? I worried a lot about the same, in the end I went for a small laptop drive for safety (it's inaudible) However, this came up on slashdot recently and if you search around the logic seems to be that: - Flash rewrites quite a few times - The good stuff has wear levelling so that most roughly speaking the whole thing should work until it suddenly all fails - Given a big enough drive with a fair bit of free space then you should find it hard to wear it out in less than quite a few years even if you are hitting it quite hard (probably multiples of this). Simply do the maths to get the rough life So basically it seems that given a large enough flash drive with decent wear levelling the lifetime should be completely ample... ...Thats the theory anyway. I feel quite bullish about the whole thing, but I think I would avoid the *really* discounted cheapo flash drives since they may not have the correct wear levelling. Decent brand names should be fine though (and you can google for details on their specs) Ed W ___ 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] How can I improve call quality?
Check first using something like testmyvoip.com to get an idea of your situation (stress the internet by opening up lots of simultaneous downloads during the test) Repeat: Try the above before you do anything else... Ed W ___ --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] UK zaptel and zapata.conf for TDM400P
Hi usecallerid=yes cidsignalling=v23 cidstart=polarity Although this is what the wiki recommends, I just couldn't get the cidstart=polarity to play well with immediate=yes, I kept loosing the callerid? This is what I ended up with and now it avoids the annoying 2 rings before the internal extensions start to ring. However, I still have a problem in that if someone hangs up while still in ringing state then asterisk continues to ring for 2 more rings (roughly). This is annoying because BT appear to do a line test every 30 hours or so and so my lines ring for 2 rings at random times of day or night [EMAIL PROTECTED] asterisk]# more zapata.conf ; ; Zapata telephony interface ; ; Configuration file [trunkgroups] [channels] language=en context=from-zaptel signalling=fxs_ks rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no echotraining=800 rxgain=0.0 txgain=0.0 group=0 callgroup=1 pickupgroup=1 immediate=no ukcallerid=yes cidsignalling=v23 cidstart=ring ;cidstart=polarity ; Added for UK CLI detection sendcalleridafter=0 immediate=yes ; as we recieve cli info before not after first ring. answeronpolarityswitch=no ___ --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] How can I improve call quality?
Hi Also - are there any useful stats/logs that I can examine to see the quality of calls? You didn't mention that you have any QOS on your router, so we can basically guarantee that your problem is the internet connection. Remember that all the research on networking has been how to saturate a single connection and download as fast as possible, so when some spod hits a website and reads a web page then he grabs basically the whole connection for a short space of time. During that time your voip packets tend to loose out and get delayed - the jitter buffer does some stuff to try and compensate, but ultimately it will loose Add some kind of priorisation to the T1 line and your quality should go up dramatically Check first using something like testmyvoip.com to get an idea of your situation (stress the internet by opening up lots of simultaneous downloads during the test) Cheap fix is to get a separate DSL line and run the voice over that... Ed W ___ --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] How can I improve call quality?
Our 'net link is a dedicated 2Mb fibre connection (of which we have ever used 50% max bandwidth). Remember in computer terms this means that you used 100% of the connection, 50% of the time Your voice will loose out against the big data packets and spoil the voice quality big time Ed W ___ --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
[asterisk-users] Hardware suggestions for 8-10 lines in the UK
Hi I have previously had good success on smaller installations with TDM400P cards. I now have a UK customer looking for 8-10+ lines and it seems like a PRI would be most economical + reliable? Has anyone here used PRI interfaces in the UK and can confirm that it works well (using Trixbox for preference?). I have had some niggling issues with the TDM400P cards which puts me off adding lots of these in a single box. Am I right in thinking that ISDN or PRI will be a better and more reliable option? Any suggestions on 2U servers that should work well? For example the DELL 2850 seem to lack any spare HD type power outlets which is irritating... I also need two internal fax machines. Can anyone confirm that the best solution is just some linksys ATA's at the fax end, then switching them directly down the PRI card? Is there a better option to guarantee best quality? It would be convenient not to have extra analogue lines in the building if we go down the PRI route... Grateful for any thoughts Ed W ___ --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] Hardware suggestions for 8-10 lines in the UK
Best bet would be to talk to insert telco of preference and ask them what they recommend. For anything more than about 8 channels a PRI is likely to be the most cost-effective route if you need physical lines on-site, especially if that's likely to grow beyond 8 lines in the future. Just for reference: - Called BT and their prices are basically the same for PRI as they are for analogue (~£10/month), but you need to buy a minimum of 8 lines for a PRI. You can then increase decrease in single lines. DIDs are quite cheap also (practically free) - Installation cost is cheaper on PRI than analogue, but they will waive the installation cost if you buy something else from them such as broadband or a mobile phone Sounds like PRI is the way we will go therefore as long as the equipment is reliable. Can anyone recommend PRI cards which are known to work flawlessly with Euro ISDN 30? (FWIW, BT tell me they now supply all new lines as Euro standard instead of the v85 that it claims on the voip wiki) You've obviously had better success with the TDM400Ps than I have in the UK. Certainly the call quality on asterisk installs we've done around Northampton/Milton Keynes has improved markedly since we switched to delivering calls via SIP or IAX rather than using physical lines on-site. The clients are also happy - each less line they have on-site is a saving of at least £10/month. Agreed. My experience is that quality is higher on Voip than it is via a TDM400p. However, my experience hasn't been that VoiP is as reliable as copper lines and so unless you can tolerate the odd outage once per month or two then you might want to stick to copper for the main carrier? Does this match with the experience from others? Still after recommendations on a server box (2U with space for a couple of PCI cards would be sensible), the PRI card and also any ATA adaptors which are known to work well with fax units Cheers Ed W ___ --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] Best phone for easy provisioning
Paul Hales wrote: I know Brett and Jurgen have been pretty happy with the Snom's - Brett even wrote an auto-provision utility for the Snom's at one time. Yes, look at the latest Trixbox for the basic SNOM templates and then off you go. You setup a tftp server (easy), the phone looks for two files, one generic snom file and another with it's MAC address in the name. So you have generic stuff in one file and specific phone stuff in the other. You can get the initial phone config done using DHCP, or just log into each phone and change the URL to point to your tftp server. Get the phone roughly setup manually and then simply look at the web config utility, it has an option to dump it's entire config out and so you just cut and paste the entries you want to override into the tftp files. Piece of cake. You can easily reboot all the phones by sending them a certain SIP message, and so it's very easy to redeploy a new config, or reboot all the phones when you reboot the server. The phones themselves re-read the config every X minutes so they pickup new config quickly even without a reboot. I have a bunch of 360s which I negotiated for about the same price as the 320s. Drop me a line if you want the name of a UK firm to buy them from. They work nicely out of the box including the flashy lights showing busy extensions. The only thing which doesn't work without a patch (it appears) is line pickup by pushing the BLF keys. I can live without that, but it would be nice to have. Happy to post my configs if anyone wants to write up the notes on the wiki? Ed ___ --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] Diagnosing poor call quality
Check from the sites in question using testmyvoip.com or whatever the site is called. In the UK I found that some strange things sometimes happen. At one point I was sure that BT were perhaps misclassifying IAX packets as P2P... However, not had a problem with SIP. Beware that ADSL uses vastly more bandwidth than you expect on small packets, eg if you are classifying using a cheap router then you probably need to at least half your claimed bandwidth in order to make the prioritisation work correctly. I added some (hack) patches to fix the linux calculation for HTB on the linux QOS list a year or two back. If you have a linux router you could use those to improve the calculation quality for QOS - or else I found a Draytek router does impressively well at getting it right for small sites... Very likely you will find that the issue is variable jitter on the line. The link above should help you figure this out Good luck Ed W ___ --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] Re: Help - Poor Voice Quality
Hi Yes, I know that I am using IAX2 and not SIP for my connection to teliax. IAX2 is the preferred protocol for connection to teliax. I have the firewall configured to prioritorize port 4569 for IAX2. 1) 4569 is only the IAX setup port. Edit rtp.conf to limit the rtp ports to some subset and then prioritise those instead 2) Uplink bandwidth is always the constraint on these lines. This is highlighted in this case 3) Shorewall can't correctly prioritise bandwidth whenever using some kind of DSL service or whenever the packets are encapsulated such as the cable service. Read the linux QOS faq for more info and as a workaround slash the theoretical bandwidth in half in your shaping script. This should get you working and you can tweak later 4) Monitor the QOS buckets as you make/break calls to check that all the packets are classified correctly. Otherwise your voip packets might be accidently in the bulk box Basically VOIP goes from perfect to horrible when the jitter rises and packet loss goes up. Probably this is happening in your case Good luck Ed W ___ --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
[asterisk-users] Glitches in voicemail prompts
I changed from using a recent asterisk system standalone to a Trixbox install and now I get clicks and minor dropouts on the voicemail prompts. System load is non-existant on this machine, interrupts *appear* to be fine, and as near as I can tell the glitch is at the same point in the prompt each time... Any suggestions on how to debug this further? To my ear it sounds like what happens when you get an overflow in some decoder code and the levels have wrapped around? Any thoughts? Ed W ___ --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
[asterisk-users] Dial plan constructions suggestions?
Can I ask for some advice on dial-plan construction please I have setup my dialplan to use 9 to get a zap trunk, leaving everything else for internal extensions. However, this creates a problem in that my callerid is correct, but doesn't work to re-dial the incoming caller. So if I simply click missed calls on my Snom phone and hit redial then it tries to dial an internal extension. So I then setup Asterisk to add a 9 to the incoming callerid for all calls which come via the Zap trunk, but now this creates some issues with applications like Snapanumber and perhaps HudLite, which are trying to map the caller ID to numbers in the addressbook (and I don't really want my internal Outlook address books to have everyone listed with a 9 in front of their number) How are others handling this? I have considered simply dropping the prefix digit and working around any clashes in internal and external numbers (not very hard). Grateful for any thoughts Ed W ___ --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] Dial plan constructions suggestions?
Hi There was a thread about this not too long ago, so the archives may have a bit more on it... The way I handle it is by forcing the caller to dial the full number starting with zero (normally 10 or 11 digits in the UK - which I'm guessing you're from too) Yes, I use something similar on another box, but there I support shorter dial codes as well. It's not to hard to make 8 dial 0208 or 7 dial 0207, etc. I happen to also map some of the 1xx codes across as well. It's still not a complete solution though because on this other box I have a business line and a personal line and I send calls to different lines based on the type of call (or more usually the time of day...). I want to have seperate billing basically. When the call comes in it makes sense to have the caller tagged with (in my dialplan) 9 for a personal call, and I use 3 (for no good reason) for my business line. I actually have one phone which defaults to business line if I don't add a prefix, another DECT phone which is my personal phone, but I can see on either where the call is coming from and also force the call to use a different route just by dialing the prefix. Basically it's tricky. I do already use custom ring tones for each line, so I guess I could drop the prefix, but it's nice to have it so that I can see at a glance whether it's a business call or not... Any other suggestions? Any suggestions on other software than Snap which does callerId lookup from Thunderbird (not Outlook). For example is HUDLite ever going to support Thunderbird...? Cheers Ed W ___ --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] Re: Dial plan constructions suggestions?
Hi I had the same situation, in that I wanted to be able to use the Voicemail 'dial back' feature, and had a few phones with internal CID-based dial features, that I wanted to be allowed to be used. Your normal context is set up to operate with a '9' (or whatever) in front; so it is clear that you will need a different context from which to dial, a context that doesn't have the '9' at the beginning. I appreciate your point, but it's not that hard to avoid having the 9 prefix at all (in a simple dialplan at least). So to be honest one might as well dump the whole dial 9 thing completely in the scenario you describe? I think the solution here is really that the CID type applications become aware of prefix digits and strip them. Anyone know of good solutions to this? Any backend solutions to get Asterisk to hook into Exchange server etc? Cheers Ed W ___ --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] Disconnect Supervision UK / BT solution?
Chris Earle (CBL) wrote: Sorry -- you're right, I didn't express the scenario properly .. The disconnect supervision problem is when I 'forward'/divert an incoming POTS call out another FXO channel to a mobile phone or POTS line. (POTS - Sangoma|Asterisk - POTS/mobile) When the incoming POTS hangs up and/or the mobile the person was connected to .. Asterisk/Sangoma doesn't hang the Zap channels up. Just to clarifydoes it all work ok if you are using SIP or IAX for the forwarded channels? Eg local SIP phones? I only have incoming zap lines in my config and with the exception of hangup on ringing I have found hangup detection to work fine. I have a fax machine forwarding in my config as well and again no problems yet with hangup on that. Does it fail to work *every* time, or just intermittently? Does CallerId work ok in your setup? (can be a clue to help diagnose your setup) Ed W ___ --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] Disconnect Supervision UK / BT solution?
Matt Brown wrote: Well, I have just phoned BT today who said they can increase the CPC value on the line - however it needs to be done at the exchange - and has been booked for Tues. I suppose I will know wether this worked on Tues :-) - I shall post my findings. I would be keen to hear your findings - however, I'm still not clear exactly what the problem is in your case. There are numerous kinds of disconnect problems - which one are you having (so we know which one the CPC fixes...) Cheers Ed W ___ --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] Disconnect Supervision UK / BT solution?
Does anyone have any thoughts/confirmation about this finally being a viable solution? This disconnect supervision problem has plagued TDM and Sangoma cards for a long time! Just to be clear, what is the exact disconnect problem that you see? I have three TDM cards in two different systems, one using PBX lines and one on a private BT line. Both of them have trouble detecting a caller who is ringing, but then hangs up before being answered by the asterisk system However, *all* of them are absolutely fine at spotting a normal hangup once the call is connected and I see no random disconnects during calls either. Can you confirm that this is what you mean, or whether it's something else? Ed W ___ --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
[asterisk-users] Asterisk not hanging up
I have a problem with calls not hanging up if for some reason the physical phone dies or gets unplugged I can demonstrate this in practice by making a call from a handset, then unplugging the handset from the power. The call remains active and asterisk never seems to disconnect it. More annoyingly when power is re-applied the handset comes back to life, won't receive incoming calls (because asterisk thinks it's busy), but likewise the handset itself doesn't think it's in a call so it can't retrieve the call or do a proper hangup. I have no NAT in place and the handsets are all set to register/login and qualify=yes set (which I had hoped would sort this...) The handsets are SNOM 360s but I don't think this is directly relevant. Asterisk is setup to use FreePBX dialplan (but again don't think this is relevant?) Can someone please suggest a way to ensure that the calls get hungup - we had a 9 hour call earlier before someone noticed It's rare, but the consequences are potentially quite dire. Cheers Ed W ___ --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] TDM404B VS TDM2401B
Hi i'm not very happy with TDM404B voice quality, low volume Check the gain set in the zap config file. You can increase the in/out gain quite a bit over standard. Echo is frequently a symptom of wrong country settings, hence wrong impedence settings. Also endpoints matter Ed W ___ --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] TDM2400 Hardware Echo Cancel
Hi Echo cancel almost works, but the users hear what they describe as a 'crackle' coming back when they talk. Just a thought, but check that your gain levels are not too high? Ed ___ --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 not hanging up calls
Simon Tennant wrote: I have noticed that Asterisk (version 1.2.13) is not hanging up a call when the wifi handset moves out of range. My setup is Nokia E61 connected to wifi access point (private IP range) and then to server on internet (public IP). I have been testing using the talking clock application, and walking out of range does not hang up the call. I can reproduce the same problem by simply unplugging a normal SIP handset from the power during a call (or it crashes and locks up). When the handset gets re-booted the call is left in progress, new incoming calls aren't taken (because asterisk thinks that the handset is still in a call) and other problems I added an L() entry on the dialplan to limit calls to something sensible in the meantime, but would like to get a proper workaround? Any thoughts Ed W ___ --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
[asterisk-users] TDM 400P in the UK - doesn't see ringing calls hanging up before answer
Using a TDM400P in the UK nearly works fine, but I have a last remaining problem in that if the incoming is ringing and then the caller hangs up, asterisk takes another couple of rings before it spots the hangup. This is annoying in that I sometimes get phantom calls late at night (possibly due to call waiting or the exchange doing a half ring to see if we are live). Also I get phantom calls on either the voicemail or when I answer there is just dial-tone because the caller hungup before the call was answered I have fiddled with a number of settings relating to polarity reversal because I believe that might be relevant to BT's implementation, but it's not made any difference from the default config. Any suggestions on how to fix this from UK users? I have tried most of the suggestions in the voip wiki to no effect (haven't tried calling BT and asking them to change any settings yet) Thanks for any thoughts Ed W ___ --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 not hanging up
Philipp Kempgen wrote: Ed W wrote: I have a problem with calls not hanging up if for some reason the physical phone dies or gets unplugged Have you tried the RTP timeout settings in sip.conf? Sounds exactly like what I need! Thanks Is there no default set then?? Cheers Ed W ___ --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
